Televisor Hisense H32n2100c LED 32" HD TDT HDMI   





Resolución HD, acerca tu televisión a la realidad Una nueva forma de entender y ver la televisión. La resolución HD mejora sustancialmente la experiencia del espectador a la hora de disfrutar de la programación televisiva. Acerca la realidad a tu televisión, vive en alta definición. Modo hotel, sintonizador satélite El modelo N2100 viene especialmente preparado para ser utilizado en hoteles. Un modo más eficiente para la configuración de canales y características específicas del televisor Diseño compacto, el compañero ideal Un televisor todoterreno, estable, de líneas clásicas de las que nunca se equivocan. El compañero ideal para cualquier estancia de tu casa, ya sea el salón, la cocina o un dormitorio Tu televisor como centro multimedia Equipado con tres puertos hdmi y un puerto USB 2.0, pensados para que conectes tus gadgets y disfrutes de tu contenido favorito en una pantalla de alta definición Tipo: H32N2100C Material / Color frontal: Plástico / Negro Material / Color trasera: Plástico / Negro Características: HD Feature TV Relación de aspecto: 16:9 Tamaño pantalla diagonal: 32 pulgadas Resolución: HD 1366x768 Ratio curvatura: 0 Ángulo de visión: 178º Colores pantalla: 8bits TCON Profundidad color: 8bits Rango de colores: 45% (BT. 2020) Frecuencia Vsync: 60Hz Contraste: 3000:1 (Tpy.) Brillo (nits): 180 (Tpy.) TV Peak brillo y duración: 200 Tecnología de retroiluminación: DLED Tecnología Quantum Dot: No Local Dimming: No Local Dimming Zones: No FRC: No Tecnología 3D: No 2D a 3D: 2D Ratio de refresco PCI: - Tecnología HDR: No Clasificación HDR: No Procesador: SIS 9203B Escalado Ultra HD: No MEMC: No Decodificador: H.264, MPEG4, MPEG2, VC1 Input Lag: <50ms 3D Digital CoMB Filter: Sí Potencia de Salida: 2 x 6W Tecnología de sonido: - NICAM / A2 Stereo: NICAM Stereo Ecualizador: No Especial: AC-3 Media Player: Sí (Vídeo, Audio, Imagen) Multi-Language OSD: 15 Idiomas Hotel TV: Sí Interfaz del usuario: Sí APPS: No Buscador web: No HbbTV: No Soporte MHEG: No Anyview Cast: No Anyview Stream: No Recepción digital (DVB): DVB-T / C Soporte LCN: Sí Soporte Teletexto: Sí Control Parental: Sí PVR: Sí Time Shift: Sí Lista de canales favoritos: Sí Common Interface (CI+): v1.3 HDMI1: HDMI 1.4, CEC HDMI2: HDMI 1.4, CEC HDMI3 y HDMI4: HDMI 1.4, CEC USB (2.0): 1 USB (3.0): 0 Salida S / PDIF: 1 (optical) Salida Auriculares: 1 Sintonizador RF: 1 Entrada Composite (AV): 1 Entrada Component YPb(Cb) / Pr(Cr): 1 SCART: 1 (sólo SCART) Entrada VGA: 1 Entrada PC Audio (3.5mm): 1 Subwoofer: No Ethernet RJ45 port: No Wireless LAN adapter Support: No Fuente de alimentación: 100V-240V 50 / 60Hz Clase Energética: A Consumo en espera: =0.5w Power off: No Power Consumption (placa): 50W Consumo anual en (placa) (Kwh): - Power Consumption (ERL): - Consumo anual (ERL): - Ratio de luminancia (%): >65% Contenido de mercurio (mg): 0 Contenido de plomo: Sí -Este televisor contiene plomo sólo en algunas partes o componentes, de conformdad con las cláusulas de exención de la Directiva RoHS. WiFi Technology: No Peso neto / Peso Bruto (Kg): 5.8Kg Peso neto con base: 4.4Kg Peso neto sin base: 4.3Kg Peso Package: 1.1Kg Peso Polispam de la caja: 0.24Kg (EPE) Package (Horizontal / Vertical): Vertical Tamaño Package (Anx×Alt×Prof): 816×137×520mm Tamaño neto con base (Anx×Alt×Prof): 734×171×474mm Tamaño neto sin base (Anx×Alt×Prof): 734×93×434mm Distancia máxima entre 2 bases (Anx): 632mm EAN: 6942147434352 Material / Color base: Plástico / Negro Base (AnxxAlt): 187x20mm VESA Mount: 200x200 Mando a distancia: ER-22601A Pilas: AAAx2 Quick start page: Sí Cable: Sí Carga contenedora (40HQC) : 1230 Formato contenedor: .AVI, .ASF, .MKV, .TS, .RM, .FLV Video: .AVI, .MPEG, .MPG, .DAT, .VOB, .TS, .TRP, .TP, .MKV, .FLV, .WMV, .ASF, .RM, .RMVB Audio: .MP3, .AAC, .MP4, .WAV Imagen: .JPG, .JPEG

Precio: 186,99 € (Iva incluído)




          Epic Stock Media Game Voices WAV | 583.30 MB |   
Epic Stock Media Game Voices WAV File size: 583.30 MB 1837 files included - Character Voice Asset Audio Pack. Includes Character Voice - Calvin Character Voice - Jimmy Character Voice - Kimberly Character Voice - Sharon Character Voice - Timmy Character Voice - Yury All are High quality vocal assets. All 24-Bit 96K perfect for editing and tweaking for your purpose. Give Your Game characters the voice they need! ......:::::: Product Specifications ::::::...... * Format: WAVs * 0363 x Character Voices - (Calvin) * 0185 x Character Voices - (Jimmy) * 0444 x Character Voices - (Kimberly) * 0274 x Character Voices - (Sharon) * 0285 x Character Voices - (Timmy) * 0286 x Character Voices - (Yury) * 1837 x Individual WAVs Files * 96.1kHz 24-Bit High Quality * Compatible With All DAWs * PC & Mac Compatible Homepage http://www.anonymz.com/?http://epicstockmedia.com/product/game-voices DOWNLOAD: http://nitroflare.com/view/2BFE64D00851DC5/mh88c.Epic.Stock.Media.Game.Voices.WAV.rar
          VIŠE OD ISTINE – Prvi odjeci   
  
Moja knjiga razgovora sa Đorđem Kadijevićem, pod naslovom Više od istine koju je nedavno objavio Orfelin, doživela je svoju prvu veliku premijeru pred publikom u beogradskoj Kinoteci, 20. juna.

Prijatno sam iznenađen brojem ljudi koji su svojim prisustvom uveličali ovo veče, tj. koji su došli da pogledaju Kadijine ređe prikazivane filmove PRAZNIK i POHOD, a naročito (još većim) brojem onih koji su hteli da čuju šta o ovoj knjizi imaju da kažu njen autor (ja), njena glavna „zvezda“ (Kadija) i kritičar Ivan Velisavljević.

U sali u Kosovskoj okupilo se pedesetak posvećenika dobrog filma i dobre knjige te poštovalaca Kadijinog lika i dela – a što je najlepše, kako se pokazalo, to nisu bili samo pasivni obožavaoci, nego ljudi koji su imali šta i da pitaju potom. Najzad, bio sam zatečen brojem onih koji su na licu mesta hteli da kupe ovu knjigu i da tu dobiju i Kadijin potpis i posvetu.
Na slikama možete videti samo jedan deo reda – ovo su „samo“ oni koji su ostali posle filma POHOD kako bi dobili potpis – bar još toliko ga je dobilo odmah posle promocije, ali nije bilo vremena za sve jer je trebalo da ubrzo krene i film...
Knjiga je, kako i dolikuje, već izazvala interesovanje medija i ozbiljnih proučavalaca. Večernje novosti su objavile opširnu najavu o knjizi i promociji, a RTS je došao i snimio prilog za Kulturni dnevnik, koji je bio emitovan iste te večeri (20. juna). Taj prilog, u trajanju od dva minuta, sa vrlo prikladnim naslovom VIŠE OD KNJIGE, možete videti OVDE.


Mene i Đorđa ste već dovoljno slušali (na blogu možete naći audio zapise brojnih naših zajedničkih tribina; ko želi neka ih potraži) pa neću sada kačiti ceo snimak; uostalom, biće još promocija i prilika da nas vidite i čujete. Za ovu prigodu ću sa promocije samo izdvojiti izlaganje o Kadiji koje je imaoIvan Velisavljević – taj audio odlomak (10 minuta) možete čuti ako skinete fajl OVDE. 


Takođe, ovom prilikom imam zadovoljstvo da vam predstavim prvi ozbiljan KRITIČKI OSVRT koji je ova knjiga dobila. Autor je Vladimir Kolarić a izvorno je objavljen na forumu sajta Pouke. S njegovim dopuštenjem sada ovaj tekst prenosim i ovde.

Filmski duh Đorđa Kadijevića u ljubavi i istini

Vladimir Kolarić
  
Izuzetno je malo srpskih filmskih autora koji su se, makar i implicitno, doticali religijske tematike, a o predstavljanju hrišćanskog duhovnog, pogotovo crkvenog, iskustva da ne govorimo. Đorđe Kadijević (1933) je jedan od retkih čije filmsko svedočenje u egzistencijalnom iskustvu čoveka nesumnjivo poseduje na samo filosofske ili uopšteno „duhovne“, već i nesporno religijske, čak konkretno hrišćanske, konotacije. Takođe, Kadijević je svoje estetičke principe motivisao u prvom redu mogućnostima filma da na jedinstven način svedoči o onome što je on nazvao transcendencijom.
Ovo potvrđuje i nedavno objavljena knjiga razgovora, „Više od istine: Kadijević o Kadijeviću“ (Orfelin, Novi Sad, 2017), koje je sa ovim rediteljem vodio teoretičar književnosti i filma Dejan Ognjanović.
Ovaj reditelj, široj publici najpoznatiji po televizijskom filmu „Leptirica“ i seriji „Vuk Karadžić“, o sebi govori kao o „religijski zainteresovanom“ čoveku i umetniku, koji nije „religiozan u pravom smislu reči“ i „ima averziju prema konfesijama“, ali nije ni ateista, jer „ateista je zagrižen čovek koji tvrdi kako boga (u knjizi napisano malim početnim slovom - prim. V. K.) nema“.
Kadijević sopstveni odnos prema životu, definisan teškim ratnim i poratnim iskustvom, određuje nedostatkom „poverenja u život kao nešto što je zagarantovano“, objašnjavajući: „Ja sam shvatio da sam zrno na vetru i da visim na dlaci, i da je to život u stvari. Da je život jedno stalno izbegavanje smrti“.
I Pinki Zelić je video Tita-Kadiju

Ovakav doživljaj prolaznosti i konačnosti života kod njega je izgradio jedan trajan senzibilitet „preziranja razlike između života i smrti, sreće i nesreće, dobra i zla“, ali koji se, vidimo, nije razvio u odricanje od stvaralaštva ili moralni relativizam, već u osoben „životni“ i umetnički etos ljubavi, solidarnosti i stvaralaštva.
Iako „ne zna da li postoji bog“, Kadijević svakako zna „da se i pored svega može voleti, i da postoji ljubav“, koja je „jača od svega“: „I u tom smislu je Sveti Pavle bio u pravu kada je rekao ‘Velika je vera, ali veća od vere je - ljubav’. To je ostalo u meni i to je moje jedino bogatstvo. Niko u meni nije ubio ljubav, ničim.“
Pored negativnog određenja svog životnog vjeruju, Kadijević u knjizi daje i jedno, uslovno rečeno, pozitivno: „Ljudi su bića kratkoročna, ugrožena sa svih strana, spolja i iznutra, i baš zato što su konačna, što imaju samo jedan život, i samo jednu šansu da u tom životu budu srećni ili nesrećni, onoliko koliko je to moguće, naročito kada je sreća u pitanju, najnormalnije ponašanje je solidarnost: znači, uzajamno pomaganje i potiskivanje zla koje je svuda oko nas.“
Mi, prema Kadijeviću, „živimo u jednom tuđem svetu u koji smo bačeni“, dok „naš, ljudski svet“ nosimo „samo u sebi, i to kao veliku iluziju, kao motiv egzistencijalne nostalgije, kao jedan san“. Ali, „umesto uzajamne solidarnosti, bratimljenja i onoga čemu nas je Hrist učio, ljubavi jednih prema drugima, umesto jednakosti i bratstva u zloj sudbini u čije šake smo ubačeni, mi imamo tu situaciju uzajamnog sukobljavanja i pospešivanja efikasnosti zla“, što govori „da je čovek na neki način nedovršeno biće, kome fali jedna greda u glavi da bi shvatilo pre svega sebe a zatim i drugog“. Stoga, sve što ljudi rade, „podrazumevajući i umetnost, jeste pokušaj da prevaziđu to žalosno iskustvo neporazuma“.
Okruženi neprijateljskim svetom smrti, „sve što dotaknemo je mrtvo, i svaka trunka života je tako čudna, neočekivana, kratkotrajna i bespomoćna u trenutku dok traje“. Ali, umesto da budemo „dirnuti time i da te iskre života „podržimo koliko je moguće“, mi „toliko radimo na tome da se pridružimo svemu drugom što uništava život i svaki smisao bitisanja“.
Kadijević svoje opredeljenje za umetnost i stvaralaštvo objašnjava upravo ovakvom zapitanošću o smislu života i postojanja: „Ja, suočen za bezizglednošću takve situacije, ne nasedam ni na kakvu nadu, ali moram da priznam da je moje bavljenje umetnošću u stvari pokušaj izbavljenja“, koje se sastoji u borbi protiv takve egzistencijalne bezizglednosti, „jer ne želim da svojim delanjem u životu doprinesem tom haosu u kojem jesmo“. Umetnost može biti sredstvo borbe protiv samouništavajućeg univerzalnog sistema agresivnosti i represije, koje nema korene samo u ovim ili onim ideološkim pretpostavkama, već u samoj, u osnovi smrtnoj, prirodi sveta u kom živimo.
Ni ovde se Kadijević ne ograničava samo „negativnim“ objašnjenjem svoje motivacije da se bavi umetnošću i, još konkretnije, filmom. Film za njega ima moć da obuhvati „totalitet pojma čoveka kao bića“. On ima sposobnost da realnost oslobodi „tereta prozaičnosti i trivijalnosti“, da „oslobodi bivstvo tog tereta i pretvori ga u nešto tako božantveno da to postaje sakralno, sveto“.
Najviše što umetnost može da učini jeste da nekom promeni život, ali ne u smislu promene „stvarnosti u životu jednog čoveka“, poput društvene stvarnosti i političkog uređenja, već stupanjem u „područje transendencije“. Umetničko delo je samo ono koje sadrži u sebi „elemente trancendencije“, što se postiže „odstupanjem od merljive stvarnosti, od svega što je rečima izrazivo“, prevazilaženjem „notorne zbilje“ i „nadilaženjem svake iskustvenosti“, u „nešto što je više od istine“.
Prema tome, precizira Kadijević, u ravni mističnog realizma jednog Dostojevskog, „transcendencija je moć nadilaženja onoga što je iskustveno i vezano isključivo za objektivnost, za fenomen egzaktne izvesnosti do koje možemo doći“.
Delo, kao i reči Đorđa Kadijevića izrečene u knjizi „Više od istine“, duboko su i ubedljivo svedočanstvo o palosti sveta, ali i mogućnosti ljubavi i solidarnosti u tom i takvom svetu, u čemu veliku ulogu može da ima celokupno ljudsko stvaralaštvo, a posebno umetnost. Ova obimna i odlično strukturisana knjiga je stoga nezaobilazno štivo ne samo za one zainteresovane za film i umetnost, nego i za temeljna pitanja smisla.

*

Knjiga Više od istine tek započinje svoj život: sledi joj veoma iscrpan kritički osvrt (na više od pola strane) u „Kulturnom dodatku“ Politike, u broju ili za 1. ili za 8. jul iz pera Igora Stanojevića, a ubrzo zatim doživeće i svoju premijeru u inostranstvu, u Sloveniji, na festivalu Grosman (11-15. jula), gde ćemo je Kadija i ja predstaviti poštovaocima njegovog dela. Njih pouzdano znamo da i tamo ima, jer su ovog autora već toplo primili pre tri godine, kada je na Grosmanu dobio i nagradu za životno delo.
Ako do sada još niste nabavili ovu izuzetnu knjigu, ne čekajte previše – ovo je mnogo više od knjige o nekom reditelju i o nekim filmovima: misli i mudrosti koje ona sadrži univerzalne su, što znači – stalno aktuelne. Možete joj se vraćati kako radi opštih životnih mudrosti tako i povodom brojnih dnevnopolitičkih zbitija u balkanskom cirkusu: ova knjiga je nešto kao Ji Đing, Knjiga Zakona i Biblija crnila, ali i prkošenja crnilu: otud joj pohvale stižu i od vernika i od nevernika...


          RAD Video Tools 2.7b/1.300b   
Mit diesem Tool kann man endlich unter Windows die Smacker-Movie-Files abspielen. Diese Filme kommen in der Regel in Spielen als Intro- oder Extro-Sequenzen vor. Der Windows Media Player kann diese Dateien nicht abspielen. Sehr ausführlich fällt das Konvertierungs-Tool aus. Sämtliche unterstützten Formate (auch sehr viele bekannte Video-, Bild- und Sound-Formate wie .mov, .avi .wav oder mp3) können in avi, wav oder eine Vielzahl an Bildformaten...
          Mark Skaife crashed out: Bathurst 1000 trackside audio commentary   
THE jubilation for Holden Racing Team's poll position driver Mark Skaife ended a second after the Bathurst 1000 race started and only got worse. Hear our Podcast commentary.
          Bathurst 1000 mid-race report and PODCAST commentary   
HOLDEN's Rick Kelly has taken the lead at the halfway mark of the Bathurst 1000 although several crashes have turned the 161-lap into a safety car parade. PODCAST commentary included.
          Bev Brock audio interview   
PETER Brock was fondly remembered before the Bathurst 1000 race started and CASRguide caught up with his wife Bev Brock shortly after the ceremony to mark the race greats place here today.
          V8 supercar engine audio   
WE'LL admit it, even for hard-boiled V8 tragics this is pretty tragic. But the pure unadulterated sound of V8 supercars screaming down the straight is a sound worth preserving.
          Episode 6: PSR-X and the Mexican Standoff   

PHP-FIG member Paul M. Jones and PHP contributor Anthony Ferrera come on the podcast with Ben, Phil and regular guest Zack Kitzmiller to discuss the new Package Orientated Autoloader Proposal (a.k.a PSR-X), and whether or not PSR’s should ever be amended.

Paul explains why the new PSR-X is a handy idea for those who are already PSR-0 users, or even those that aren’t, Zack flips a shit and gets bored of talking about standards wishing everyone could just use Python and their awesome PEP-8, we convince Paul to try swearing for once and we all try badgering Anthony into agreeing with us for some reason.

Nobody wins, but the argument brings up a lot of interesting topics and points of view, and that is mostly what we are here for.


          Comment on FMS.NEXT Feature Open Thread by Robert M. Hall   
Here are some requests for FMS that relate more to music streaming and the player itself:1) The ability to handle all non-standard sampling rates in MP3's (odd rates, and MP3 Pro headers) that otherwise produce the "chipmunk" effect.2) Better calculation of the duration of VBR mp3's - currently it can be way off and using CBR mp3's is not the most efficient use of bandwidth/resources.3) Support for more ID3 tags - including retrieving image data - there are some nice AS3 based solutions - but it would be great if the FMS server could return that info as native objects or as bitmap data directly.4) Dynamic VBR streaming of MP3 data from .wav and .aiff sources and live data sources (alternative to nelly moser for live audio)5) Licensing model just for streaming audio6) Licensing model just for streaming out videos7) Licensing model just for streaming media (audio/video) without other features8) Plugable encryption for securing streams - IE: we can plug in our own routine in AS2.0 or AS3.0 for byte encryption on server and client side.9) Multiple audio tracks synched to one video source10) Multiple video tracks synched to one audio source - ie multiplexed and muxed streaming of multiple assets through one netstream11) More documentation with more examples on real world usage12) Standardized framework for allowing CDN networks deploying FMS to provide server side AS or a subset of server side AS scripts to CDN service subscribers that is easy to manage and push out to all edge servers. IE: autosync to all servers from one app source, with granular control over server side AS.More to come...Thanks!Rob
          Convert WMA to MP3 in Ubuntu   
In order to convert WMA file into MP3 format in Ubuntu (or Debian) you should install the following requirements first: 1. mplayer (sudo aptitude install mplayer) 2. lame (sudo aptitude install lame) 3. win32 codecs (sudo aptitude install ubuntu-restricted-extras) and then convert file.wma into file.mp3 using the following console command: mplayer -vo null -vc dummy -af resample=44100 -ao pcm:waveheader file.wma;lame -m s -V 3 audiodump.wav;mv audiodump.wav.mp3 file.mp3;rm audiodump.wav This also […]
          Woman Arrested After Faking Constipation   

What would you do if you knew you were about to be arrested? Would you give yourself up peacefully? Would you try to run? Would you plead? Would you... pretend to be constipated? Guess which one this week's perp derp picked.

According to the folks at WAVE-TV, Myra Cook was at the University of Louisville Hospital when police showed up to let her know that she would be arrested for drug charges upon discharge. This put Cook into a panic, but she quickly came up with a foolproof plan—or so she thought. She told the officers she needed to use the bathroom before being ushered off to jail. The officer found this to be reasonable and allowed her to go.

After about an hour, she peeked out of the bathroom to let the officers know that she was constipated and her arrest was going to have to wait. But the officer was having none of it and told her she had another minute before she would be charged with resisting arrest as well. So Cook improvised by faking a seizure. This did not go well, and she was ultimately sedated. Even after being sedated, she was not going to jail without a fight—it took seven cops just to get her into a police car! To be fair, being constipated can make you rather cranky.

Read the full article


          How to extract Wechat voice messages   
1) The voice messages are stored under /sdccard/tencent/MicroMsg/[akskjdkdkdjdkdkdajhueiieuiuowpq]/voice2. Each voice message has its own folder and is an amr file. Copy these *.amr files from your phone to your PC.

2) Download "wechat_silk" utility from here and extract the "wechat_silk.exe" file onto your computer. Drag the amr files and drop them onto "wechat_silk.exe". You'll get *.wav files on your desktop.

Now you can play these messages on your PC.

Optionally, you may convert and combine the wav files into a single MP3 file:

3) Go to a wav-to-mp3 site such as this to convert wav files to MP3 files.

4) Go to a MP3-combine site such as this and you'll finally have all your wechat voice messages in one MP3 file.
          Comment on Onoe Caponoe – “Milkyway 1311” by Onoe Caponoe – “Lazer Riddim” – .wavdiggers   
[…] has been cold for ages. A UK creative with a flare for the dramatic as the visuals for his Mutant Joe Lazer Riddim shows. […]
          Podcast Ep.19 – Respuestas a las preguntas de los Fans   
https://gallamijos.files.wordpress.com/2017/06/gallamijos-ep-19.wav Charly Lacy pide su bono por hab
          FFMPEG: how to extract audio from a specific timecode point?   
Hi but sorry I'm a Cat :o please I have a .MXF file that have a specific initial "timecodeTimeCode_FirstFrame : 16:39:24:24" for example now I would like extract an audio .wav file from timecode 20:02:05 and 10 frames (20:02:05.10) by this .mxf file and have - as duration...
          12 Agents Representing Short Story Collections   
Finding an agent to represent a short story collection is not easy. The reason most agents prefer to avoid short story collections is that they are difficult to sell to publishers - unless the author is already famous.

Before you submit your collection, I strongly recommend that you get as many stories as possible published in literary magazines. Having a track record will help, and prior publication will not harm your chances of getting a collection published. (However, you should avoid publishing in magazines that are online. If people can read your stories for free, why should they buy them?)

For where to publish short stories see:

297 Paying Markets for Short Stories, Poetry, Nonfiction

Speculative Fiction Magazines Accepting Submissions

18 Paying Markets for Humor 

And if you prefer to submit directly to a publisher read:

17 Publishers Accepting Unagented Short Story Collections

IMPORTANT: These twelve agents have listed Short Stories on their MSWL (manuscript wish list) profiles, however you should NEVER query an agent without checking the agency website first. Submission requirements change, and agents may close their lists, or switch agencies.
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Jennifer Kim (Sandra Dijkstra Literary Agency)

A graduate from the University of California Irvine, Jennifer holds a B.A. in English Literature and Spanish Literature, and spent a year studying Spanish literature and culture at the University of Barcelona. She also works as a bookseller, having done so since 2012.
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Renée Zuckerbrot founded the agency after working as an editor at Doubleday and Franklin Square Press/Harper’s Magazine. She is a member of the AAR and Authors Guild. She serves on PEN’s Membership Committee, and is a Board member of the Council of Literary Magazines and Presses (CLMP) and Slice Magazine. You can read an interview with Renée and her colleagues at Poets & Writers. See her top ten list of short stories at Storyville. 
______________________

Waverly Place Literary Agency 

Seeking short story and poetry collections with popular appeal.
______________________

Helen Boyle (Pickled Ink) (UK)

Helen Boyle has over fifteen years experience in the children’s publishing industry. She began her career at Hodder Children’s Books in the marketing and publicity department but quickly felt the draw of editorial and design. She has worked as a consultant, editor and reviewer for UK book and magazine publishers and has an extensive knowledge of the global children’s book market. She is seeking YA connected short stories. (A novel told as a series of short stories.)
______________________

Chad Luibl (Janklow and Nesbit

I tend to lean more toward darker tales and gritty settings, culture-crossing perspectives, structures that are a bit experimental (see David Mitchell’s Cloud Atlas), and always narratives with a strong emotional core. Having lived in Poland and Hungary, I have a niche-interest in books that feel Eastern European in voice/perspective (or explore post-Soviet and Cold War themes), and I find anything that deals with exile and expatriation immediately arresting.
______________________


I’m officially open to all genres. Whatever the age group, I tend to love contrast–highbrow sentences and lowbrow content, beautiful settings and ugly motives–the books that are beautiful and scary, heartbreaking and hilarious. I love secrets, scheming, revenge, plotting–and stories that have to be written forward and backward to make sense (I LOVE discovering a very cleverly planted clue that makes sense in retrospect).
______________________


Most generally, I focus on adult literary fiction, narrative non-fiction, middle grade, and young adult fiction.  Across all genres and ages, I’ll always be interested in the darker and weirder side of the human condition as well as previously under- or misrepresented experiences and voices.
______________________


I’ve enjoyed being an agent for 26 years, and love to find new voices. I love to lose myself in a story and to be transported to another reality, whether it be in the future, contemporary, or in the past. I am a complete francophile and have spent lots of time living in Paris and the South of France, and yes, I have sold lots of books from abroad, as I’m always working! I love to read.
______________________

Laura Biagi (Jean V. Naggar Literary Agency Inc.)

Some things I’m especially interested in at the moment: Magical realism in the vein of Gabriel Garcia Marquez, a literary fiction journey story where the characters change as much as the landscape around them, absurdism a la George Saunders, anything reminiscent of Karen Russell, strong female edgy voices like Miranda July, a story with a gothic or magical realist twist set in Kentucky or the South, literary YA with Romani characters, and contemporary YA with characters following their passions and trying to figure out life as in Rainbow Rowell’s novels.
______________________

Monica Odom (Bradford Literary)

Monica is most actively seeking adult projects, but is open to some YA and MG (especially if it is NF or illustrated). She holds the same criteria no matter the age group: original storytelling, incredible voice, compelling characters, and vivid, detailed setting. She also likes to see a strong sense of narrative tension. Monica is serious about the fact that We Need Diverse Books and is looking for authentic representation of all characters, diverse or otherwise
______________________

Rachel Crawford (Wolf Literary)

I’m on the look out for literary and commercial fiction and YA. I’m drawn to stories that defy genre conventions and play with reader expectations, and I particularly enjoy dystopian fiction, eco-fiction, and apocalyptic narratives, as well as anything with a scientist protagonist. I love books that explore big ideas through compelling narrative.
______________________

Sarah Yake (Frances Collin Literary Agency)

A quirky, interesting voice is my number one consideration. I love a touch of humor, whether overt or sly. My reading tastes are wide-ranging and my goal is to keep building a similarly diverse, multi-genre list.

          【转载】缘来缘去,随心随意   

庄子曰:水静则明烛须眉,平中准,大匠取法焉。

水静伏明,而况精神。


圣人之心,静,天地之鉴也,万物之镜。

所谓,心静,自然凉!


听说过:禅的最高境地是无声;音乐的最高境地是无词;

爱的最高境地是无言;生活的最高境地是理解;

朋友的最高境地是此时无声胜有声;

那么,是否人的最高境地是心静?


心静如水,依物随形,只是试问世间有谁能做到呢?

心静如水,依物随形,

凡事到达了极致,剩下的只有恰似明月清泉自在怀的静谧!




缘来缘去,随心随意 - 逍遥客 - 逍遥客的网易博客
 



我们每个人都是赤条条来到这个世界上,

又手握空拳,一无所有地离世而去,并且终将化为尘土。


因此我想,我们不必渴求生活的至善至美,

凡是生活给予我们的,我们都可以在平静中欣然接纳,

不必去追求那些可有可无,并不影响生命内涵的东西。


缘来缘去,随心随意 - 逍遥客 - 逍遥客的网易博客
 

我们只需用平和的心态,坦然地面对一切,

你就会感到险壑也是滩涂,

峻岭也是平原,沧海也是一粟;


用宽容的态度,包容一切,

虽然你临时蒙冤受屈,但心里坦荡安稳;


看待事物,取他积极向上的一面,

什么荣辱得失、沉浮笑骂、是非曲直,

一切都能视之等闲。




缘来缘去,随心随意 - 逍遥客 - 逍遥客的网易博客
 
 
 



既然心静如水,那就应该是一汪的水面,

平静如镜,没有波涛汹涌,

却有思想,他的思想像微澜的涟漪,

一波一波向前推进,不会聚结在某一处,

停滞不前,任其杂草丛生,腐朽变质,

他永远是新奇的,充满活力的!


所以说,心静如水是一种奢求,是一种理想,

是我们追求的境地。




缘来缘去,随心随意 - 逍遥客 - 逍遥客的网易博客
 
 



寻求这样一种状态,纵然经历了风风雨雨,

却没有惊涛骇浪,虽然有过微波荡漾,

涟漪不断,却能逐渐趋向于微波不惊,宁静,淡定。


心静如水,不是消极怠世,

也不是躲避现实、漠对人生,

而是静水深流,在纷杂的尘世里,

为自己留下一片纯静的心灵空间;


不管是潮起潮落,也不管是阴晴圆缺,

你都可以免去浮躁,义无反顾,

勇往直前,轻松自如地走好人生路上的每一步。





缘来缘去,随心随意 - 逍遥客 - 逍遥客的网易博客
 
 



所以我一直觉得,

真正的唯美应该都是自然而然地从心底深处真实地出发的,

我们只能从生活里去寻找,这样努力的寻找是一种自助和自救,

而不是一种自欺欺人。


也就是说,我们必须要面对现实,

不躲避生活里所有的挫折和风雨,

我想幸福就是宁静心绪下深埋的一种感觉吧,

这种感觉就是我们人生中永远的美丽、永远的希望、永远的信心,

也就是我们生命得以存在与连续的意义。





缘来缘去,随心随意 - 逍遥客 - 逍遥客的网易博客
 
 



很多时候,我们明明知道在时光遂道有多少事物都永远不会重回,

可是那一种幸福的感觉却能够重新再次完完整整地进入我们的思维,

进入我们的心中,这样幸福感觉中的唯美,

也永远是我们心中彼此向往的如诗的心境。


当爱恨在岁月中消融,当看尽千帆过尽之后,

我们对于生活的态度便剥离了那些浮躁与不甘,

心湖干净得只剩下一汪沉静的碧水,

终于在朝朝暮暮的简单重复的生活中,

寻找到了宿世的安宁与平静,不让烦恼占据我们的思维,

那颗曾经火热的心依旧在跳动,却静静地如一潭清水,

能够在市井人流的熙攘喧闹中博取一份超然物外的心境,

这时我们会笑着对自己说:人生至此,夫复何求?



缘来缘去,随心随意 - 逍遥客 - 逍遥客的网易博客

 



逍遥客的网易博客 - 逍遥客 - 逍遥客
 
 
 

http://zgzmzr.blog.163.com/


逍遥客祝福朋友幸福、快乐!



          马云纽约演讲 彻底震撼了美国商界   
著名的纽约经济俱乐部首次邀请中国企业家发表主题演讲,1200个位置超卖,火爆程度堪比NBA总决赛。这些掏钱抢门票的美国商界精英,为什么说他们的认知被改变?马云到底讲了什么?
马云纽约演讲    彻底震撼了美国商界 - 逍遥客 - 逍遥客

 

美国打篮球很棒,帮助大企业;中国就要打乒乓球,帮助中小企业。


亚马逊们是购物网站,人们知道要什么买什么;阿里巴巴是生活方式,人们买到的是惊喜和体验。


未来十年将有五亿中国人成为中产阶层,他们对优质产品的需求非常强大。


阿里巴巴要帮助美国小企业和全球小企业,将产品卖到中国。


DT时代诞生了新的商业形式——平台,“第三次世界大战”也将因此打响。


这是一场给美国企业界带来强烈冲击的演讲。

美国纽约时间6月9日中午(北京时间6月10日凌晨),马云受邀在著名的纽约经济俱乐部,向近千位商界领袖发表主题演讲,阐述阿里巴巴的美国策略和全球化策略。这是纽约经济俱乐部首次邀请中国企业家担任演讲嘉宾。此前朱镕基和董建华曾作为政界领袖受邀发表演讲。


原本预定800人企业家参加的活动,最终现场1200多个位置全部爆满。这些美国商界精英,都掏了钱来抢门票。演讲结束之后,全场将长时间的掌声送给这位来自中国的新经济领袖。


马云在演讲中提到,“阿里巴巴成长过程中,尤其是上市之后,不断有人问我们,啥时到美国跟Amazon和eBay竞争啊?" 他对此的回答是:我们尊敬这两家公司,不过我们的机会和策略在于帮助美国中小企业的发展。马云在演讲中用篮球和乒乓球比较两种思想差异:美国打篮球很棒,帮助大企业;中国就要打乒乓球,帮助中小企业。


马云说,亚马逊至今以自营为主,而阿里巴巴提供的是交易平台,是电子商务基础设施。“阿里巴巴和亚马逊的区别是,亚马逊们是一个购物网站,人们知道要什么买什么;阿里巴巴则是一种生活方式,人们买到的是惊喜和体验。”


马云还说,他此行是来帮助美国小企业,帮助他们将更多的优质产品带到中国,满足中国日益增长的中产阶层对优质产品和优质服务的强大需求。


现场演讲完之后,不少企业家上来和马云聊天,说这次演讲改变了他们对中国电子商务的很多认知。他到底讲了什么?


以下为马云演讲全文:

非常荣幸,从来没想到会有这多人来听我的演讲。

我站在这里的时候,感觉自己如此重要,如此责任重大。谢谢你们!


正式开始演讲之前,我想请问一下在座有多少人在使用阿里巴巴的服务?好的,不是非常多。那么,你们当中有多少人从来没有去过中国?好的。


20年前,我第一次踏上美国,美国之旅的第一站是西雅图。来到美国之前,我从课本、老师、学校和父母那里了解美国,我以为自己已经非常了解美国。但是,当我踏上这片土地的时候,我才发现我完全错了,美国这个社会和我从课本学到的根本不一样。在西雅图,我平生第一次认识了互联网。


回到中国之后,我告诉朋友们,我打算开一家互联网公司。我邀请了24位好友,讨论了两个小时。到了最后,还是没有人理解我想要做的东西,我们进行了投票,23人选择反对。我的朋友们劝我说:“忘了它吧!根本就不存在这么一个叫做互联网的东西,千万不要去尝试。”只有一个人对我说:“马云,我相信你,虽然我不知道你想要做什么,如果你想做,就大胆去做吧,因为你还年轻。”那年,我30岁。


没有任何的计算机知识、商业知识,我开始了创业之路。我开办了第一家公司,和我的妻子,还有一位同学。我们东拼西凑了1000美元,创业之路非常艰难。当时,我觉得自己是骑在盲虎背上的盲人(成语:盲人摸象,又骑虎难下)。20年风雨过去,我存活了下来。但是创业前三年,生活真的非常糟糕。我清楚地记得,我想向银行贷款3000美元,这花去了我整整三个月时间,我动用了所有的关系,还是没拿到贷款。每个人都认为马云在撒谎,因为1996的时候他们不相信有互联网这个东西的存在。


有一天,1996下半年,中国正式接入了互联网。于是,我邀请了10位媒体朋友到我家里来,想告诉他们我没有撒谎,确实存在互联网这个东西。为了下载一张照片,我们当时花了三个半小时。大家说,“那东西真的能行得通吗?”我说,“是的,行得通的,不是现在,是在未来十年之内。”不管怎样,这至少证明了我没有撒谎。


我还记得,当我们尝试帮助小公司在网络上销售产品,但是没有人愿意来,因为还没有人在网络上买东西。所以在第一周,我们的7个员工自己去买,自己去卖。到了第二周,有人开始在我们的平台上卖东西,我们买光了他出售的所有商品。我们有两个房间,堆满了我们那两个星期买的东西,而且是没有什么用的东西。这只是为了告诉大家互联网是行得通的,但这并不容易。1995年到1999年这期间,我们的创业失败了,没有一样条件是成熟的。


到了1999年,我邀请18个好友到家里,我们决定再一次尝试,并且把网站命名为alibaba.com。人们问为什么叫阿里巴巴?我们希望互联网就如同一个宝库,可以让小企业芝麻开门。另外,这个名字容易拼写,也朗朗上口。我们想做的事情,是帮助小企业。


当时我们注意到美国的电子商务致力于帮助大公司,帮助他们节约成本,而当时中国没有那么多的大公司,而是有很多的小企业,对于他们来说生存是如此艰难,假如我们能够用互联网技术来帮助这些小公司,这会非常有意思。


美国习惯于帮助大型企业,这就好比美国人擅长于打篮球;而在中国,我们应该会去打乒乓球,去帮助那些小公司。我们需要做的不是帮助小公司去节约成本,因为他们知道如何节约成本,他们需要学习的是如何赚钱。因此,我们的业务一直专注于帮助小企业在网络上赚钱。

马云纽约演讲    彻底震撼了美国商界 - 逍遥客 - 逍遥客

 

我们希望阿里巴巴这个公司可以活102年。人们会好奇地问,为什么是102年?因为阿里巴巴诞生于1999年,上个世纪我们经历了1年,这个世纪将是完整的100年,下一个世纪再经历1年,这样横跨三个世纪,102年。我们这是给了所有员工一个清晰的目标。无论我们有多少盈利,无论我们赚了多少钱,不论我们已经取得什么成绩,都不要认为我们已经成功。不要忘记我们希望活102年,现在才过了16年而已,前面还有86年。这86年中的任何一个时间,如果公司倒闭了,我们就谈不上成功。当我听到这个俱乐部(纽约经济俱乐部)已经有108年的历史,我十分惊讶,十分震惊。这其中一定有很多值得阿里巴巴学习的东西。


没有人相信阿里巴巴可以活下去。以前人们说:“你们的平台是免费的,你们的公司那么小。”尤其是我们在美国上市的时候,人们又说:“你们阿里巴巴是做电子商务的,就像亚马逊一样。”可能亚马逊是美国人眼中唯一的电子商务模式,但是我们不一样。


我们和亚马逊不一样的是,我们自己不做买卖,我们帮助中小企业做买卖。在阿里巴巴的平台上,有一千万家小企业每天做交易。我们自己不送快递,但每天有二百万人帮着我们配送三千万包裹。我们也没有自己的仓库,但是我们帮助那些中小物流快递公司管理成千上万个物流仓库。我们也没有任何商品库存,但是我们有3亿5千万的买家,每天有超过1亿2千万的消费者光顾我们的网站。去年我们的销售额是3900亿美元。今年,我们预计销售成交会超过沃尔玛全球,你要知道沃尔玛用了230万员工,而我们只是从18人扩大到了3.4万人。


我们和亚马逊不一样的还有,亚马逊是一个购物中心在美国,电子商务是商务,而在中国,电子商务是人们的一种生活方式。年轻人交换他们的思想,互相沟通,建立信任,建立个人信用记录。就好像星巴克一样,你不是去星巴克品尝它的咖啡有多么美味。这是一种生活方式。这也正是互联网电商如何改变中国的地方。


我们感到自豪的,并不是我们卖了多少东西。我前面提到,今年我们的成交总额会超过沃尔玛——是的,我们对此很自豪。阿里巴巴会在未来五年,达到1万亿美元的成交额。这是我的目标,我认为我们会达到这个目标。更让我们自豪的是,我们为中国直接和间接地提供了1400万个就业机会。我们在中国乡村创造就业机会,我们为中国女性提供就业机会。中国互联网上成功的卖家中,超过51%是女性。


我们为这些事情感到自豪。有人又会说,阿里巴巴现在做到了这些,你们的下一步是什么?阿里巴巴无处不在,你们的未来打算是什么?今天,超过80%的在线交易是由阿里巴巴所创造,我们未来的目标是将阿里巴巴的业务拓展到全球。这不只是要成为最会卖货的公司。我们希望电子商务的基础设施能够全球化。相比美国,为什么中国的电子商务成长速度如此惊人?因为在中国的商业基础建设太差。不像在美国,你们有汽车,线下有无处不在的沃尔玛和凯马特(Kmart,美国现代超市零售企业鼻祖)。但是在中国,我们并没有这么好的基础设施。


马云纽约演讲    彻底震撼了美国商界 - 逍遥客 - 逍遥客

 

电子商务在美国如同餐后甜点,它是对主流商业的补充。但是在中国,电子商务已经成为主菜。我们建设了电子商务的基础设施。所以,如果我们将我们的电子商务基础设施全球化,包括在全球范围内提供支付工具、物流中心和透明公开的交易平台,帮助全球的小公司将他们的产品卖到世界各个角落,帮助全球的消费顺利地买到世界各地的产品。我们的愿景是,未来十年内帮助全球20亿消费者在线购买全世界的产品,而且做到全球范围内72小时内收到商品,在中国范围内,无论你身在何处,24小时内收到商品。阿里巴巴的全球化战略,仍然是致力于帮助小企业,帮助他们以最有效的方式来做生意。我们会在自己的电商平台上,帮助到另外一千万家小企业。


我们会给这些小企业赋能。我们会引导更多流量给他们,会给提供他们支付系统,提供物流配送系统,让他们更加快捷和便利地在全球任何角落开展业务。我们希望在中国以外的地方拥有40%的业务量。现在,我们的海外业务只占到2%。


可能人们继续会问,现在阿里巴巴的业务做大了,也募集到大量资金。你们会在美国做什么?你们会来美国吗?如果来美国,你们是打算入侵美国吗?马云你什么时候来和亚马逊竞争?什么时候来和eBay竞争?其实,我对eBay和亚马逊抱有敬仰之心。而我来这里的目的,也就是阿里巴巴的下一步战略,是帮助美国的小企业走进中国,帮助他们将产品卖到中国。


现在的中国,中产阶层的数量和美国人口大致相当。而我们认为,未来十年中国将有五亿人口成为中产阶层。他们对优质产品和优质服务的需求是非常强大,非常惊人的。我认为中国的现状没有办法满足他们对于优质产品和优质服务的需求。过去20年,中国一直致力于出口,而我认为接下来10年-20年,中国应该把注意力集中在进口方面。中国要学会进口,学会如何消费。中国应该去消费,去做全球买手。同时,我认为美国的小企业,美国的品牌产品,也应该利用互联网,进入中国市场。


过去20年,美国的大公司已经遍布整个中国。对于美国的小公司来说,利用好电子商务,将是巨大的机会。阿里巴巴已经帮助很多美国的农民将产品卖到中国。西雅图的车厘子就是一个例子。你不能想象,美国驻华大使跑来问我:“马云,你们可以帮我们销售西雅图的车厘子吗?”那时车厘子还长在树上,我们就做了预售,8万个中国家庭抢购了这些订单,我们摘下车厘子,在24小时内运到中国。24小时内,8万个中国家庭就将160吨车厘子一抢而空。去年,我们卖掉了300吨车厘子。我不知道今年又会是怎样一个惊喜的数字。


我们同样也帮助销售阿拉斯加的海鲜。我们还帮助销售加拿大的龙虾,所卖出的龙虾数量,是他们十年不可能达到的。许多美国的知名品牌也借助我们的平台销售。像Costco超市入驻以后第一个月就卖出了600吨坚果,销售额达到650万美元。既然我们能够帮助卖龙虾,卖车厘子,为什么不可以利用我们的系统来帮助大量的美国中小企业?我还想借用一天的数据来举一个例子,11月11日,中国的光棍节,阿里巴巴把它变成了一个购物节。去年双十一,我们的交易额高达97亿美元,第一分钟有240万消费者涌入。今年,我们认为这个数字会更加惊人,所以这也是我此次来美国的目的——中国消费者需要更多的美国产品。


每天,阿里巴巴平台都有上亿“饥渴”的消费者来购物。这就是我们来到这里的原因,我们不是来竞争的,我们来这里是希望将美国的中小企业带到中国。我们的愿景是“全球买、全球卖”,未来10年-20年,无论你身在何处都可以买到任何地方的产品,也把可以自己的产品卖到世界各地。菲律宾人可以买到挪威的鲑鱼,挪威人可以将产品卖到阿根廷,阿根廷人可以买中国的产品,也可以将产品卖到中国。这是互联网未来变革的方向。最后,阿里巴巴为已经改变中国而感到骄傲。变革的力量是强大的。


第一次技术革命以后,世界有了新的商业组织形式——工厂,这些工厂带来了武器和力量,然后催生了第一次世界大战。第二次能源革命以后,世界又有了新的商业组织形式——公司,然后催生了二战。


在这个互联网时代,DT时代(DataTechnology,数据时代),我认为诞生了一个新的商业形式——平台,“第三次世界大战”也随之即将打响——但这不是一场国与国之间的战争,这是一场我们携手对抗疾病、贫穷和气候变化的战争。我认为这是人类的未来。所有的人类,所有的国家,都应该联合起来。依靠年轻一代,而不是武器;依靠计算机和大数据,来解决人类的问题,社会的问题。这是我所热衷的,和钱无关,这是梦想。改变世界的不仅是技术,更是你相信可以改变世界的梦想。我们知道这条路漫长且不易。我以前说过,过去20年在中国做互联网生意很不容易,今天很困难,明天更困难,后天很美好。但是绝大多数人死在了明天晚上,如果你不努力的话。

谢谢大家!

逍遥客祝福朋友幸福、快乐!



          Halloween Automated Scarer 2015   

A pinch of pygame, a hint of scapy and the Halloween concoction is ready!

This year's automated scarer was a bit more challenging than last. I stitched this together in a couple of days and ... it worked mostly as I wanted it to be. The most important thing is that children and parents enjoyed being scared by a few monstrous pumpkins and a bunch of my son's very horrific (but beautifully made) drawings.
Raspberry thirsty Vampire
Scary Witch of the South East

I wanted the animation to be easy to be activated and yes, I could have used other methods to detect people to be scared (which from now on I will refer to as scarables: a person suitable for the scarer) but I was curious about scapy and the possibility of detecting mobiles passing by.






I also found quite interesting digging through the enormous amount of information available on pygame which served perfectly the purpose of displaying the pictures and playing the sounds all of this without needing to drop into X leaving ample resources available on my RPi B

It is also entertaining to lookup the MAC addresses I collected to establish if we live in a posh area or not. The amount of Apple devices wasn't so many in the end, I will consider selling and buy elsewhere.
Ah and don't worry, the MACs I collected will be handed over to TalkTalk for safe keeping.



The Python script revolves mainly around the scapy command sniff which I used to search any broadcasting WiFi device within reach of my dongle.
The first thing to do is to setup the WiFi card into monitor mode which is done outside of python on the shell:

#/usr/bin sh
sudo service ifplugd stop
sudo ifconfig wlan0 down
sudo iwconfig wlan0 mode monitor
sudo ifconfig wlan0 up
sudo service ifplugd start

You can verify things are set properly by issuing iwconfig, this is my output:

wlan0     IEEE 802.11bgn  Mode:Monitor  Frequency:2.412 GHz  Tx-Power=20 dBm
          Retry short limit:7   RTS thr:off   Fragment thr:off
          Power Management:off

Within scapy's lamda function we need to create a list to hold the MAC addresses that we'll acquire. Around the area where I live there are normally quite a few Access Points and other WiFi devices. The first thing to do then is to store all these MACs on a file so that they will not trigger any false "scares" each time that the Pi starts. I did this by running the script in "silent mode" for a couple of hours. I had created in this way a WiFi fingerprint of the area which I stored on a file that gets read at the start of the script and pre-loads all the MACs in the neighbourhood.

if path.isfile(fileName):
    with open(fileName, 'rb') as MAC_file:
        MAC_list = pickle.load(MAC_file)

for the serialisation I am using pickle which for the purpose works just fine

From here on any new MAC detected is a potential scarable,

Scapy offers a sniff command that can take as arguments an interface and a function amongst other options.

sniff(iface = "wlan0", prn = PacketHandler, store=0)

The store=0 turned out to be vitally important as it tells the sniff command to just monitor the packets and not store them in memory. I learned this the hard way as I the Pi stopped responding all together as its memory got full.
I have obviously specified to monitor packets on wlan0 and given a PacketHandler function which I use to specify what I am interested in about the packets seen by wlan0

def PacketHandler(pkt):
        if pkt.haslayer(Dot11):
                if pkt.type == 0:
                        if pkt.addr2 not in MAC_list:
                                MAC_list.append(pkt.addr2)
                                with open(fileName, 'wb') as MAC_file:
                                        pickle.dump(MAC_list, MAC_file)
                                        Call the Scare function

the layer I am interested in is Dot11 as in 802.11x and the packet type is 0 as in management. If the MAC is not in my list I will add it and write the list on a file.
The other very important thing that I do in the last if block is that I will start the animation for each new MAC added to the list.
Now, this could have done it in a number of ways, I wanted to explore the pygame events, I wanted to use two threads on for WiFi scanning the other for the animation, in the end at 16:30 of the 31st of November I decided to call a function directly after writing the list to a file.

image = images[random.randint(0, len(images)) - 1]
sound = sounds[random.randint(0, len(sounds)) - 1]
Scare(image, sound, screenSize)

The Scare function receives a random image and a random sound from two preloaded lists. I paste the PrepareImages only as the PrepareSounds is essentially the same

def PrepareImages():
        imagesNames = glob.glob('*.png')
        imageCache = []

        for imageName in imagesNames:
                try:
                        imageCache.append(pygame.image.load(imageName).convert())
                except:
                        raise UserWarning, "Could not load images"

        return imageCache

All .wav and .png files where in the same path as the script.
Everywhere I looked for examples of pygame.image.load I found that it is highly recommended to convert the images to improve performance. I certainly noted an improvement

The Scare function looks like this

def Scare(image, sound, screenSize):
        image = pygame.transform.scale(image, screenSize)
        frame = image.get_rect()
        white = (255, 255, 255)
        black = (0, 0, 0)
        screen = pygame.display.set_mode(screenSize)

        playing = sound.play()
        while playing.get_busy():
                screen.fill(white)
                screen.blit(image, frame)
                pygame.display.flip()
                pygame.time.delay(1000)
        screen.fill((black))
        pygame.display.flip()

it will resize the picture (sometimes deforming it as the source did not have the same proportions of the screenSize I selected), set the screen, play the sound.
Whilst playing the sound the picture would be displayed (blit) and a 1 second delay also added to get the shivers deep in the scarable's spine.
After the sound is played the screen turns black again.

Here is the script scarer.py

That's it really, ciao!








          Did Fargo get a reprieve?   

It's the 29th. The API was set to expire yesterday. But Fargo still works.

I tried creating a post and updating it. And sure enough it worked.

I even reloaded the app to see if the updates are still there, and they are.

Not complaining! But what the what?

I'm trying to think but nothing happens!


          引用 【转载】一组妇幼偏方、验方、秘方   
一组妇幼偏方、验方、秘方 - 逍遥客 - 逍遥客
 
一组妇幼偏方、验方、秘方 - 逍遥客 - 逍遥客
 
一组妇幼偏方、验方、秘方 - 逍遥客 - 逍遥客
 
一组妇幼偏方、验方、秘方 - 逍遥客 - 逍遥客
 
一组妇幼偏方、验方、秘方 - 逍遥客 - 逍遥客
 
一组妇幼偏方、验方、秘方 - 逍遥客 - 逍遥客
 
一组妇幼偏方、验方、秘方 - 逍遥客 - 逍遥客
 
一组妇幼偏方、验方、秘方 - 逍遥客 - 逍遥客
 
一组妇幼偏方、验方、秘方 - 逍遥客 - 逍遥客
 

逍遥客祝福朋友幸福、快乐!


          Zillow Protects Broker Data   

Zillow Logo

WAV Group supports brokers and MLSs in constructing data licensing agreements with users of data records that belong to the broker. When a company like Zillow ingests broker data, like all recipients, they must adhere to the data license agreement which typically requires that you cannot allow the data to be used by a third party. The blog McMansionHell.com is learning the hard way that using data from Zillow without proper authority is a copyright violation that Zillow will pursue. Not only will Zillow pursue the violation, but they may be contractually bound to pursue the violation. The media around the case is a bit confused, so let me try to break it down in layman terms. There are two methods for licensing data to Zillow Group. I say Zillow Group because they do a great job of creating one license agreement for all of their websites. In the absence of a license agreement, the default agreement is the Terms of Use on their respective websites. Zillow manages a number of license agreements depending on how they are acquiring the data. The agreements that they have with Franchise Organizations like Realogy, Keller Williams, RE/MAX, etc have a certain flavor. Agreements with brokerages are a little different. Agreements directly with the Multiple Listing Service are also different. The primary covenants of these agreements is that Zillow Group will use the data for its intended purpose and not allow anyone else to use the data. If the broker sends the data to Zillow, its generally for advertising purposes. They would not expect Zillow to collect the data for the purpose of advertising a home for sale, then have Zillow turn around and ship it elsewhere for some other purpose. In this case, Zillow did not provide a data feed to McMansionHell.com. McMansionHell.com is subject to the Terms of Use that are published on Zillow as a user. McMansionHell.com scraped the data off of Zillow. The process is pretty easy to do on any computer. Simply right click on the image and save it to you computer. Here is where it gets interesting. Sometimes you can use an image from Zillow legally, and at other times you cannot. The digital millennium copyright act allows for some content sharing online. However, the Zillow Terms of Use are very specific. You can only use Zillow Group websites for their stated intent – looking for a […]
          Douchebag of the Week (Aug. 15-21) Does anybody remember a time...   








Douchebag of the Week (Aug. 15-21)

Does anybody remember a time not so long ago when we actually didn’t mind Brett Favre because he could pull off a movie cameo and memorably remind Matt Dillon’s character why he was in Miami? You could darn near say we loved the sumbitch after that.

Oh, but how times have changed. Nowaday, it’s an annual thing to see this blubbering mongoloid get all misty-eyed at the conclusion of a football season, pretend to wrestle with thoughts of retirement while avoiding any training camp and then, ultimately, announce he’s gotta come back because he just loves the game too much.

And you don’t just have to be an aficionado of the Green Bay Packers to be understandably irritated when the NFL’s premier prima donna gets the president of his fan club SI reporter Peter King to ask him fully-loaded questions that further fuel Favre’s attempted victimization.

From King’s Aug. 4 column after Favre announced he would not be playing for the Vikings:

Favre was down. He just sounded beat, like he had nothing left to give, and a little depressed. “I’m sure I’ll regret it down the road,” he said.

I asked him about the toll this had taken on his reputation. “Two years ago you were ‘Sportsman of the Year’ and an American folk hero,” I said. “Now there are kids and adults who are sick of you, who don’t love you anymore. How does it feel?”

“Well, then they really didn’t love me in the first place,” he said.

Oh, and just to clarify, Favre wanted everybody to know this had nothing to do with “revenge against the Packers” or “getting back at [Green Bay GM] Ted Thompson"—which is good, because those two rather important groups weren’t mentioned in the question either, but it’s nice to know that this yearly charade certainly has nothing to do with them. And when he announced his inevitable return on Aug. 18—nearly one week after training camp’s conclusion, of course—the headline declared ”this is not about revenge.“ Indeed, it all now has to do entirely with feeding Brett Favre’s massive ego and hoping that some city somewhere every season will be willing to play along with this tired routine.

Great athletes are forever indebted to the cities whose fans supported them by purchasing tickets and wearing their jerseys. While Brett Favre now seems fully intent on donning a new uniform every year for his No. 4, he’s also committed to becoming No. 1 among douchebags.


             



D'OH



           XEBRA(プレイステーションエミュレータ)の使い方+αメモ(なんちゃって TAS 機能とか)   

知っている人には今さらだと思いますが、XEBRA / ARBEX というエミュレータが互換性・速度の面で非常に良かったです。本体にマニュアルが付属しているので、その通りに使えば良いのですが、若干落とし穴や目立たない機能もあるのでメモしておきます。

だんだん「外部ツールを使って高度な機能を追加するには?」みたいになってきました。

雑多な情報

  • Help... ダイアログの表示をスクロールするとじつは日本語の説明もある
  • OS ROM(よく BIOS と呼ばれるもの)は OSROM というファイルで配置する*1
  • CD イメージファイルはイメージ本体を選択する(ccd や cue は同階層に配置するが、オープンダイアログでは iso, img などを選択する)
  • CD イメージの読み込み後は、Power(Run) を選択しないと起動しない
  • Controller... ダイアログでキー設定が可能。ただし、最初に「Controller 1」などの設定対象を選択しなければならないことに注意
  • Simulation Model Image はハードウェア動作のシミュレーション設定を記録したファイル(SIM.EXE で作成・編集、XEBRA 本体で読込)
  • Running Image がどこでもセーブ(savestate)に相当する
  • File/History にある Pad History (パッド履歴) はキームービーに近いもの(後述)
  • Sound Output... で Save にチェックを入れると SOUND.WAV にプレイ音声が出力される
  • .kkapture (1.01) で AVI 形式の動画を出力可能
  • PK201.DLL を置いてあれこれするとポケステもいけるらしいです(見てない)
  • Debug 周りは不明(気になる)
  • 本体は gccコンパイル、UPX で圧縮されていて、画面描画は OpenGL を使用

パッド履歴について(ムービー機能として使う)

File/History にある Pad History (パッド履歴) は強いて言えばキームービーに近いものです。ただし、あくまでムービー機能ではないので、パッドの入出力だけを扱い、ステートなどは一切記録しません。

最初に、インタフェースが独特であることを説明すべきでしょう。

  • Read Pad... - 既存のパッド履歴を読込、再生開始(ユーザー入力無効化、いわゆる「再生終了後」もそのまま)
  • Write Pad... - パッド履歴を新規作成、記録開始
  • Load Pad... - 記録中のパッド履歴の内容を、別のパッド履歴で丸ごと上書き(使い方は後述)
  • Save Pad... - 記録中のパッド履歴の複製を別名保存

「停止・クローズ」という選択肢はないので、別のファイルを新規作成するなりして行うしかなさそうです。

これを実行イメージ(Running Image)と組み合わせることで、一般的なムービー機能として動作します(どの程度 desync せずに動作するかは不明ですが)。

注:必要に応じて事前にメモリーカードを初期化・変更すべきかもしれません。(実行イメージに含まれるかもしれませんが未確認です)

注:より正確に記録するには Simulation Image の入出力が必要だと考えます。しかし、すべて下記の手順では省略しています。

電源投入時からのムービーを記録
  1. ムービーを記録
    1. CD イメージを読み込む
    2. File/History/Write Pad... でパッド履歴を新規作成(例:from_power_on.pad)
    3. Power(Run) でゲームを開始、普通にプレイ
    4. 記録を止めたくなったら Write Pad... で別のファイルに切り替える(例:dummy_for_close.pad)か、単に XEBRA を閉じる
  2. ムービーを再生
    1. CD イメージを読み込む
    2. File/History/Read Pad... でパッド履歴を読込
    3. Power(Run) でゲームを開始、鑑賞
特定のステートからムービーを記録
  1. ムービーを記録
    1. 記録を開始したい箇所で Pause する
    2. File/Save/Running Image... で実行イメージを保存(例:from_snapshot.sav)
    3. File/History/Write Pad... でパッド履歴を新規作成(例:from_snapshot.pad)
    4. Power(Run) でゲームを開始、普通にプレイ
    5. 記録を止めたくなったら Write Pad... で別のファイルに切り替える(例:dummy_for_close.pad)か、単に XEBRA を閉じる
  2. ムービーを再生
    1. CD イメージを読み込んだ状態で Pause する
    2. File/Open/Running Image... で実行イメージを読込
    3. File/History/Read Pad... でパッド履歴を読込
    4. Power(Run) でゲームを開始、鑑賞
ムービースナップショットの記録と追記

いわゆる追記(rerecording)もじつはできます。

  1. ムービースナップショットの作成
    1. パッド履歴(例:main.pad)記録中に、スナップショットを作成したい時点で Pause する
    2. File/Save/Running Image... で実行イメージを保存(例:snapshot.sav)
    3. File/History/Save Pad... でパッド履歴を複製保存(例:snapshot.pad)
  2. ムービースナップショットの読込
    1. CD イメージを読み込む
    2. File/History/Write Pad... で記録対象のパッド履歴(例:main.pad)を選択する(この時点で内容が失われるので、不安ならバックアップしておくこと!)
    3. File/Open/Running Image... で実行イメージ(例:snapshot.sav)を読込
    4. File/History/Load Pad... でパッド履歴(例:snapshot.pad)を読込
    5. Power(Run) でゲームを開始、記録続行
プレイ速度も変えたい!

本体の機能である程度のスローダウンが可能でした(後述)

すろぉもぉしょん というアプリを使うとこの上なく簡潔にスローダウンできます(XEBRA 本体とホットキーが重複していますが)。

もっと柔軟に!という方には MHS (Memory Hacking Software)Cheat Engine という2つのツールを紹介します。どちらもプロセスメモリエディタですが、Hotkey に Speed Hack (スロー・倍速)や Process Pause を割り当てて使うことができます。私はメモリエディタとしては MHS を好んでいますが、Speed Hack 用途では Cheat Engine の方が使いやすいように感じました。試してみて好きな方を使ってください。速度変更を行ってもフレームスキップ(?)が動作して操作単位が大まかになってしまうらしいので、"それなり"の効果しか期待できないかもしれません。

フレーム単位で操作するというのは現状では無理そうです。フレームカウンタに相当するものをメモリから探して、監視しながら動かすと、どの程度スローダウンに成功しているか確認できます。

見落としていましたが本体の機能で一定のスローダウンが可能です。起動後、Run/Sync を選択し(デフォルトで on なのが off になる)、その後 Run/Misc/Slow を選択することで速度を落とせます。4段階変速で、選択するたび 0→1→2→3→0... と変化します。より遅くしたければ複数回 Slow を選択してください。

チート(改造コード)

XEBRA の作者は cep というプロセスメモリエディタも公開しています。PAR コードが使えるのできっとおすすめです。プロセスメモリエディタなので XEBRA 向けの設定行を BASEARCH.TXT に書かなければなりませんが、ググれば出てくると思います(希望的観測)。

メインメモリ上の変数を入出力したいだけなら、どのプロセスメモリエディタでも簡単だと思います(本体からメモリ内容をダンプできるのでサーチが楽)。私は MHS を使用しました*2

ショートカット

Help ダイアログから抜粋。今後変更される可能性も考えられます。

  • F1 .... = メニュー : File/Save/Running Image/XEBRA.RUN
  • F2 .... = メニュー : File/Save/Running Image/????????.000
  • F3 .... = メニュー : File/Save/Running Image/????????.001
  • F4 .... = メニュー : File/Run/Sync
  • F5 .... = メニュー : File/Run/Power(Run)
  • F6 .... = メニュー : File/Run/Pause
  • F7 .... = メニュー : File/Open/Running Image/XEBRA.RUN
  • F8 .... = メニュー : File/Open/Running Image/????????.000
  • F9 .... = メニュー : File/Open/Running Image/????????.001
  • F12 ... = メニュー : View/Full Screen/

コマンドライン

これも抜粋。不必要だと思う人もいるかもしれませんが、.kkapture を使いたいときなどは楽になりそうです。

  • -SPTI "PATH" .... = メニュー : File/Open/CD-ROM via SPTI/"PATH"
  • -IMAGE "PATH" ... = メニュー : File/Open/CD-ROM Image/"PATH"
  • -OSROM "PATH" ... = メニュー : File/Open/OS ROM Image/"PATH"
  • -BU00 "PATH" .... = メニュー : File/Open/Memory Card 1 Image/"PATH"
  • -BU10 "PATH" .... = メニュー : File/Open/Memory Card 2 Image/"PATH"
  • -SIM "PATH" ..... = メニュー : File/Open/Simulation Image/"PATH"
  • -RUN "PATH" ..... = メニュー : File/Open/Running Image/"PATH"
  • -TOC "PATH" ..... = メニュー : View/CD-ROM Drive/Open TOC/"PATH"
  • -CUE "PATH" ..... = メニュー : View/CD-ROM Drive/Open TOC/"PATH" .. CUE
  • -CCD "PATH" ..... = メニュー : View/CD-ROM Drive/Open TOC/"PATH" .. CCD
  • -INI "PATH" ..... 初期設定イメージのロード
  • -RUN1 ..... = メニュー : Run/Power(Run)
  • -RUN2 ..... = メニュー : Run/Debug Run/No I Cache
  • -RUN3 ..... = メニュー : Run/Debug Run/I Cache
  • -RUN4 ..... = メニュー : Run/Debug Run/I Cache Compile
  • -SYNC ..... = メニュー : Run/Sync
  • -BLANK .... = メニュー : Run/Misc/Blank
  • -FULL ..... = メニュー : View/Full Screen/

アイコンにドロップ = XEBRA.EXE "PATH" = XEBRA.EXE -IMAGE "PATH" -RUN1

AutoHotKey によるホットキー追加

AutoHotkey を流行らせるページを見ると、AutoHotkey というアプリケーションで、スクリプトを使った高度なホットキー割り当てが記述できそうです。AutoHotkeyJp に日本語のリファレンスも充実しています。

下記、試しに擬似 Movie Snapshot の保存を行うホットキーを追加してみました。AutoHotkey.ahk の末尾に下記の記述を追加して保存し、XEBRA を起動後に AutoHotkey を起動すると、追加のホットキーが使えるようになっているはずです。

あくまで擬似的なものなので、動作に失敗することもありますが、何もないよりマシです。

また、XEBRA の挙動が変わればスクリプトも修正が必要になると思います。

  • 【重要】前提条件:予め CD イメージを読込済で、かつ Write Pad... で適当なパッド履歴を出力対象に指定済であること。
  • 作業ファイルはデスクトップに保存されますが、嫌ならスクリプト中の A_Desktop を変更すれば良いです。
  • Xebra_AutoUnpause を 0 にすると読み書き後の Pause 解除を行いません(F5 で手動解除することになります)……という感じだったのですが一長一短あるので、Backspace を独自の Pause に割り当てて、このキーが使用された際にスナップショット入出力後の Pause 解除の有無をトグル切り替えするようにしました。
  • ダイアログへのキー送信がうまくいかなかった場合、Pause 状態のはずなので落ち着いてダイアログを閉じて、もう一度保存しなおせば多分大丈夫です。多分……。
; XEBRA (PlayStation emulator)
#IfWinActive XEBRA
Xebra_AutoUnpause := 1

XebraLoadMovieSnapshot(TargetPath, AutoUnpause)
{
	; Pause
	;SendInput,!rp{Enter}
	SendInput,{F6}
	; File/Open/Running Image...
	SendInput,!for
	Sleep 400
	SendInput,%TargetPath%.RUN{Enter}
	Sleep 100
	; File/History/Load Pad...
	SendInput,!fhl
	Sleep 400
	SendInput,%TargetPath%.PAD{Enter}
	Sleep 100
	; Power(Run)
	if (AutoUnpause)
	{
		;SendInput,!r{Enter}
		SendInput,{F5}
	}
}

XebraSaveMovieSnapshot(TargetPath, AutoUnpause)
{
	; Pause
	SendInput,!rp{Enter}
	SendInput,{F6}
	; File/Save/Running Image...
	SendInput,!fsr
	Sleep 500
	SendInput,%TargetPath%.RUN{Enter}
	; File/History/Save Pad...
	SendInput,!fhs
	Sleep 400
	SendInput,%TargetPath%.PAD{Enter}
	Sleep 100
	; Power(Run)
	if (AutoUnpause)
	{
		;SendInput,!r{Enter}
		SendInput,{F5}
	}
}

; 1
1::
	BlockInput On
	XebraLoadMovieSnapshot(A_Desktop . "\XEBRA_001", Xebra_AutoUnpause)
	BlockInput Off
	return
; Shift+1
+1::
	BlockInput On
	XebraSaveMovieSnapshot(A_Desktop . "\XEBRA_001", Xebra_AutoUnpause)
	BlockInput Off
	return
; 2
2::
	BlockInput On
	XebraLoadMovieSnapshot(A_Desktop . "\XEBRA_002", Xebra_AutoUnpause)
	BlockInput Off
	return
; Shift+2
+2::
	BlockInput On
	XebraSaveMovieSnapshot(A_Desktop . "\XEBRA_002", Xebra_AutoUnpause)
	BlockInput Off
	return
; 3
3::
	BlockInput On
	XebraLoadMovieSnapshot(A_Desktop . "\XEBRA_003", Xebra_AutoUnpause)
	BlockInput Off
	return
; Shift+3
+3::
	BlockInput On
	XebraSaveMovieSnapshot(A_Desktop . "\XEBRA_003", Xebra_AutoUnpause)
	BlockInput Off
	return
; 4
4::
	BlockInput On
	XebraLoadMovieSnapshot(A_Desktop . "\XEBRA_004", Xebra_AutoUnpause)
	BlockInput Off
	return
; Shift+4
+4::
	BlockInput On
	XebraSaveMovieSnapshot(A_Desktop . "\XEBRA_004", Xebra_AutoUnpause)
	BlockInput Off
	return
; 5
5::
	BlockInput On
	XebraLoadMovieSnapshot(A_Desktop . "\XEBRA_005", Xebra_AutoUnpause)
	BlockInput Off
	return
; Shift+5
+5::
	BlockInput On
	XebraSaveMovieSnapshot(A_Desktop . "\XEBRA_005", Xebra_AutoUnpause)
	BlockInput Off
	return
; 6
6::
	BlockInput On
	XebraLoadMovieSnapshot(A_Desktop . "\XEBRA_006", Xebra_AutoUnpause)
	BlockInput Off
	return
; Shift+6
+6::
	BlockInput On
	XebraSaveMovieSnapshot(A_Desktop . "\XEBRA_006", Xebra_AutoUnpause)
	BlockInput Off
	return
; 7
7::
	BlockInput On
	XebraLoadMovieSnapshot(A_Desktop . "\XEBRA_007", Xebra_AutoUnpause)
	BlockInput Off
	return
; Shift+7
+7::
	BlockInput On
	XebraSaveMovieSnapshot(A_Desktop . "\XEBRA_007", Xebra_AutoUnpause)
	BlockInput Off
	return
; 8
8::
	BlockInput On
	XebraLoadMovieSnapshot(A_Desktop . "\XEBRA_008", Xebra_AutoUnpause)
	BlockInput Off
	return
; Shift+8
+8::
	BlockInput On
	XebraSaveMovieSnapshot(A_Desktop . "\XEBRA_008", Xebra_AutoUnpause)
	BlockInput Off
	return
; 9
9::
	BlockInput On
	XebraLoadMovieSnapshot(A_Desktop . "\XEBRA_009", Xebra_AutoUnpause)
	BlockInput Off
	return
; Shift+9
+9::
	BlockInput On
	XebraSaveMovieSnapshot(A_Desktop . "\XEBRA_009", Xebra_AutoUnpause)
	BlockInput Off
	return
; 0
0::
	BlockInput On
	XebraLoadMovieSnapshot(A_Desktop . "\XEBRA_000", Xebra_AutoUnpause)
	BlockInput Off
	return
; Shift+0
+0::
	BlockInput On
	XebraSaveMovieSnapshot(A_Desktop . "\XEBRA_000", Xebra_AutoUnpause)
	BlockInput Off
	return
; Backspace
BS::
	Xebra_AutoUnpause ^= 1
	if (Xebra_AutoUnpause)
	{
		; Power(Run)
		;SendInput,!r{Enter}
		SendInput,{F5}
	}
	else
	{
		; Pause
		;SendInput,!rp{Enter}
		SendInput,{F6}
	}
	return
#IfWinActive

TAS 的な機能について

小手先ではどうしても実用的に厳しいものがあります(操作性の面で)。当たり前ですが。

ただ、上述の通り rerecording できる仕組み自体は整っているようなので、技術を持った人であれば、本体を UPX で展開して、適当に解析して、適当にコード改変すれば、その手の方法にしては比較的簡単にまともな TAS 機能が付けられるのではないかと私は思います。*3

興味のある人はチャレンジしてみてはどうでしょうか?

それと、もう一つ違う方法を提案することができます。パッド履歴の中身はおそらく単純なキーコードで構成されているので、調べれば簡単に編集できると思います。根気のある人はバイナリエディタと再生確認の繰り返しだけで動画を作ることができるかもしれません。

*1:ただし、ARBEX は OS ROM を使わないので、OSROM を配置する必要はない。

*2:執筆時点では [0x454D78] + オフセット でメインメモリの内容をウォッチできましたが、バージョンアップされたら変わると思うので再度サーチが必要です

*3:パッド履歴の記録はフレームスキップとは無関係に行われますので、(後々出力先を応用することを考えても)パッド履歴出力周りを起点に解析するとやりやすいかもしれません。


          Strategi dan Trik promosi Bisnis di Internet   


IKLAN MINI

Gunakan Iklan Mini Secara Rutin
Sarana yang paling sering dan paling akrab digunakan oleh para pengiklan
adalah iklan mini. Di Internet Indonesia,... iklan mini meraih posisi nomor satu
dalam kemampuan fungsinya mempermudah promosi dan pemasaran produk dan
website kita. Apabila kita pandai memilih situs iklan gratis... akan lebih mudah bagi
kita untuk menerima hasil dari posting iklan yang kita lakukan beberapa waktu yang
lalu.
     
Pemasangan iklan di iklan mini Internet tidak lepas dari strategi untuk
membuat iklan yang menghasilkan respon dari pembaca.
Bagaimana Cara Pintar Memperoleh Manfaat Besar dari Pemasangan Iklan
Mini di Internet?
        
Walaupun iklan mini memiliki beberapa kelebihan dan kepraktisan, dia juga
memiliki satu kekurangan. Anda harus memasang ulang iklan anda pada situs-situs
gratis tersebut, kadangkala anda tidak bisa memasang iklan yang sama pada waktu
yang bersamaan. Anda tidak bisa memasang iklan dengan sekali kerja.... sepuluh
atau dua puluh iklan anda terpampang pada situs iklan gratis tersebut.
       
 Padahal... bisnis di Internet tidak lebih dari sekedar permainan angka saja.
Semakin banyak orang yang melihat iklan kita... akan semakin banyak orang
tersebut yang mengunjungi website kita. Hal ini berarti... semakin sering website
kita dilihat orang, maka akan semakin besar jumlah kemungkinan adanya pembeli
untuk produk yang kita tawarkan.
 
Berikut daftar situs iklan mini Internet yang memiliki pengunjung terbanyak :
 
www.bekas.com
www.iklanmini.co.id
www.iklaniklan.com
www.jual-beli.com
www.iklanbandung.com
www.iklanbaris.com
www.iklanbaris.co.id
www.globaliklan.most.cc
www.iklan88.virtualave.net
www.iklanumum.com
www.jubelindo.co.id
www.iklan-gratis.cjb.net
www.iklankita.com
www.iklanpr.com
www.iklanbaris.pameranoke.com
www.bursaiklan.com
www.adhoo.i-p.com

www.e-klan.net
www.iklan-25.co.id
www.iklansolo.com
www.iklanaja.com
www.i-klik.com
www.iklaneka.com
www.iklan.situsmedia.com
www.jualbeli.com
www.koinku.com
www.iklanlaris.com
www.mediaiklan.com
www.iklanet.com
www.iklanmini.kemana.com
www.iklanpojok.com
www.gadogado.com
www.iklanmini.com
www.iklaninternet.com

Ini berarti, anda harus menampilkan promosi anda ke sebanyak mungkin
sarana iklan.
       
Di sini anda akan mempelajari bagaimana kita bisa berpikir sedikit lebih
pintar untuk mencermati setiap persyaratan yang diberikan oleh suatu iklan mini
berkaitan dengan jumlah tampilan kita.
      
Ada beberapa aturan secara terurut yang harus anda penuhi untuk
meningkatkan efektifitas iklan anda berkali-kali lipat.
 
1. Selalu berusaha mengotomatiskan pemasangan iklan anda.
 
Apa maksudnya? Anda pasti memasang iklan secara berulang-ulang pada
beberapa kegiatan yang sama. Anda juga lakukan pemasangan tersebut secara
terus menerus, misalnya: mengisi form isian dengan nama, alamat, email,
alamat URL, judul iklan, naskah iklan anda, password dan sebagainya....
 
Jika dihitung, kegiatan ini akan menghabiskan waktu anda sekitar 5 sampai 10
menit untuk setiap kegiatan pengisian form posting iklan yang dilakukan. Jika
anda hanya memasang iklan sebanyak satu kali selama satu minggu dan itupun
hanya satu iklan... tentu tidak akan menjadi masalah bagi anda. Tapi apa yang
bisa anda harapkan hanya dari satu iklan?
 
Dilemanya, jika anda memasang sebanyak mungkin iklan, berarti anda harus
melakukan kegiatan mengetik secara berulang-ulang.
 
Apa yang harus anda lakukan? Manfaatkan software khusus. Yaitu TypeIt in dan
Ad-Wizard.
Kedua software ini akan sangat membantu pekerjaan anda.

TypeIt in
TypeIt In memiliki fungsi menyimpan data isian anda seperti nama, alamat,
email, iklan, naskah iklan dalam bentuk tombol.
 
Jika anda berniat mengisi form nama, silakan anda klik tombol nama... jadilah
form isian nama anda. Jika anda ingin mengisi form naskah iklan, klik tombol
naskah iklan... jadilah naskah iklan anda terisi pada form tersebut.
 
Saya selalu menggunakan software ini untuk mengisi form-form panjang dan
menjemukan yang ada di hadapan kita. Pekerjaan saya lebih cepat 450% dengan
software ini.
 
TypeIt in dapat anda download di http://www.wavget.com/dltypeitin.html


Software kedua adalah... Ad-Wizard
 
Software ini memiliki kemampuan yang sama dengan TypeIt in. Hanya saja dia
khusus digunakan untuk mengisi satu halaman form sekaligus. Sepuluh kali lebih
cepat dari pada TypeIt in.
 
Misalnya anda menjumpai satu halaman yang berisi 50 box isian mengenai data
tertentu yang harus diisi dan anda memiliki kegiatan rutin untuk mengisinya. Jika
anda mengisi secara manual, akan membutuhkan sekitar 30 menit untuk
melengkapi form tersebut... dan itupun harus mengulangnya jika suatu saat anda
ingin mengisinya lagi.
 
Jika anda menggunakan Type It in, anda hanya cukup merancangnya dalam
tombol-tombol khusus, dan setiap box yang harus anda isi cukup anda lakukan
dengan meng-klik tombol bersangkutan. Apabila suatu saat anda harus mengisi
lagi form tersebut, anda tinggal menggunakan tombol-tombol yang sudah anda
rancang sebelumnya. Sampai kapanpun anda lakukan... Anda tidak perlu
mengetik ulang lagi!
 
Ad Wizard mampu berbuat jauh lebih baik lagi. Dari 50 box pada form tersebut,
anda rancang terlebih dahulu daftar isian 50 box. Ini membutuhkan waktu sekitar
15-30 menit. Setelah anda selesai merancang AdWizard, dengan sekali klik...
seluruh 50 box tersebut akan terisi.
 
Apabila suatu saat anda harus mengisi ulang form (50 Box) tersebut, anda cukup
menggunakan AdWizard rancangan anda. Dengan sekali klik saja... seluruh box
akan penuh terisi dengan data anda. Hanya memerlukan waktu dua detik untuk
melakukan ini. Anda tidak perlu melakukan pengetikan ulang... satu huruf pun!!
 
Bayangkan... berapa banyak iklan mini yang bisa anda isi dalam satu jam jika
anda menggunakan alat ini.
 
Tata cara pemakaian kedua software tersebut bisa anda ketahui pada bagian help
masing-masing software.
 
Ad Wizard dapat anda download di http://www.wavget.com/dladwizard.html

2. Selalu berusaha meningkatkan nilai iklan anda.
Apa maksudnya? Kalau dalam bahasa Internet disebut dengan Ads Convertion
Rate. Yaitu ratio perbandingan dari jumlah orang yang melihat iklan anda
dengan keputusan dia untuk meng-klik iklan anda.
 
Misalnya CR 10% berarti dari 100 orang yang membaca iklan anda, ada 10 orang
yang mengklik iklan untuk mengetahui informasi anda lebih jauh. Semakin besar
rasio iklan anda, maka akan semakin besar jumlah pengunjung yang akan anda
miliki.
 
Prinsip dasar penulisan iklan ini sama dengan prinsip yang digunakan dalam
penulisan headline sales letter. Silakan pelajari di bab tersebut untuk
meningkatkan nilai iklan anda. Anda akan menemukan strategi dasar untuk
membuat setiap iklan yang anda tulis mampu menghasilkan respon yang
memuaskan.
 
3. Selalu berusaha meningkatkan nilai waktu anda.
Lakukan pemasangan iklan berulang-ulang setiap hari dengan menggunakan
waktu yang teratur, setiap waktunya memasang iklan gunakan sesilisih satu atau
dua jam perhari untuk memasang kembali iklan anda.
 
Mengapa begitu?
Agar tampilan iklan anda sesering mungkin setiap saat berada pada urutan
pertama!
 
Misalnya:
Pada pukul 07.00 anda memasang 3 iklan dengan kategori/kelompok yang
berbeda dan pasang lagi pada jam berikutnya pukul 09.00 atau 11.00 (selisih 2
atau 3 jam)! Lakukan ini dengan cara melakukan pemasangan iklan 6 s/d 10 kali
waktu yang berbeda dalam sehari.
 
Gunakan teknik di atas untuk mempromosikan bisnis anda. Hanya membutuhkan
waktu beberapa menit untuk setiap masing-masing pemasangan iklan setiap
hari.
 
Bagaimana caranya? Gunakan teknik strategi pemasangan iklan di atas. Pakai
softwarenya. Kemauan anda untuk sedikit lebih bekerja lagi akan menghasilkan
imbalan yang sangat besar di kemudian hari.
 
Catatan: Software pengirim iklan baris otomatis dengan sekali klik yang dapat
mengirim ke puluhan iklan baris di Indonesia bisa juga anda dapatkan di
http://www.promosi.cintasejati.com
 
Yang paling penting dalam pemasangan iklan di Iklan Mini Internet adalah:
JANGAN PERNAH MELAKUKAN SPAM!!!

Strategi dan Trik promosi Bisnis di Internet Artikel Tentang Iklan Mini
Thanks sudah membaca dan Mudah Mudahan informasi nya bisa membantu anda.





          Waverley   
Waverley
author: Walter Scott
name: Jamie
average rating: 3.48
book published: 1814
rating: 3
read at: 2013/12/02
date added: 2013/12/02
shelves:
review:


          Episode 6: PSR-X and the Mexican Standoff   

PHP-FIG member Paul M. Jones and PHP contributor Anthony Ferrera come on the podcast with Ben, Phil and regular guest Zack Kitzmiller to discuss the new Package Orientated Autoloader Proposal (a.k.a PSR-X), and whether or not PSR’s should ever be amended.

Paul explains why the new PSR-X is a handy idea for those who are already PSR-0 users, or even those that aren’t, Zack flips a shit and gets bored of talking about standards wishing everyone could just use Python and their awesome PEP-8, we convince Paul to try swearing for once and we all try badgering Anthony into agreeing with us for some reason.

Nobody wins, but the argument brings up a lot of interesting topics and points of view, and that is mostly what we are here for.


              
Don't forget your hair extensions G.Wave (GEEwig) on your wedding day 👰🏻
          Special Ecstatic Dance Event - Liz Temple comes to Spokane!   
Hello, Dancers!

Since my first experience of dancing ecstatically in community in Hawaii a few years ago, I have been exploring various aspects of "movement as spiritual practice."  As my experience continues to deepen, I've discovered  tremendous wisdom and teaching in Gabrielle Roth's  5Rhythms map for ecstatic dance.  With its beginnings over twenty years ago, Gabrielle's dynamic work now moves globally in an rapidly growing, interconnected community of practice.  I first encountered that community at Burning Man last year, and then again in Olympia this past New Year's at Waves Studio .  Inspired by the energy of their community, I committed to bringing a connection to our fledgling community here, and began a conversation with Liz Temple, the owner and home teacher at Waves.  On Sunday, May 3rd, Liz will be joining us at Harmony Yoga for a special Sunday morning Sweat Your Prayers dance, followed by an afternoon session for those who want to experience 5Rhythms and move deeper into the potential of the work.  Liz is an exceptional teacher and DJ, and so I invite you and your friends to what I believe will be a heart-opening, joyous experience.  

Please email me at ian.daddio@gmail.com or call (509.990.6434 or 509.747.2993), if you would like to register for the afternoon session, or would like further information.   You can do either session or both.  The cost for Sweat Your Prayers is $15 and for Deep Waves $45.  There will be a $10 discount for those participating in both the morning and afternoon dance sessions.

Descriptions: 
Sweat Your Prayers
Sunday 10:30am -1 pm This class is drop in $15
This uplifting Sunday morning dance is the perfect antidote for our overburdened lives. This is a time to come together in community and do the practice of the 5Rhythms Wave. In Sweat Your Prayers, you are free to follow your own dance as we move through the rhythms of flowing, staccato, chaos, lyrical and stillness. After a warm-up dance we move through an uninterrupted wave of the 5Rhythms, freeing your body and opening your heart. Each person is free to dance alone or with others.
Deep Waves
Sunday 2:00pm - 5:00pm This class is $45, Registration requested
The 5Rhythms® are a dynamic map of the way energy moves. Just like light, sound, or ocean waves, a dancing body when moving freely passes through 5 distinct rhythmic patterns. We navigate this map by putting ourselves into motion, finding the shape of our bodies in each new moment, and releasing into our creative spirit. In this special day of in-depth discovery, we play with the elements of the map and discover new ways of being in motion.

          Newly Energized and Creatively Inspired   
After a single posting back in the beginning, I'm going to get this party going again! Since starting Dance Night, we've been going strong with monthly dances, now being held at Harmony Yoga. Last August, I undertook the fire initiation that is Burning Man, and discovered 5Rhythms at the RhythmWave camp, an incredible experience of what a dance community can create. Over New Years, I traveled over the mountains to Olympia, WA and danced with Amara and Sara Pagano at Waves Studio, my first 5Rhythms workshop called "Patterns." Last week, I returned to Waves for another workshop with Kate Shela, a gifted teacher who weaves the shamanic journey with 5Rhythms. In the coming months, I hope to bring 5Rhythms to Spokane, WA. In addition to our once-monthly Dance Night, we will hold a Sunday morning "Dance Your Prayers."

Let's see how this experiment flows.

"Holiness should be wholeness, and wholeness is always a dancing state, a state where the dancer disappears into the dance...Dance your way to God. Dance till the stars come down from the rafters. Dance creates the fire in which all that is useless is burned, only the essential remains. Dance creates the fire in which gold becomes purer and purer and purer, and ultimately only purity is left." - Osho
          PCDJ DEX 3.8.0.1   

PCDJ DEX to potężne narzędzie do miksowania muzyki, przeznaczone jednocześnie dla początkujących jak i zaawansowanych użytkowników. Aplikacja obsługuje najpopularniejsze formaty plików audio: *.MP3, *.OGG, *.WMA, *.FLAC, *.WAV. Program umożliwia obsługę wtyczki VST. Ponadto narzędzie pozwala na zapis w czasie rzeczywistym granego setu do formatów *.MP3, *.WAV, *.OGG.

PCDJ DEX to aplikacja idealnie nadająca się do występów na żywo. Dużą zaletą jest możliwość współpracy z urządzeniami zewnętrznymi takimi jak: PCDJ DAC-3, DENON DN-HC4500, Behringer BCD-2000, Hercules DJ Console, Vestax VCI-100, EKS XP10, FinalScratch, VirtualDJ, SSL, MsPinky, oraz ze wszelkimi kontrolerami MIDI.
Uwaga:

Dostępna jest równiez darmowa wersja programu na naszej stronie.

 


          Musicator 1.0.0.2   
Программа для проигрывания .MPJ, воспроизведения .WAV, .MID-файлов.
          Comment on Opeth – Deliverance [Bruce Soord Vinyl Remix] – Review by Yalcin Inel   
While I don't own a vinyl setup to experience this, I can say that the remix on CD is leaps and bounds better than the original. The LP-rips of the original scattered around the net are still better than the remix cd tho'. Anyway, everyone who owns the dvd's of the remix can produce a little miracle in downmixing the surround mix to stereo with sox.exe commands. You need to rip the dts stream from the dvd with a program such as dvd-decrypter, convert it to at least 24 bit wav and with a little experimentation on sox.exe you can get the most dynamic, lush and detailed sound of the album which will create eargasm after eargasm. I guess this is right now the best way to experience what Lord AMG described here, if you don't own a vinyl player. If anyone is keen on trying this, my version was created using the command: Sox.exe -V -G deliverance.wav c:outputdeliverance.wav remix 1v0.39,3v0.06,4v0.52,5v0.34 2v0.39,3v0.06,4v0.52,6v0.34 norm -1
          #2325: MP4 AAC Audio is delayed by 2ms when converted to PCM   

Summary of the bug:

When using ffmpeg to convert an aac audio stream from an mp4 to pcm, the result is out of sync by about 2ms. By adding -ss 00:00:00.02 after the input, then output is correctly aligned.

How to reproduce:

% ffmpeg -i test100.mp4 -c:a pcm_s16le test100_audio.wav
ffmpeg version 1.1.git
git revision: faa0068
built on Mar  4 2013 11:40:27

          #5866: mxf regression with xiph sample file   

Lutz Vieweg reported an mxf regression since bf2290a80c124c5168f78be64ee87d9efdd4040d for the following file: https://media.xiph.org/video/derf/meridian/MERIDIAN_SHR_C_EN-XX_US-NR_51_LTRT_UHD_20160909_OV/MERIDIAN_SHR_C_EN-XX_US-NR_51_LTRT_UHD_20160909_OV_01_EN_20_B.mxf

$ ffmpeg -i MERIDIAN_SHR_C_EN-XX_US-NR_51_LTRT_UHD_20160909_OV_01_EN_20_B.mxf out.wav
ffmpeg version N-81774-g64545dd Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      55. 30.100 / 55. 30.100
  libavcodec     57. 58.100 / 57. 58.100
  libavformat    57. 50.100 / 57. 50.100
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 63.100 /  6. 63.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
[mxf @ 0x293c3a0] OPAtom misinterpreted as OP1a? KLV for edit unit 0 extending into next edit unit is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[mxf @ 0x293c3a0] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, mxf, from 'MERIDIAN_SHR_C_EN-XX_US-NR_51_LTRT_UHD_20160909_OV_01_EN_20_B.mxf':
  Metadata:
    application_platform: Microsoft Windows 7 Professional Service Pack 1 (Build 7601)
    uid             : cdb57bb1-4f06-4c77-64e4-c0293f2c7163
    generation_uid  : 2d654592-04ae-4cf4-35ab-69707dc0de3a
    company_name    : Colorfront
    product_name    : Transkoder
    product_uid     : 3a4fe380-0d01-11e4-869f-3cd92b5c1dfc
    product_version : 2.7.3.20150121
    modification_date: 2016-09-09T11:29:38.000000Z
    material_package_umid: 0x060A2B340101010501010F20130000002C3E49A0BD1548ACD65D70EA10FAEB6B
    timecode        : 00:00:17;12
  Duration: 00:11:58.92, start: 0.000000, bitrate: 2304 kb/s
    Stream #0:0: Audio: pcm_s24le, 48000 Hz, stereo, s32 (24 bit), 2304 kb/s
    Metadata:
      file_package_umid: 0x060A2B340101010501010F2013000000AE2179BC93E24600C47112668D9E21FB
Output #0, wav, to 'out.wav':
  Metadata:
    application_platform: Microsoft Windows 7 Professional Service Pack 1 (Build 7601)
    uid             : cdb57bb1-4f06-4c77-64e4-c0293f2c7163
    generation_uid  : 2d654592-04ae-4cf4-35ab-69707dc0de3a
    company_name    : Colorfront
    product_name    : Transkoder
    product_uid     : 3a4fe380-0d01-11e4-869f-3cd92b5c1dfc
    product_version : 2.7.3.20150121
    modification_date: 2016-09-09T11:29:38.000000Z
    material_package_umid: 0x060A2B340101010501010F20130000002C3E49A0BD1548ACD65D70EA10FAEB6B
    ISMP            : 00:00:17;12
    ISFT            : Lavf57.50.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s
    Metadata:
      file_package_umid: 0x060A2B340101010501010F2013000000AE2179BC93E24600C47112668D9E21FB
      encoder         : Lavc57.58.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s24le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=       0kB time=00:00:00.00 bitrate=N/A speed=   0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)

          #1670: -filter_complex doesn't handle well finite video + infinite audio   

Given a 60 seconds video stream:

./ffmpeg -v 0 -f lavfi -i testsrc=d=60 -y video.mp4

and a 5 seconds audio stream:

./ffmpeg -v 0 -f lavfi -i 'aevalsrc=sin(440*2*PI*t)::d=5' -y audio.wav

We can make a video presentation of 60 seconds with the 5 seconds audio padded with silence with the following command line:

./ffmpeg -f lavfi -i 'movie=video.mp4 [out0]; amovie=audio.wav [audio]; aevalsrc=0 [silence]; [audio][silence] concat=v=0:a=1 [out1]' -y out.avi

Unfortunately, this doesn't work with the -filter_complex option:

% ./ffmpeg -i video.mp4 -i audio.wav -filter_complex 'aevalsrc=0 [silence]; [1:0][silence] concat=v=0:a=1 [padded_audio]' -map 0:0 -map '[padded_audio]' -y out.avi
ffmpeg version N-43791-g2c7a622 Copyright (c) 2000-2012 the FFmpeg developers
  built on Aug 21 2012 21:01:23 with gcc 4.7.1 (GCC) 20120721 (prerelease)
  configuration: --enable-gpl --enable-fontconfig --enable-libfreetype --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-libx264 --enable-libvpx --enable-libtheora --enable-x11grab --enable-libopenjpeg --enable-libass --enable-libmodplug --enable-libv4l2 --cc='ccache colorgcc' --samples=/home/ubitux/fate-samples --prefix=/tmp/ffinstall
  libavutil      51. 70.100 / 51. 70.100
  libavcodec     54. 54.100 / 54. 54.100
  libavformat    54. 25.104 / 54. 25.104
  libavdevice    54.  2.100 / 54.  2.100
  libavfilter     3. 13.100 /  3. 13.100
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf54.25.104
  Duration: 00:01:00.00, start: 0.000000, bitrate: 37 kb/s
    Stream #0:0(und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv444p, 320x240 [SAR 1:1 DAR 4:3], 34 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc
    Metadata:
      handler_name    : VideoHandler
Guessed Channel Layout for  Input Stream #1.0 : mono
Input #1, wav, from 'audio.wav':
  Duration: 00:00:05.01, bitrate: 705 kb/s
    Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
Output #0, avi, to 'out.avi':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    ISFT            : Lavf54.25.104
    Stream #0:0(und): Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 320x240 [SAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 25 tbn, 25 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, mono, s16
Stream mapping:
  Stream #1:0 (pcm_s16le) -> concat:in0:a0 (graph 0)
  Stream #0:0 -> #0:0 (h264 -> mpeg4)
  concat (graph 0) -> Stream #0:1 (libmp3lame)
Press [q] to stop, [?] for help
frame= 1500 fps=177 q=9.6 Lsize=   13088kB time=00:01:00.00 bitrate=1786.9kbits/s

This creates some kind of infinite loop, with fps slightly decreasing over time (+memory leak)


          #2120: mmf files generated by ffmpeg does not play with yamaha software player   

Currently FFmpeg creates mmf files which are uplayable by yamaha software.

"ma2_mono_8khz.mmf" is a file created with yamaham software. What I can notice comparing this file and output from FFmpeg is that yamaha file have additional AspI chunk, so maybe the problem is that FFmpeg doesn not write it.

Mobile Contents Player (MCP-MA7) Ver.1.6.2

http://smaf-yamaha.com/tools/downloads.html

C:\>ffmpeg -i test.wav -ac 1 -ar 8000 ffmpeg_ma2_mono_8khz.mmf
ffmpeg version N-48610-gb23aff6 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jan  8 2013 02:37:29 with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.
0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-w32threads --enable-memalign-hack --enable-runtime-cpudetect
--enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -
lwinmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snap
shots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/s
napshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3
--enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-
libvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-
libopencore-amrwb --enable-libmp3lame --enable-libvpx --disable-decoder=libvpx
  libavutil      52. 13.100 / 52. 13.100
  libavcodec     54. 86.100 / 54. 86.100
  libavformat    54. 59.106 / 54. 59.106
  libavdevice    54.  3.102 / 54.  3.102
  libavfilter     3. 32.100 /  3. 32.100
  libswscale      2.  1.103 /  2.  1.103
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[wav @ 0x1efea60] max_analyze_duration 5000000 reached at 5015510
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, wav, from 'test.wav':
  Duration: 00:00:30.01, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16
, 1411 kb/s
Output #0, mmf, to 'ffmpeg_ma2_mono_8khz.mmf':
  Metadata:
    encoder         : Lavf54.59.106
    Stream #0:0: Audio: adpcm_yamaha, 8000 Hz, mono, s16, 32 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> adpcm_yamaha)
Press [q] to stop, [?] for help
size=     118kB time=00:00:30.20 bitrate=  32.0kbits/s
video:0kB audio:118kB subtitle:0 global headers:0kB muxing overhead 0.076139%

          #2283: Slow counter / incorrect duration for aac-in-flv   

(videolan ticket 8195) I will upload an audio-only flv sample that can be decoded but the shown duration is wrong, both the duration shown before decoding and the duration shown after decoding. The time counter is wrong for both ffplay and MPlayer, seeking works fine with MPlayer, fails with ffplay.

$ ffmpeg -i sky_the80s_aacplus.flv out.wav
ffmpeg version N-50086-gd41efc1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Feb 19 2013 09:36:55 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 17.102 / 52. 17.102
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.100 / 54. 63.100
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 38.103 /  3. 38.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[flv @ 0x32bdfc0] max_analyze_duration 5000000 reached at 5014000 microseconds
Input #0, flv, from 'sky_the80s_aacplus.flv':
  Duration: 00:01:08.77, start: 0.000000, bitrate: 74 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf54.63.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac -> pcm_s16le)
Press [q] to stop, [?] for help
[flv @ 0x32bdfc0] Stream discovered after head already parsed
[flv @ 0x32bdfc0] New data stream 0:1 at pos:254270 and DTS:27.274s
size=   21168kB time=00:01:08.81 bitrate=2519.9kbits/s
video:0kB audio:21168kB subtitle:0 global headers:0kB muxing overhead 0.000369%

Duration and time claim that the sample is ~68 seconds, but it actually is over two minutes long:

$ ffmpeg -i out.wav -f null -
ffmpeg version N-50086-gd41efc1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Feb 19 2013 09:36:55 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 17.102 / 52. 17.102
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.100 / 54. 63.100
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 38.103 /  3. 38.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[wav @ 0x1c49fc0] max_analyze_duration 5000000 reached at 5015510 microseconds
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, wav, from 'out.wav':
  Metadata:
    encoder         : Lavf54.63.100
  Duration: 00:02:02.88, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Output #0, null, to 'pipe:':
  Metadata:
    encoder         : Lavf54.63.100
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)
Press [q] to stop, [?] for help
size=N/A time=00:02:02.88 bitrate=N/A
video:0kB audio:21168kB subtitle:0 global headers:0kB muxing overhead -100.000101%

The same can be shown with ffplay:

$ time ffplay sky_the80s_aacplus.flv
ffplay version N-50086-gd41efc1 Copyright (c) 2003-2013 the FFmpeg developers
  built on Feb 19 2013 09:36:55 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 17.102 / 52. 17.102
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.100 / 54. 63.100
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 38.103 /  3. 38.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[flv @ 0x7f7d6c0008c0] max_analyze_duration 5000000 reached at 5014000 microseconds
Input #0, flv, from 'sky_the80s_aacplus.flv':
  Duration: 00:01:08.77, start: 0.000000, bitrate: 74 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp
  23.20 A-V:  0.000 fd=   0 aq=    1KB vq=    0KB sq=    0B f=0/0
real    0m43.481s
user    0m0.937s
sys     0m0.474s

          #2504: Audio glitches and distortion when recording alsa   

Summary of the bug: I try to record from integrated audio card input line/micro and get distortions. How to reproduce:

$ ffmpeg -debug 1 -vn -f alsa -ac 2 -channel_layout stereo -i hw:0,0 -c:a pcm_s16le /Store3/Test/pcm_`date +%m%d_%H%M`.wav
ffmpeg version N-52061-g8137ea3 Copyright (c) 2000-2013 the FFmpeg developers
  built on Apr 13 2013 21:10:51 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3)
  configuration: --prefix=/opt/ffmpeg-opt/usr --enable-shared --bindir=/opt/ffmpeg-opt/usr/bin --datadir=/opt/ffmpeg-opt/usr/share/ffmpeg-opt --incdir=/opt/ffmpeg-opt/usr/include/ffmpeg-opt --libdir=/opt/ffmpeg-opt/usr/lib64 --mandir=/opt/ffmpeg-opt/usr/share/man --shlibdir=/opt/ffmpeg-opt/usr/lib64 --arch=x86_64 --optflags='-O2 -g' --enable-bzlib --disable-crystalhd --enable-gnutls --enable-libass --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --disable-libv4l2 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --build-suffix=1 --disable-doc --enable-runtime-cpudetect
  libavutil      52. 26.100 / 52. 26.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  2.100 / 55.  2.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 53.101 /  3. 53.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
 matched as AVOption 'debug' with argument '1'.
Reading option '-vn' ... matched as option 'vn' (disable video) with argument '1'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'alsa'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'.
Reading option '-channel_layout' ... matched as option 'channel_layout' (set channel layout) with argument 'stereo'.
Reading option '-i' ... matched as input file with argument 'hw:0,0'.
Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'pcm_s16le'.
Reading option '/Store3/Test/pcm_0426_1018.wav' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Successfully parsed a group of options.
Parsing a group of options: input file hw:0,0.
Applying option vn (disable video) with argument 1.
Applying option f (force format) with argument alsa.
Applying option ac (set number of audio channels) with argument 2.
Applying option channel_layout (set channel layout) with argument stereo.
Successfully parsed a group of options.
Opening an input file: hw:0,0.
[alsa @ 0x20f7720] All info found
Input #0, alsa, from 'hw:0,0':
  Duration: N/A, start: 1366967936.157276, bitrate: 1536 kb/s
    Stream #0:0, 1, 1/1000000: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Successfully opened the file.
Parsing a group of options: output file /Store3/Test/pcm_0426_1018.wav.
Applying option c:a (codec name) with argument pcm_s16le.
Successfully parsed a group of options.
Opening an output file: /Store3/Test/pcm_0426_1018.wav.
Successfully opened the file.
[graph 0 input from stream 0:0 @ 0x20e8280] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0x20e8280] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0x20e8280] Setting 'sample_fmt' to value 's16'
[graph 0 input from stream 0:0 @ 0x20e8280] Setting 'channel_layout' to value '0x3'
[graph 0 input from stream 0:0 @ 0x20e8280] tb:1/48000 samplefmt:s16 samplerate:48000 chlayout:0x3
[audio format for output stream 0:0 @ 0x20effc0] Setting 'sample_fmts' to value 's16'
[AVFilterGraph @ 0x20f7320] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed
Output #0, wav, to '/Store3/Test/pcm_0426_1018.wav':
  Metadata:
    ISFT            : Lavf55.2.100
    Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)
Press [q] to stop, [?] for help
size=    1076kB time=00:00:05.73 bitrate=1536.1kbits/s
video:0kB audio:1076kB subtitle:0 global headers:0kB muxing overhead 0.007263%
31021 frames successfully decoded, 0 decoding errors
Statistics: 4 seeks, 31024 writeouts

Result: very bad audio, distortion Same result with

$ ffmpeg -debug 1 -vn -f alsa -ac 2 -channel_layout stereo -i hw:0,0 -ar 44100 -ab 320k -c:a mp3 /Store3/Test/mp3_`date +%m%d_%H%M`.mp3

Result: very bad audio, distortion

$ ffplay -debug 1 -f alsa -ac 2 -i hw:0,0

Result: statics after almost every bip.

More tests with other applications: Audacity With Alsa, input: default then pulse then ALC892 (hw:0,0) Result: with these 3 different parameters, audio is perfect.

recordmydesktop input: DEFAULT result: audio is perfect.

gnome-sound-recorder 2.29.91 Record from input: Capture format flac, ogg, pm3 Result: with these 3 different formats, audio is perfect.

arecord arecord -f cd -t wav out.wav Result: audio is perfect.

VLC input hw:0,0 Result: static after every bip.


          #3433: using "shortest" results in mismatched audio and video lengths   

Summary of the bug:

"-shortest" does not add trim/atrim filters, so the length of each output stream will be a multiple of sample size. In the given example, the video is exactly 1s, but the resulting output is 1.02s. If "-t 1.000" is passed instead of "-shortest", the resulting output is exactly 1s.

The difference is that using "-t" adds trim/atrim filters, so the last frame->nb_samples encoded will be truncated to fit the given duration.

How to reproduce:

% ffmpeg -y -probesize 500000 -f image2 -r 25 -i colorbars.%04d.png -i 100Hz_44100Hz_16bit_05sec.wav -shortest -c:v png -c:a pcm_s16le -q:v 1 -2 -f mov colorbars.mov
Note that input streams have durations of 5s and 1s.
% ffprobe colorbars.mov
Note that the output stream has a duration of 1.02s.
ffmpeg version N-60960-gf3eef02
built on Feb 28 2014 10:31:20

          #3455: aresample=matrix_encoding=dplii doesn't downmix backleft and center correctly   

Summary of the bug: DPLII files have too-loud right channel. Seems to be some problem with the interaction of back left and center channel How to reproduce:

ffmpeg -i surround_blbrc.wav -af "aresample=matrix_encoding=dplii" -ac 2 dplii_blbrc.wav
ffmpeg version git-2014-03-09-8dfd0e6
built on Mar  9 2014 22:53:26 with gcc 4.8.2 (Gentoo 4.8.2 p1.3r1, pie-0.5.8r1)

You can see the problem easiest by crafting a 5.1 surround wave file with empty channels except the center channel with noise and the two back channels, first half of the back left with noise and the second half of the back right with noise. However, all surround files that use the bl and center channels at the same time will be too loud in the resulting file's right channel.

Can provide example files if you like.


          #3853: silence in front of generated audio mpegts   

Let a.ts be an mpegts with aac encoded audio only. Then

/ffmpeg/ffmpeg -i a.ts -acodec aac -strict -2 -loglevel debug audio.ts

Input #0, aac, from 'a.ts': Duration: 00:00:05.06, bitrate: 126 kb/s Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 126 kb/s

if I convert a.ts and audio.ts into .wav I will get two wav files, one of them has silence in front of it. I used audacity to determine that about 0.031s audio.ts will have some silence. I can see that a.ts has 215 audio frames and audio.ts 216.

Any explanations why is silence added to ts? As far as I can see it could be a bug. Summary: the command at top introduces one extra frame to the start of aac stream, making aac longer as it should.


          #3887: ffmpeg cant multiplex IMX video essence (m2v) with correct vbv into mov or mxf   

Summary of the bug: How to reproduce:

% ffmpeg -i <file>  -vcodec mpeg2video -pix_fmt yuv422p -b:v 30000k -maxrate 30000k -minrate 30000k -g 1 -flags +ildct+low_delay -dc 10 -intra_vlc 1 -non_linear_quant 1 -qmin 1 -qmax 8 -q:v 1 -ps 1 -top 1 -aspect 16:9 -bufsize 9437184 -rc_init_occupancy 9437184 -rc_buf_aggressivity 0.25 -an <intermediate.m2v>
ffmpeg -i <intermediate.m2v> -i stereo.wav -vcodec copy -r 25 -acodec copy -vtag mx3p <final.mov>
ffmpeg -i <intermediate.m2v> -i stereo.wav -vcodec copy -r 25 -acodec copy -d10_channelcount 4 -f mxf_d10 <final.mxf>
mov file shows complete green field (like video error). mxf file shows "[mxf_d10 @ 04714260] cannot fill d-10 video packet" errors during mpxing, and final video has many video errors.
vbv = 9437184 is correct value for MPEG2 422P@ML, and e.g. Telestream encoder will correctly generate mxf at least with this vbv.
Apple's own IMX.mov encoding shows vbv 1835008, which is actually MPEG2 MP@ML vbv, meant for max bitrate of 15Mbit/sec
Maximum vbv before error appears to be 1200000 for IMX30 and 2000000 for IMX50, assuming -rc_init_occupancy is set to same value.
ffmpeg version git 1aa153d
built on … Aug 23rd 2014

Thanks you very much for looking at this.


          #4160: problem encoding opus file with unusual channel mask   

Summary of the bug: trying to encode opus audio file with unusual channel mask (for example FL+FR+BL) generates an error. How to reproduce: ffmpeg -i 3flfrbl.wav 3flfrbl.opus

% ffmpeg -i 3flfrbl.wav 3flfrbl.opus
ffmpeg version 2.5
built on December 3rd 2014

          #4692: opus + webm - inconsistent meta?   
ffmpeg -i src.wav -ar 48000 -af atrim=start_sample=24000*0:end_sample=240000*1 -c:a opus 0.webm

I'm a bit confused by ffmpeg while encoding opus into webm. I have a wav file with exactly 240000 samples @ 48kHz. The resulting file gives me the following information:

Duration: 5.01, Start 0.007 discard 648/900 samples 240312 samples decoded.

If I have several of these files how would I play them seamlessly without gaps?

i.e. in a browser I've tried:

sourceBuffer.timestampOffset = 5 * n - 0.007;
sourceBuffer.appendWindowStart = 5 * n;
sourceBuffer.appendWindowEnd = 5 * (n+1);
sourceBuffer.appendBuffer(new Uint8Array(buffer[n]));

However, there are audible gaps.

How many samples am I actually supposed to discard? 0.007 * 48000, 648, or 240312 - 240000?

ffprobe -i 0.webm
ffprobe version 2.6.3 Copyright (c) 2007-2015 the FFmpeg developers
  built with Apple LLVM version 6.1.0 (clang-602.0.53) (based on LLVM 3.6.0svn)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/2.6.3 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc --enable-libxvid --enable-libtheora --enable-libvorbis --enable-libvpx --enable-librtmp --enable-libass --enable-ffplay --enable-libfdk-aac --enable-openssl --enable-libopus --enable-libwebp --enable-nonfree --enable-vda
  libavutil      54. 20.100 / 54. 20.100
  libavcodec     56. 26.100 / 56. 26.100
  libavformat    56. 25.101 / 56. 25.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 11.102 /  5. 11.102
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Input #0, matroska,webm, from '0.webm':
  Metadata:
    encoder         : Lavf56.25.101
  Duration: 00:00:05.01, start: 0.007000, bitrate: 94 kb/s
    Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp (default)
h59ec2765:sintel ronagy$ ffmpeg -loglevel debug -i 1.webm asd.wav
ffmpeg version 2.6.3 Copyright (c) 2000-2015 the FFmpeg developers
  built with Apple LLVM version 6.1.0 (clang-602.0.53) (based on LLVM 3.6.0svn)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/2.6.3 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc --enable-libxvid --enable-libtheora --enable-libvorbis --enable-libvpx --enable-librtmp --enable-libass --enable-ffplay --enable-libfdk-aac --enable-openssl --enable-libopus --enable-libwebp --enable-nonfree --enable-vda
  libavutil      54. 20.100 / 54. 20.100
  libavcodec     56. 26.100 / 56. 26.100
  libavformat    56. 25.101 / 56. 25.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 11.102 /  5. 11.102
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-i' ... matched as input file with argument '1.webm'.
Reading option 'asd.wav' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input file 1.webm.
Successfully parsed a group of options.
Opening an input file: 1.webm.
[matroska,webm @ 0x7fbfa300da00] Format matroska,webm probed with size=2048 and score=100
st:0 removing common factor 1000000 from timebase
[matroska,webm @ 0x7fbfa300da00] Before avformat_find_stream_info() pos: 473 bytes read:32768 seeks:0
[matroska,webm @ 0x7fbfa300da00] All info found
[matroska,webm @ 0x7fbfa300da00] After avformat_find_stream_info() pos: 942 bytes read:32768 seeks:0 frames:1
Input #0, matroska,webm, from '1.webm':
  Metadata:
    encoder         : Lavf56.25.101
  Duration: 00:00:05.01, start: 0.007000, bitrate: 94 kb/s
    Stream #0:0, 1, 1/1000: Audio: opus, 48000 Hz, stereo, fltp (default)
Successfully opened the file.
Parsing a group of options: output file asd.wav.
Successfully parsed a group of options.
Opening an output file: asd.wav.
File 'asd.wav' already exists. Overwrite ? [y/N] y
Successfully opened the file.
detected 8 logical cores
[graph 0 input from stream 0:0 @ 0x7fbfa42003a0] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0x7fbfa42003a0] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0x7fbfa42003a0] Setting 'sample_fmt' to value 'fltp'
[graph 0 input from stream 0:0 @ 0x7fbfa42003a0] Setting 'channel_layout' to value '0x3'
[graph 0 input from stream 0:0 @ 0x7fbfa42003a0] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x3
[audio format for output stream 0:0 @ 0x7fbfa42008a0] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0x7fbfa42008a0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
[AVFilterGraph @ 0x7fbfa4000160] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0x7fbfa4201060] ch:2 chl:stereo fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz
Output #0, wav, to 'asd.wav':
  Metadata:
    ISFT            : Lavf56.25.101
    Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s (default)
    Metadata:
      encoder         : Lavc56.26.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (opus (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[opus @ 0x7fbfa3011400] skip 0 samples due to side data
[opus @ 0x7fbfa3011400] discard 648/960 samples
[output stream 0:0 @ 0x7fbfa4200620] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
size=     939kB time=00:00:05.00 bitrate=1536.0kbits/s
video:0kB audio:939kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.008114%
Input file #0 (1.webm):
  Input stream #0:0 (audio): 251 packets read (57217 bytes); 251 frames decoded (240312 samples);
  Total: 251 packets (57217 bytes) demuxed
Output file #0 (asd.wav):
  Output stream #0:0 (audio): 251 frames encoded (240312 samples); 251 packets muxed (961248 bytes);
  Total: 251 packets (961248 bytes) muxed
251 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x7fbfa2f00000] Statistics: 4 seeks, 254 writeouts
[AVIOContext @ 0x7fbfa2c357c0] Statistics: 59467 bytes read, 0 seeks

          #5039: Incorrect TrueHD decoding   

http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/203994 A user uploaded a TrueHD sample that is supposed to only contain 0xFFFFFF but the two first channels of the output file always contain 0x01 or 0x02.

$ ffmpeg -i warnings_cut.thd -acodec pcm_s24le out.wav
ffmpeg version N-76844-g3885ef0 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      55.  9.100 / 55.  9.100
  libavcodec     57. 16.100 / 57. 16.100
  libavformat    57. 19.100 / 57. 19.100
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 15.100 /  6. 15.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, truehd, from 'warnings_cut.thd':
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: truehd, 48000 Hz, 7.1, s32 (24 bit)
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.19.100
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 7.1, s32 (24 bit), 9216 kb/s
    Metadata:
      encoder         : Lavc57.16.100 pcm_s24le
Stream mapping:
  Stream #0:0 -> #0:0 (truehd (native) -> pcm_s24le (native))
Press [q] to stop, [?] for help
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 05.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 08.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0b.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 08.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0f.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 08.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0f.
...
...
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0c.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 09.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0e.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0f.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 04.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 03.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0a.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 01.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 02.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0a.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 09.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 05.
size=   56934kB time=00:00:50.60 bitrate=9216.0kbits/s
video:0kB audio:56933kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000175%

          #5053: Cannot select sample rate for the link between filters audio format   

Summary of the bug:

I upgraded from 2.8 to 2.8.3 and my project no longer worked.

Query

ffmpeg -y \
-ss 0 -i testdata/video.webm \
-ss 0 -i backingtrack.mp3 \
-ss 0 -i testdata/audio.wav \
-i overlay.png \
-i suffix.mp4 \
-filter_complex \
\
        "[1:a][2:a] amix=inputs=2:duration=shortest [mergedAudio]; \
        [mergedAudio] asplit=3 [fullAudio][a1][a2]; \
        [a1] atrim=0:2 [vineAudio]; \
        [a2] atrim=0:4 [instagramAudio]; \
        [0]setpts=PTS-STARTPTS[video]; \
        [video][3]overlay[overlayed]; \
        [4]setpts=PTS-STARTPTS[suffix]; \
        [suffix] split=3 [suffix1][suffix2][suffix3]; \
        [overlayed] split=3 [fullIn][instagramIn][vineIn]; \
        [vineIn] trim=0:2 [vineOut]; \
        [instagramIn] trim=0:4 [instagramOut]; \
        [fullIn][suffix1] concat=n=2:v=1:a=0 [full]; \
        [vineOut][suffix2] concat=n=2:v=1:a=0 [vine]; \
        [instagramOut][suffix3] concat=n=2:v=1:a=0 [instagram]" \
\
-acodec libvorbis -b:a 128k -ac 2 -r 25 -vcodec libvpx -b:v 1024k \
-map [full] -map [fullAudio] testdata/output/full.webm \
-map [instagram] -map [instagramAudio] testdata/output/instagram.webm \
-map [vine] -map [vineAudio] testdata/output/vine.webm

Output:

ffmpeg version 2.8.3-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 5.2.1 (Debian 5.2.1-26) 20151125
  configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg --enable-libopus --enable-libass --enable-gnutls --enable-libvidstab --enable-libsoxr --enable-frei0r --enable-libfribidi --disable-indev=sndio --disable-outdev=sndio --cc=gcc
  libavutil      54. 31.100 / 54. 31.100
  libavcodec     56. 60.100 / 56. 60.100
  libavformat    56. 40.101 / 56. 40.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 40.101 /  5. 40.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc    53.  3.100 / 53.  3.100
Input #0, matroska,webm, from 'testdata/video.webm':
  Metadata:
    encoder         : whammy
  Duration: 00:00:08.08, start: 0.000000, bitrate: 2097 kb/s
    Stream #0:0: Video: vp8, yuv420p, 480x320, SAR 1:1 DAR 3:2, 1k tbr, 1k tbn, 1k tbc (default)
[mp3 @ 0xbc38740] Skipping 0 bytes of junk at 4096.
[mp3 @ 0xbc38740] Estimating duration from bitrate, this may be inaccurate
Input #1, mp3, from 'backingtrack.mp3':
  Metadata:
    TSS             : Logic 10.2.0
  Duration: 00:00:08.07, start: 0.000000, bitrate: 164 kb/s
    Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 160 kb/s
Guessed Channel Layout for  Input Stream #2.0 : stereo
Input #2, wav, from 'testdata/audio.wav':
  Duration: 00:00:08.92, bitrate: 1411 kb/s
    Stream #2:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s
Input #3, png_pipe, from 'overlay.png':
  Duration: N/A, bitrate: N/A
    Stream #3:0: Video: png, rgba(pc), 480x320 [SAR 2835:2835 DAR 3:2], 25 tbr, 25 tbn, 25 tbc
Input #4, mov,mp4,m4a,3gp,3g2,mj2, from 'suffix.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf55.33.100
  Duration: 00:00:01.57, start: 0.000000, bitrate: 117 kb/s
    Stream #4:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 480x320 [SAR 1:1 DAR 3:2], 111 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
    Metadata:
      handler_name    : VideoHandler
[audio format for output stream 0:1 @ 0xbc74960] Cannot select sample rate for the link between filters audio format for output stream 0:1 and output stream 0:1.
Error configuring complex filters.
Invalid argument

Also tested with git build g6b978da built on 20151130, same result.

It works if I add a "aformat=sample_rates=44100" step in the audio filter chain, but this was not needed before.


          #5075: Duration of M4A file is wrong?   

I think duration is an issue after all. I attached files to demonstrate it.

8.wav and 9.wav are original ones. There should not be any gap in between (those are small pieces of tracks 8 and 9 from Depeche Mode Violator).

When I concatenate files 8.wav and 9.wav and then open resulting 10w.wav in Audacity I see that there's no gap in between, so this is OK. It sound fine as well.

ffmpeg -f concat -i ccw.txt  10w.wav

8.m4a and 9.m4a are ones created with FFmpeg. When I concatenate them like this and then open resulting 10.wav in Audacity I see that there is a gap in between. That can be heard as well.

ffmpeg -f concat -i cc.txt  10.wav

          #5091: Icecast SSL support (icecast:// is http-only)   

Summary of the bug: icecast:// does not work if server is running with https How to reproduce: Output icecast:// to an icecast server running https

% ffmpeg -i song.wav -acodec libfdk_aac -vbr 4 -content_type "audio/aac" -f adts icecast://source:hackme@exampleserver.com:8000/stream.aac
ffmpeg version 2.8.3
built on ... Linux, today

Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker.


          #5179: Native AAC encoder does not respect channelmap assignments   

Tried to generate a 4-channel AAC stream with quad(side) layout, but the output MP4 reports stream layout as 4.0. WAV is generated with correct layout. Using Zeranoe static binary.


Generation of dummy stereo source.

ffmpeg -f lavfi -i "aevalsrc=0|0:d=5" null0.mp4 -report 2> null0.log
ffmpeg version N-77715-gfc703f5 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.2.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 12.100 / 55. 12.100
  libavcodec     57. 21.100 / 57. 21.100
  libavformat    57. 21.101 / 57. 21.101
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 23.100 /  6. 23.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, lavfi, from 'aevalsrc=0|0:d=5':
  Duration: N/A, start: 0.000000, bitrate: 5644 kb/s
    Stream #0:0: Audio: pcm_f64le, 44100 Hz, stereo, dbl, 5644 kb/s
Output #0, mp4, to 'null0.mp4':
  Metadata:
    encoder         : Lavf57.21.101
    Stream #0:0: Audio: aac (LC) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc57.21.100 aac
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_f64le (native) -> aac (native))
Press [q] to stop, [?] for help
size=       3kB time=00:00:05.01 bitrate=   4.7kbits/s speed=79.6x
video:0kB audio:1kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 122.289612%
[aac @ 00000000003ad8c0] Qavg: 65536.000

Generation of 4 channel AAC in MP4

ffmpeg -i null0.mp4 -lavfi "[0]asplit[a][b];[a][b]amerge[o];[o]channelmap=0|1|2|3:quad(side)[out]" -map "[out]" null1.mp4 2> null1.log
ffmpeg version N-77715-gfc703f5 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.2.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 12.100 / 55. 12.100
  libavcodec     57. 21.100 / 57. 21.100
  libavformat    57. 21.101 / 57. 21.101
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 23.100 /  6. 23.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'null0.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf57.21.101
  Duration: 00:00:05.04, start: 0.023220, bitrate: 4 kb/s
    Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 2 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
[Parsed_amerge_1 @ 00000000029a3cc0] No channel layout for input 1
[Parsed_amerge_1 @ 00000000029a3cc0] Input channel layouts overlap: output layout will be determined by the number of distinct input channels
Output #0, mp4, to 'null1.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf57.21.101
    Stream #0:0: Audio: aac (LC) ([64][0][0][0] / 0x0040), 44100 Hz, quad(side), fltp, 128 kb/s (default)
    Metadata:
      encoder         : Lavc57.21.100 aac
Stream mapping:
  Stream #0:0 (aac) -> asplit
  channelmap -> Stream #0:0 (aac)
Press [q] to stop, [?] for help
size=       4kB time=00:00:05.03 bitrate=   7.1kbits/s speed=46.7x
video:0kB audio:3kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 56.716942%
[aac @ 00000000029a22c0] Qavg: 65536.000
ffprobe null1.mp4
=================
ffprobe version N-77715-gfc703f5 Copyright (c) 2007-2016 the FFmpeg developers
  built with gcc 5.2.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 12.100 / 55. 12.100
  libavcodec     57. 21.100 / 57. 21.100
  libavformat    57. 21.101 / 57. 21.101
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 23.100 /  6. 23.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'null1.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf57.21.101
  Duration: 00:00:05.06, start: 0.023220, bitrate: 7 kb/s
    Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, 4.0, fltp, 4 kb/s (default)
    Metadata:
      handler_name    : SoundHandler

Generation of 4-channel WAV

ffmpeg -i null0.mp4 -lavfi "[0]asplit[a][b];[a][b]amerge[o];[o]channelmap=0|1|2|3:quad(side)[out]" -map [out] null1.wav 2> null1wav.log
ffmpeg version N-77715-gfc703f5 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.2.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 12.100 / 55. 12.100
  libavcodec     57. 21.100 / 57. 21.100
  libavformat    57. 21.101 / 57. 21.101
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 23.100 /  6. 23.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'null0.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf57.21.101
  Duration: 00:00:05.04, start: 0.023220, bitrate: 4 kb/s
    Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 2 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
[Parsed_amerge_1 @ 0000000002613c20] No channel layout for input 1
[Parsed_amerge_1 @ 0000000002613c20] Input channel layouts overlap: output layout will be determined by the number of distinct input channels
Output #0, wav, to 'null1.wav':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    ISFT            : Lavf57.21.101
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, quad(side), s16, 2822 kb/s (default)
    Metadata:
      encoder         : Lavc57.21.100 pcm_s16le
Stream mapping:
  Stream #0:0 (aac) -> asplit
  channelmap -> Stream #0:0 (pcm_s16le)
Press [q] to stop, [?] for help
size=    1736kB time=00:00:05.03 bitrate=2822.6kbits/s speed= 157x
video:0kB audio:1736kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.005738%
ffprobe null1.wav
=================
ffprobe version N-77715-gfc703f5 Copyright (c) 2007-2016 the FFmpeg developers
  built with gcc 5.2.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 12.100 / 55. 12.100
  libavcodec     57. 21.100 / 57. 21.100
  libavformat    57. 21.101 / 57. 21.101
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 23.100 /  6. 23.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, wav, from 'null1.wav':
  Metadata:
    encoder         : Lavf57.21.101
  Duration: 00:00:05.04, bitrate: 2822 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, quad(side), s16, 2822 kb/s

          #5273: adpcm_swf: 11khz mode decodes as 5khz   

I don't know if this is a bug in ffmpeg or in encoding software.

C:\>ffmpeg -i 11khz_mono_44kbps.flv out.wav
ffmpeg version N-78410-g588e2e3 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
ble-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --
enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-lib
x265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-d
ecklink --enable-zlib
  libavutil      55. 17.100 / 55. 17.100
  libavcodec     57. 24.102 / 57. 24.102
  libavformat    57. 25.100 / 57. 25.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 31.100 /  6. 31.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, flv, from '11khz_mono_44kbps.flv':
  Metadata:
    creationdate    : Thu Feb 25 04:13:06 2016
    Encoded_With    : Sorenson Squeeze
    Encoded_By      :
  Duration: 00:00:02.43, start: 0.000000, bitrate: 183 kb/s
    Stream #0:0: Audio: adpcm_swf, 5512 Hz, mono, s16, 22 kb/s
Output #0, wav, to 'out.wav':
  Metadata:
    creationdate    : Thu Feb 25 04:13:06 2016
    Encoded_With    : Sorenson Squeeze
    ITCH            :
    ISFT            : Lavf57.25.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 5512 Hz, mono, s16, 8
8 kb/s
    Metadata:
      encoder         : Lavc57.24.102 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_swf (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=     208kB time=00:00:09.74 bitrate= 175.0kbits/s speed=N/A
video:0kB audio:208kB subtitle:0kB other streams:0kB global headers:0kB muxing o
verhead: 0.036584%

          #5457: mp3 delay issue   

This issue is reproduced with

ffmpeg version N-79546-g13406b6 Copyright (c) 2000-2016 the FFmpeg developers

Description: Run below commands to generate a.wav and b.wav

ffmpeg.exe -i big50.mp4 -acodec libmp3lame a.mp3
ffmpeg.exe -i a.mp3 a.wav
ffmpeg.exe -i big50.mp4 -vcodec libx264 -acodec libmp3lame b.avi
ffmpeg.exe -i b.avi b.wav

Open a.wav and b.wav in wav file editor you can see there's 50 ms delay in b.wav. (sample rate is 22050, so that 50 ms should be delay introduced by LAME that are don't properly handled by ffmpeg?)

Source mp4 is here: https://dl.dropboxusercontent.com/u/89678527/big50.mp4 and generated wav files: https://dl.dropboxusercontent.com/u/89678527/a.wav https://dl.dropboxusercontent.com/u/89678527/b.wav

here's comparison of a.wav and b.wav https://dl.dropboxusercontent.com/u/89678527/mp3.PNG


          #5481: g723 in au: fix decoding   

decodes ok with sox

https://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket1955/g723_44khz_3bit.au

C:\>ffmpegx -i C:\g723_44khz_3bit.au out.wav
ffmpeg version N-79632-g3ce1988 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-pthreads --enable-memalign-hack --enable-runtime-cpudetect --
enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -lw
inmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snapsh
ots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/sna
pshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3 --
enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-li
bvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-li
bopencore-amrwb --enable-libmp3lame --enable-libfreetype --enable-libvpx --disab
le-decoder=libvpx
  libavutil      55. 22.101 / 55. 22.101
  libavcodec     57. 38.100 / 57. 38.100
  libavformat    57. 34.103 / 57. 34.103
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 44.100 /  6. 44.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, au, from 'C:\g723_44khz_3bit.au':
  Duration: 00:00:30.01, start: 0.000000, bitrate: 132 kb/s
    Stream #0:0: Audio: adpcm_g726le ([25][0][0][0] / 0x0019), 44100 Hz, mono, s
16, 132 kb/s
File 'out.wav' already exists. Overwrite ? [y/N] y
[wav @ 0x2c3eb60] Using AVStream.codec to pass codec parameters to muxers is dep
recated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.34.103
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16,
705 kb/s
    Metadata:
      encoder         : Lavc57.38.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_g726le (g726le) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[g726le @ 0x2c3e6a0] Frame invalidly split, missing parser?
    Last message repeated 484 times
size=    2585kB time=00:00:30.00 bitrate= 705.6kbits/s speed=66.2x
video:0kB audio:2585kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.002947%

          #5725: ffmpeg: error during decoding mp3 with APE tag   

FFmpeg tool as well as certain applications have problems decoding MP3 files with APE tags. The tag is reported as an error while decoding stream and some applications using ffmpeg libraries abort decoding in such case.

Issue is also reproducible in ffmpeg 2.7.7 and probably in all releases up to date.

ffmpeg -loglevel trace -i Ice\ Wharf.mp3 Ice\ Wharf.wav
ffmpeg version N-80901-gfebc862 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
  configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab
  libavutil      55. 28.100 / 55. 28.100
  libavcodec     57. 48.101 / 57. 48.101
  libavformat    57. 41.100 / 57. 41.100
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 47.100 /  6. 47.100
  libavresample   3.  0.  0 /  3.  0.  0
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'trace'.
Reading option '-i' ... matched as input file with argument 'Ice Wharf.mp3'.
Reading option 'Ice Wharf.wav' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument trace.
Successfully parsed a group of options.
Parsing a group of options: input file Ice Wharf.mp3.
Successfully parsed a group of options.
Opening an input file: Ice Wharf.mp3.
[file @ 0xa9af6e0] Setting default whitelist 'file,crypto'
Probing aac score:1 size:1875
Probing mp3 score:25 size:1875
Probing aac score:1 size:3923
Probing mp3 score:51 size:3923
[mp3 @ 0xa9aef60] Format mp3 probed with size=4096 and score=51
[mp3 @ 0xa9aef60] id3v2 ver:3 flags:00 len:163
[mp3 @ 0xa9aef60] pad 576 1681
[mp3 @ 0xa9aef60] Skipping 0 bytes of junk at 590.
[mp3 @ 0xa9aef60] Before avformat_find_stream_info() pos: 590 bytes read:32768 seeks:0 nb_streams:1
[mp3 @ 0xa9aef60] demuxer injecting skip 1105 / discard 0
[mp3 @ 0xa9b01e0] skip 1105 / discard 0 samples due to side data
[mp3 @ 0xa9b01e0] skip 1105/1152 samples
[mp3 @ 0xa9aef60] All info found
[mp3 @ 0xa9aef60] stream 0: start_time: 0.025 duration: 238.341
[mp3 @ 0xa9aef60] format: start_time: 0.025 duration: 238.341 bitrate=128 kb/s
[mp3 @ 0xa9aef60] After avformat_find_stream_info() pos: 22094 bytes read:32768 seeks:0 frames:50
Input #0, mp3, from 'Ice Wharf.mp3':
  Metadata:
    title           : Ice Wharf
    artist          : Work Drugs
    album           : Aurora Lies
    track           : 6
  Duration: 00:03:58.34, start: 0.025057, bitrate: 128 kb/s
    Stream #0:0, 50, 1/14112000: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
    Metadata:
      encoder         : LAME3.96r
    Side data:
      replaygain: track gain - -6.200000, track peak - unknown, album gain - unknown, album peak - unknown,
Successfully opened the file.
Parsing a group of options: output file Ice Wharf.wav.
Successfully parsed a group of options.
Opening an output file: Ice Wharf.wav.
[file @ 0xa9bfd40] Setting default whitelist 'file,crypto'
Successfully opened the file.
detected 8 logical cores
[graph 0 input from stream 0:0 @ 0xa9b2880] Setting 'time_base' to value '1/44100'
[graph 0 input from stream 0:0 @ 0xa9b2880] Setting 'sample_rate' to value '44100'
[graph 0 input from stream 0:0 @ 0xa9b2880] Setting 'sample_fmt' to value 's16p'
[graph 0 input from stream 0:0 @ 0xa9b2880] Setting 'channel_layout' to value '0x3'
[graph 0 input from stream 0:0 @ 0xa9b2880] tb:1/44100 samplefmt:s16p samplerate:44100 chlayout:0x3
[audio format for output stream 0:0 @ 0xa9c0820] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0xa9c0820] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
[AVFilterGraph @ 0xa9b2800] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0xa9c1e80] [SWR @ 0xa9db240] Using s16p internally between filters
[auto-inserted resampler 0 @ 0xa9c1e80] ch:2 chl:stereo fmt:s16p r:44100Hz -> ch:2 chl:stereo fmt:s16 r:44100Hz
[wav @ 0xa9b16c0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, wav, to 'Ice Wharf.wav':
  Metadata:
    INAM            : Ice Wharf
    IART            : Work Drugs
    IPRD            : Aurora Lies
    IPRT            : 6
    ISFT            : Lavf57.41.100
    Stream #0:0, 0, 1/44100: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      encoder         : Lavc57.48.101 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[mp3 @ 0xa9b0720] skip 1105 / discard 0 samples due to side data
[mp3 @ 0xa9b0720] skip 1105/1152 samples
[mp3 @ 0xa9aef60] demuxer injecting skip 0 / discard 1152eed= 242x
[mp3 @ 0xa9b0720] skip 0 / discard 1152 samples due to side data
[mp3 @ 0xa9b0720] Header missing
Error while decoding stream #0:0: Invalid data found when processing input
[output stream 0:0 @ 0xa9c0b80] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
size=   41049kB time=00:03:58.29 bitrate=1411.2kbits/s speed= 245x
video:0kB audio:41049kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000347%
Input file #0 (Ice Wharf.mp3):
  Input stream #0:0 (audio): 9125 packets read (3813711 bytes); 9123 frames decoded (10508591 samples);
  Total: 9125 packets (3813711 bytes) demuxed
Output file #0 (Ice Wharf.wav):
  Output stream #0:0 (audio): 9123 frames encoded (10508591 samples); 9123 packets muxed (42034364 bytes);
  Total: 9123 packets (42034364 bytes) muxed
9123 frames successfully decoded, 1 decoding errors
[AVIOContext @ 0xa9b26c0] Statistics: 4 seeks, 9126 writeouts
[AVIOContext @ 0xa9b7880] Statistics: 3814255 bytes read, 0 seeks

          #5820: WMA Lossless 24-bit Distortions via RTSP   

Summary of the bug: When trans-coding an RTSP stream with WMA Lossless 24-bit audio to a WAV file, ffmpeg produces many audible artifacts. Stream is publicly available for testing: rtsp://djxmmx.net/rap-rnb. Please see next post for links to the output files produced by ffmpeg. How to reproduce:

ffmpeg -i rtsp://djxmmx.net/rap-rnb C:\Temp\Out-RTSP.wav
ffmpeg version N-81516-gbe07c25 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.4.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --ena
ble-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --ena
ble-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --e
nable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-lib
openh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-lib
soxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvo
rbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-li
bxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 29.100 / 55. 29.100
  libavcodec     57. 54.101 / 57. 54.101
  libavformat    57. 48.101 / 57. 48.101
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 58.100 /  6. 58.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
[wmalossless @ 00000000003fe4e0] Decoding audio at 24 bit-depth
Input #0, rtsp, from 'rtsp://djxmmx.net/rap-rnb':
  Metadata:
    MediaFoundationVersion: 2.112
    DeviceConformanceTemplate: N2
    WM/WMADRCPeakReference: 32767
    WM/WMADRCPeakTarget: 32767
    WM/WMADRCAverageReference: 10194
    WM/WMADRCAverageTarget: 10194
    genre           : R&B
    WMFSDKVersion   : 11.0.5721.5145
    WMFSDKNeeded    : 0.0.0.0000
    album           : Thr33 Ringz
    track           : 11
    WM/Track        : 10
    WM/Year         : 2008
    PeakValue       : 32673
    AverageLevel    : 7035
    IsVBR           : 1
    title           : Long Lap Dance
    artist          : T-Pain
  Duration: 00:04:36.47, start: 199.296000, bitrate: N/A
    Stream #0:0: Audio: wmalossless (c[1][0][0] / 0x0163), 96000 Hz, stereo, s32p (24 bit), 1152 kb/s
[wmalossless @ 0000000002d04120] Decoding audio at 24 bit-depth
[wav @ 0000000002d045c0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar in
stead.
Output #0, wav, to 'C:\Temp\Out-RTSP.wav':
  Metadata:
    MediaFoundationVersion: 2.112
    DeviceConformanceTemplate: N2
    WM/WMADRCPeakReference: 32767
    WM/WMADRCPeakTarget: 32767
    WM/WMADRCAverageReference: 10194
    WM/WMADRCAverageTarget: 10194
    IGNR            : R&B
    WMFSDKVersion   : 11.0.5721.5145
    WMFSDKNeeded    : 0.0.0.0000
    IPRD            : Thr33 Ringz
    IPRT            : 11
    WM/Track        : 10
    WM/Year         : 2008
    PeakValue       : 32673
    AverageLevel    : 7035
    IsVBR           : 1
    INAM            : Long Lap Dance
    IART            : T-Pain
    ISFT            : Lavf57.48.101
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 96000 Hz, stereo, s16, 3072 kb/s
    Metadata:
      encoder         : Lavc57.54.101 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (wmalossless (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 141 packets
[wmalossless @ 0000000002d04120] broken frame: channel len(6656) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(4352) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(4608) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(7680) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(5632) > samples_per_frame(4096)
[wmalossless @ 0000000002d04120] broken frame: channel len(4864) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(4352) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] Order[0][1] 608 > max (256), not supported
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 15 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq 3 vs 6
[wmalossless @ 0000000002d04120] broken frame: channel len(5120) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] Arithmetic coding is not implemented. Update your FFmpeg version to the newest one from
 Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[wmalossless @ 0000000002d04120] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/
and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(4352) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(4352) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] Order[0][0] 776 > max (256), not supported
Error while decoding stream #0:0: Invalid data found when processing input
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 6 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq 6 vs 8
[wmalossless @ 0000000002d04120] broken frame: channel len(4608) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(6912) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 24 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq 2 vs 6
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 6 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq c vs e
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 24 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq 4 vs 8
[rtsp @ 000000000030a920] RTP: missed 9 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq 8 vs a
[wmalossless @ 0000000002d04120] broken frame: channel len(4608) > samples_per_frame(4096)
[wmalossless @ 0000000002d04120] broken frame: channel len(6144) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] Expect wrong output since inverse LPC filter is not implemented. Update your FFmpeg ver
sion to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been i
mplemented.
[wmalossless @ 0000000002d04120] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/
and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
[wmalossless @ 0000000002d04120] Expect wrong output since inverse LPC filter is not implemented. Update your FFmpeg ver
sion to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been i
mplemented.
[wmalossless @ 0000000002d04120] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/
and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
[wmalossless @ 0000000002d04120] broken frame: channel len(4608) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] Order[0][0] 968 > max (256), not supported
Error while decoding stream #0:0: Invalid data found when processing input
size=   24720kB time=00:01:08.18 bitrate=2970.1kbits/s speed=1.27x
video:0kB audio:24720kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000735%
Exiting normally, received signal 2.

          #5829: ADPCM SWF wave decode error, but directly decode *.flv right   

Summary of the bug:

ADPCM SWF wave decode error, but directly decode *.flv right

% ./ffmpeg -i ~/program/swf_adpcm/mono_11k.flv -vn -acodec copy mono_11k_bs.wav % ./ffmpeg -i mono_11k_bs.wav mono_11k_pcm.wav Got mono_11k_pcm.wav error.

% ./ffmpeg -i ~/program/swf_adpcm/mono_11k.flv -vn mono_11k_pcm_right.wav Got mono_11k_pcm_right.wav right.

*.flv ------> right.wav *.flv --> swf.wav --> error.wav

Also, use SWF ADPCM encode to a wav, then decode it to PCM, meet error. PCM --> swf.wav --> PCM(error)

Root cause in my opinion: SWF adpcm take block_align as once decoded frame_size. When decode *.flv, we can get the right frame_size. When decode swf.wav get the wrong frame_size.

How to reproduce:

CMD:
% ./ffmpeg -i ~/program/swf_adpcm/mono_11k.flv -vn -acodec copy mono_11k_bs.wav
% ./ffmpeg -i mono_11k_bs.wav mono_11k_pcm.wav
% ./ffmpeg -i ~/program/swf_adpcm/mono_11k.flv -vn mono_11k_pcm_right.wav
Version:
From https://git.ffmpeg.org/ffmpeg
   369ed11..ba7be8c  master     -> origin/master
   8887daf..c32ce24  release/2.8 -> origin/release/2.8
   b62191f..3512ed3  release/3.0 -> origin/release/3.0
   4275b27..677ea4a  release/3.1 -> origin/release/3.1
 * [new tag]         n3.0.3     -> n3.0.3
From https://git.ffmpeg.org/ffmpeg
 * [new tag]         n3.1.3     -> n3.1.3
Updating 369ed11..ba7be8c
Log:
ligd@bjdevel03 ffmpeg $./ffmpeg -i ~/program/swf_adpcm/mono_11k.flv -vn -acodec copy mono_11k_bs.wav
ffmpeg version N-81571-gba7be8c Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.4.5 (Debian 4.4.5-8)
  configuration: --prefix=/home/ligd/git/
  libavutil      55. 29.100 / 55. 29.100
  libavcodec     57. 55.101 / 57. 55.101
  libavformat    57. 48.102 / 57. 48.102
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 60.100 /  6. 60.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
[flv @ 0xa48b260] video stream discovered after head already parsed
[flv @ 0xa48b260] audio stream discovered after head already parsed
Input #0, flv, from '/home/ligd/program/swf_adpcm/mono_11k.flv':
  Duration: 00:01:00.46, start: 0.000000, bitrate: 810 kb/s
    Stream #0:0: Video: flv1, yuv420p, 640x352, 12 fps, 12 tbr, 1k tbn
    Stream #0:1: Audio: adpcm_swf, 11025 Hz, mono, s16, 44 kb/s
[wav @ 0xa4e4b00] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
[wav @ 0xa4e4b00] requested bits_per_coded_sample (16) and actually stored (4) differ
Output #0, wav, to 'mono_11k_bs.wav':
  Metadata:
    ISFT            : Lavf57.48.102
    Stream #0:0: Audio: adpcm_swf (FS[0][0] / 0x5346), 11025 Hz, mono, 44 kb/s
Stream mapping:
  Stream #0:1 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=     327kB time=00:01:00.46 bitrate=  44.3kbits/s speed=1.73e+03x
video:0kB audio:327kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.027772%
ligd@bjdevel03 ffmpeg $
ligd@bjdevel03 ffmpeg $
ligd@bjdevel03 ffmpeg $./ffmpeg -i mono_11k_bs.wav mono_11k_pcm.wav
ffmpeg version N-81571-gba7be8c Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.4.5 (Debian 4.4.5-8)
  configuration: --prefix=/home/ligd/git/
  libavutil      55. 29.100 / 55. 29.100
  libavcodec     57. 55.101 / 57. 55.101
  libavformat    57. 48.102 / 57. 48.102
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 60.100 /  6. 60.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, wav, from 'mono_11k_bs.wav':
  Metadata:
    encoder         : Lavf57.48.102
  Duration: 00:01:00.46, bitrate: 44 kb/s
    Stream #0:0: Audio: adpcm_swf (FS[0][0] / 0x5346), 11025 Hz, 1 channels, s16, 44 kb/s
[wav @ 0xb44c400] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, wav, to 'mono_11k_pcm.wav':
  Metadata:
    ISFT            : Lavf57.48.102
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 11025 Hz, mono, s16, 176 kb/s
    Metadata:
      encoder         : Lavc57.55.101 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_swf (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    1904kB time=00:01:28.42 bitrate= 176.4kbits/s speed=2.61e+03x
video:0kB audio:1904kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.004001%

Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker.


          #6071: loudnorm filter always returns LRA of 0.0 LU with short files (less than ~3 seconds)   

I was trying to execute a 2 pass loudnorm filter on a file less than 1 second and it wouldn't work properly - the output of the second pass was not changed by much and the target output values were missed by a long way. On the first pass, the measured input_lra value was always 0.0 which seems wrong and the output_i is nowhere near the target.

I verified my build was working ok by running a test on the loudnorm_before.mp3 found at http://k.ylo.ph/2016/04/04/loudnorm.html

ffmpeg version N-83034-gf48b6b8 Copyright (c) 2000-2017 the FFmpeg developers

built with gcc 4.8.3 (GCC) 20140911 (Red Hat 4.8.3-9) configuration: --prefix=/root/ffmpeg_build --extra-cflags=-I/root/ffmpeg_build/include --extra-ldflags='-L/root/ffmpeg_build/lib -ldl' --bindir=/root/bin --pkg-config-flags=--static --enable-gpl --enable-nonfree --enable-libfdk_aac libavutil 55. 43.100 / 55. 43.100 libavcodec 57. 71.100 / 57. 71.100 libavformat 57. 62.100 / 57. 62.100 libavdevice 57. 2.100 / 57. 2.100 libavfilter 6. 68.100 / 6. 68.100 libswscale 4. 3.101 / 4. 3.101 libswresample 2. 4.100 / 2. 4.100 libpostproc 54. 2.100 / 54. 2.100

I setup a new test file and concatenated it together several times. The file bonjour.wav is 1.17 seconds long and the LRA could not be measured. The file bonjour_twice.wav is 2.33 seconds and the LRA could not be measured. The file bonjour_three.wav is 3.49 seconds and the LRA CAN be measured.

I have successfully used those measurements to properly normalise all 3 input files. ie. the normalisation pass works properly on small files.

ffmpeg -i bonjour_twice.wav -af loudnorm=I=-16:tp=-1.5:LRA=11:print_format=json -f null -

size=N/A time=00:00:02.33 bitrate=N/A speed=74.1x

{

"input_i" : "-22.85", "input_tp" : "-10.19", "input_lra" : "0.00", "input_thresh" : "-32.85", "output_i" : "-22.33", "output_tp" : "-9.66", "output_lra" : "0.00", "output_thresh" : "-32.33", "normalization_type" : "linear", "target_offset" : "6.33"

}

ffmpeg -i bonjour_three.wav -af loudnorm=I=-16:tp=-1.5:LRA=11:print_format=json -f null -

size=N/A time=00:00:03.49 bitrate=N/A speed=46.2x

{

"input_i" : "-22.84", "input_tp" : "-10.19", "input_lra" : "0.30", "input_thresh" : "-32.84", "output_i" : "-15.94", "output_tp" : "-3.29", "output_lra" : "0.00", "output_thresh" : "-25.94", "normalization_type" : "dynamic", "target_offset" : "-0.06"

}


          #6098: qcelp: "IFQ: Badly received packets in frame."   
C:\>ffmpeg -i qcelp_13k.mov out.wav
ffmpeg version 3.2.git Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 4.6.2 (GCC)
  configuration: --disable-pthreads --enable-gpl --disable-ffprobe
  libavutil      55. 43.100 / 55. 43.100
  libavcodec     57. 72.100 / 57. 72.100
  libavformat    57. 62.100 / 57. 62.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 69.100 /  6. 69.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
[qcelp @ 0667d680] Warning: not compiled with thread support, using thread emula
tion
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'qcelp_13k.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2016-12-29T13:36:49.000000Z
  Duration: 00:01:00.00, start: 0.000000, bitrate: 14 kb/s
    Stream #0:0(eng): Audio: qcelp (Qclp / 0x706C6351), 8000 Hz, mono, flt (defa
ult)
    Metadata:
      creation_time   : 2016-12-29T13:36:49.000000Z
      handler_name    : Procedura obs│ugi skrˇtˇw danych Apple
[qcelp @ 06a221c0] Warning: not compiled with thread support, using thread emula
tion
[pcm_s16le @ 06a36fc0] Warning: not compiled with thread support, using thread e
mulation
Output #0, wav, to 'out.wav':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    ISFT            : Lavf57.62.100
    Stream #0:0(eng): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, mono, s
16, 128 kb/s (default)
    Metadata:
      creation_time   : 2016-12-29T13:36:49.000000Z
      handler_name    : Procedura obs│ugi skrˇtˇw danych Apple
      encoder         : Lavc57.72.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (qcelp (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[qcelp @ 06a221c0] Frame #105, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #577, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1152, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1154, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1155, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1156, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1157, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1158, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1159, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1160, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1161, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1162, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1163, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1164, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1165, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1166, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1167, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1168, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1169, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1170, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1171, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1172, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1173, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1825, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1826, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1828, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1831, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1832, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1833, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1834, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1835, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1836, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1837, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1838, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1839, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1840, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1841, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1842, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1843, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1844, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1845, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1846, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1848, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1849, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1850, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1851, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1852, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1853, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1854, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1855, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1856, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1857, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1858, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1859, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1860, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1861, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1862, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1863, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1864, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1865, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1866, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1867, IFQ: Badly received packets in frame.
size=     938kB time=00:01:00.00 bitrate= 128.0kbits/s speed=240x
video:0kB audio:938kB subtitle:0kB other streams:0kB global headers:0kB muxing o
verhead: 0.008125%

          #6156: Issue with FFMPEG ALSA CPU usage   

I am on a Raspberry Pi Zero with a Pi Camera and an I2S MEMS mic, running the latest Raspbian Jessie. I am attempting to livestream audio and video. I've discovered that when I record ALSA audio with FFMPEG, that process takes 95% of my CPU and produces stuttering, unusable audio in the stream.

I isolated the issue to the following:

  1. ffmpeg -f alsa -acodec pcm_s32le -i mic_sv temp.wav

This single command takes 95% of my CPU.

  1. arecord -Dmic_sv -c2 -r48000 -fS32_LE -twav temp.wav

This single command takes <5% of my CPU.

I have added a custom hook to ffmpeg so we can see the PCM settings. I have ensured that settings (buffer size, period size, etc.) are identical between the two commands.

Here is the full console output of FFMPEG:

$ ALSA_CONFIG_PATH=/usr/share/alsa/alsa.conf:./dumpasoundrc  /home/pi/special/ffmpeg/ffmpeg -v 9 -loglevel 99  -f alsa -acodec pcm_s32le -i michooks -f  null -
ffmpeg version git-2017-02-05-e57fd92 Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 4.9.2 (Raspbian 4.9.2-10)
  configuration: --extra-cflags=-I/opt/vc/include/IL --enable-nonfree --enable-omx-rpi --logfile=CONFIG.TXT
  libavutil      55. 46.100 / 55. 46.100
  libavcodec     57. 75.100 / 57. 75.100
  libavformat    57. 66.101 / 57. 66.101
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 73.100 /  6. 73.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
Splitting the commandline.
Reading option '-v' ... matched as option 'v' (set logging level) with argument '9'.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument '99'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'alsa'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'pcm_s32le'.
Reading option '-i' ... matched as input url with argument 'michooks'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'null'.
Reading option '-' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set logging level) with argument 9.
Successfully parsed a group of options.
Parsing a group of options: input url michooks.
Applying option f (force format) with argument alsa.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument pcm_s32le.
Successfully parsed a group of options.
Opening an input file: michooks.
[ALSA Debug Hook] Installing hooks.
Hooks PCM
Slave: Soft volume PCM
Control: Boost Capture Volume
min_dB: -3
max_dB: 50
resolution: 256
Slave: Route conversion PCM
  Transformation table:
    0 <- 0
    1 <- 0
Slave: Mmap emulation PCM
Slave: Hardware PCM card 1 'mems-mic' device 0 subdevice 0
[ALSA Debug Hook] hw_params:
Hooks PCM
Slave: Soft volume PCM
Control: Boost Capture Volume
min_dB: -3
max_dB: 50
resolution: 256
Its setup is:
  stream       : CAPTURE
  access       : RW_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Route conversion PCM
  Transformation table:
    0 <- 0
    1 <- 0
Its setup is:
  stream       : CAPTURE
  access       : MMAP_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Mmap emulation PCM
Its setup is:
  stream       : CAPTURE
  access       : MMAP_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Hardware PCM card 1 'mems-mic' device 0 subdevice 0
Its setup is:
  stream       : CAPTURE
  access       : RW_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1073741824
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
  appl_ptr     : 0
  hw_ptr       : 0
[alsa @ 0x31ea450] All info found
[alsa @ 0x31ea450] stream 0: start_time: 1487382765.290 duration: -9223372036854.775
[alsa @ 0x31ea450] format: start_time: 1487382765.290 duration: -9223372036854.775 bitrate=3072 kb/s
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, alsa, from 'michooks':
  Duration: N/A, start: 1487382765.289814, bitrate: 3072 kb/s
    Stream #0:0, 1, 1/1000000: Audio: pcm_s32le, 48000 Hz, stereo, s32, 3072 kb/s
Successfully opened the file.
Parsing a group of options: output url -.
Applying option f (force format) with argument null.
Successfully parsed a group of options.
Opening an output file: -.
Successfully opened the file.
detected 1 logical cores
[graph_0_in_0_0 @ 0x31f9310] Setting 'time_base' to value '1/48000'
[graph_0_in_0_0 @ 0x31f9310] Setting 'sample_rate' to value '48000'
[graph_0_in_0_0 @ 0x31f9310] Setting 'sample_fmt' to value 's32'
[graph_0_in_0_0 @ 0x31f9310] Setting 'channel_layout' to value '0x3'
[graph_0_in_0_0 @ 0x31f9310] tb:1/48000 samplefmt:s32 samplerate:48000 chlayout:0x3
[format_out_0_0 @ 0x31f9f10] Setting 'sample_fmts' to value 's16'
[format_out_0_0 @ 0x31f9f10] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
[AVFilterGraph @ 0x31f9980] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto_resampler_0 @ 0x31faf10] [SWR @ 0x321e130] Using fltp internally between filters
[auto_resampler_0 @ 0x31faf10] ch:2 chl:stereo fmt:s32 r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz
Output #0, null, to 'pipe:':
  Metadata:
    encoder         : Lavf57.66.101
    Stream #0:0, 0, 1/48000: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
    Metadata:
      encoder         : Lavc57.75.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s32le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
size=N/A time=00:00:17.84 bitrate=N/A speed=   1x
video:0kB audio:3345kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Input file #0 (michooks):
  Input stream #0:0 (audio): 26372 packets read (6851424 bytes); 26372 frames decoded (856428 samples);
  Total: 26372 packets (6851424 bytes) demuxed
Output file #0 (pipe:):
  Output stream #0:0 (audio): 26372 frames encoded (856428 samples); 26372 packets muxed (3425712 bytes);
  Total: 26372 packets (3425712 bytes) muxed
26372 frames successfully decoded, 0 decoding errors
Exiting normally, received signal 2.

Here is the console output of Arecord:

$ arecord -v -r48000 -c2 -fS32_LE --period-size=16384 --buffer-size=65536 -Dmic_sv -twav /dev/null
Recording WAVE '/dev/null' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo
Soft volume PCM
Control: Boost Capture Volume
min_dB: -3
max_dB: 50
resolution: 256
Its setup is:
  stream       : CAPTURE
  access       : RW_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Route conversion PCM
  Transformation table:
    0 <- 0
    1 <- 0
Its setup is:
  stream       : CAPTURE
  access       : MMAP_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Mmap emulation PCM
Its setup is:
  stream       : CAPTURE
  access       : MMAP_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Hardware PCM card 1 'mems-mic' device 0 subdevice 0
Its setup is:
  stream       : CAPTURE
  access       : RW_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1073741824
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
  appl_ptr     : 0
  hw_ptr       : 0
^CAborted by signal Interrupt...

After a long chat in the FFMPEG IRC channel, we believe there may be a bug in the way FFMPEG waits for ALSA packets. Arecord uses snd_pcm_wait() to wait for packets, and this command is not used in FFMPEG sources. Inserting -re before -i into my FFMPEG command reduces CPU usage to <5%, but is not the right approach, and causes other problems (bitrate is almost 0kbps, audio file generated is <5 kilobytes).


          #6216: TrueHD encoder creates non bitexact files   

Summary of the bug: How to reproduce:

./ffmpeg -i truehd.wav -f md5 -
ffmpeg version N-83737-g68ee800a9d Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 6.3.0 (Rev2, Built by MSYS2 project)
  configuration: --enable-gpl --enable-nonfree --enable-libx264 --enable-libx265 --enable-libfdk_aac --enable-libvpx --enable-libmp3lame --enable-libvorbis --enable-libopus --target-os=mingw32 --arch=x86_64 --cpu=haswell --extra-cflags='-D_WIN32_WINNT=0x0602' --cc='ccache gcc' --samples=../samples --prefix=/mingw64
  libavutil      55. 47.101 / 55. 47.101
  libavcodec     57. 82.100 / 57. 82.100
  libavformat    57. 66.103 / 57. 66.103
  libavdevice    57.  3.100 / 57.  3.100
  libavfilter     6. 74.100 /  6. 74.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'truehd.wav':
  Metadata:
    encoder         : Lavf57.66.103
  Duration: 00:00:02.02, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, md5, to 'pipe:':
  Metadata:
    encoder         : Lavf57.66.103
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      encoder         : Lavc57.82.100 pcm_s16le
MD5=d886c9feaf7dddcdff3a5adcfa4c57bc
size=       0kB time=00:00:02.02 bitrate=   0.1kbits/s speed= 952x
video:0kB audio:348kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
% ./ffmpeg -i truehd.wav -c:a truehd -strict -2 truehd.thd
ffmpeg version N-83737-g68ee800a9d Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 6.3.0 (Rev2, Built by MSYS2 project)
  configuration: --enable-gpl --enable-nonfree --enable-libx264 --enable-libx265 --enable-libfdk_aac --enable-libvpx --enable-libmp3lame --enable-libvorbis --enable-libopus --target-os=mingw32 --arch=x86_64 --cpu=haswell --extra-cflags='-D_WIN32_WINNT=0x0602' --cc='ccache gcc' --samples=../samples --prefix=/mingw64
  libavutil      55. 47.101 / 55. 47.101
  libavcodec     57. 82.100 / 57. 82.100
  libavformat    57. 66.103 / 57. 66.103
  libavdevice    57.  3.100 / 57.  3.100
  libavfilter     6. 74.100 /  6. 74.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'truehd.wav':
  Metadata:
    encoder         : Lavf57.66.103
  Duration: 00:00:02.02, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> truehd (native))
Press [q] to stop, [?] for help
Output #0, truehd, to 'truehd.thd':
  Metadata:
    encoder         : Lavf57.66.103
    Stream #0:0: Audio: truehd, 44100 Hz, stereo, s16, 128 kb/s
    Metadata:
      encoder         : Lavc57.82.100 truehd
size=      57kB time=00:00:02.02 bitrate= 231.8kbits/s speed= 103x
video:0kB audio:57kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%
./ffmpeg -i truehd.thd -f md5 -
ffmpeg version N-83737-g68ee800a9d Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 6.3.0 (Rev2, Built by MSYS2 project)
  configuration: --enable-gpl --enable-nonfree --enable-libx264 --enable-libx265 --enable-libfdk_aac --enable-libvpx --enable-libmp3lame --enable-libvorbis --enable-libopus --target-os=mingw32 --arch=x86_64 --cpu=haswell --extra-cflags='-D_WIN32_WINNT=0x0602' --cc='ccache gcc' --samples=../samples --prefix=/mingw64
  libavutil      55. 47.101 / 55. 47.101
  libavcodec     57. 82.100 / 57. 82.100
  libavformat    57. 66.103 / 57. 66.103
  libavdevice    57.  3.100 / 57.  3.100
  libavfilter     6. 74.100 /  6. 74.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
Input #0, truehd, from 'truehd.thd':
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: truehd, 44100 Hz, stereo, s32 (24 bit)
Stream mapping:
  Stream #0:0 -> #0:0 (truehd (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, md5, to 'pipe:':
  Metadata:
    encoder         : Lavf57.66.103
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      encoder         : Lavc57.82.100 pcm_s16le
[truehd @ 0000000002544f40] Lossless check failed - expected 07, calculated 04.
[truehd @ 0000000002544f40] Lossless check failed - expected 06, calculated 04.
[truehd @ 0000000002544f40] Lossless check failed - expected 00, calculated 02.
[truehd @ 0000000002544f40] Lossless check failed - expected 01, calculated 03.
MD5=cba280ea373fa6287016c9bca7900bac
size=       0kB time=00:00:01.99 bitrate=   0.1kbits/s speed= 104x
video:0kB audio:346kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown

The resulting stream is also for some reason reported as 24bit (s32) instead of 16bit.


          #6330: Strange behavior with native FFmpeg AAC encoder, when converting audio to mono   

Summary of the bug:

When converting stereo to mono, even it seems to really convert to mono, it's been reported as stereo and with a strange bitrate (not CBR??) by MediaInfo? and VLC.

How to reproduce:

% ffmpeg -i input -c:v libx264 -b:v 750k -profile:v baseline -level 3.0 -pix_fmt yuv420p -c:a aac -b:a 64k -ac 1 output
ffmpeg version FFmpeg version: 20170330-ad7aff0

MediaInfo? and VLC report stereo audio.

MediaInfo? also reports a strange 64.5 kbps bitrate (when I wrote 64k), with 2 channels, 1 original channel and Front C position:

Audio
ID                                       : 2
Format                                   : AAC
Format/Info                              : Advanced Audio Codec
Format profile                           : LC
Codec ID                                 : 40
Duration                                 : 1min.
Duration_LastFrame                       : -11ms
Bit rate mode                            : Constante
Bit rate                                 : 64,5Kbps
Channel(s)                               : 2canales
Channel(s)_Original                      : 1canal
Channel positions                        : Front: C
Sampling rate                            : 48,0KHz
Stream size                              : 472Kib (8%)
Language                                 : Español

Testing other bitrates: 48k -> 48Kbps (Constant) 64k -> 64,5Kbps (Constant) 96k -> 96,0Kbps (Constant) 112k -> 114Kbps (Constant) 128k -> 129Kbps (Constant) 192k -> 159Kbps and Variable (with a max bitrate of 192Kbps)

So, it makes me think maybe it's not really CBR?

And althought "-b:a" is supossed to make CBR, when compressing only audio, MediaInfo? always reports Variable Bitrate, so no bitrate is visible with it.

ffmpeg -i "1min stereo.wav" -c:a aac -b:a 64k -ac 1 "1min 64kb mono.aac"
Audio
Formato                                  : AAC
Formato/Info                             : Advanced Audio Codec
Formato de la versión                    : Version 4
Formato del perfil                       : LC
Tipo de tasa de bits                     : Variable
Canal(es)                                : 1canal
Posiciones del canal                     : Front: C
Velocidad de muestreo                    : 48,0KHz
Tamaño de pista                          : 491Kib (100%)

          #6351: encodec lossy audio LFE how to do not to process   

ffmpeg -i "F:\8.flac" -af "pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c4|c5=c5|c6=c6|c7=c7" "F:\8.wav"

ffmpeg -i "F:\8.flac" -af "pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c4|c5=c5|c6=c6|c7=c7" -c aac "F:\8.m4a"

ffmpeg -i "F:\8.flac" -af "pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c4|c5=c5|c6=c6|c7=c7" -c dca -strict -2 "F:\8.dts"

flac & wav,LFE is no process,but aac & dca is do EQ and down volume. aac & dca,the eq is no same,dca is lower.

??? which set can do not to process LFE ???


          #1520: allow mathematical expressions for -t and -ss   

Summary of the bug:

It seems that -t and -ss support integer and decimals up to the six places after the decimal (-t 1.033366 works but -t 1.0333666 doesn't). Sometimes greater precision is needed. Is it possible to support mathematical expression with -t and -ss as it is with some of the timing in the filters.

For instance if I try to create a wav file that contains one NTSC frame's worth of silence, it will produce an error. The closest I can seem to get is with '-t 1.033367'.

How to reproduce:

ffmpeg -t '1/(30000/1001)' -f lavfi -i aevalsrc=0 -c:a pcm_s16le out.wav
ffmpeg version 0.11.1.git Copyright (c) 2000-2012 the FFmpeg developers
  built on Jul  5 2012 11:22:03 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/HEAD --enable-shared --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-libfreetype --cc=/usr/bin/gcc-4.2 --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libass --enable-libvo-aacenc
  libavutil      51. 64.100 / 51. 64.100
  libavcodec     54. 33.100 / 54. 33.100
  libavformat    54. 15.100 / 54. 15.100
  libavdevice    54.  1.100 / 54.  1.100
  libavfilter     3.  0.101 /  3.  0.101
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
Invalid duration specification for t: 1/(30000/1001)

          #2493: support filter options within ffprobe   

Summary of the bug:

This ticket is inspired by comments by Nicolas George at this post: http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2013-March/141071.html.

I'd like to make use of metadata from filters within ffprobe without having to use amovie|movie to provide the input.

For instance:

ffprobe 2>/dev/null -i 038Robin.wav -af ebur128=metadata=1 -show_frames -show_format

provides

ffprobe -i 038Robin.wav -af ebur128=metadata=1 -show_frames -show_format
ffprobe version 1.1.git Copyright (c) 2007-2013 the FFmpeg developers
  built on Apr 13 2013 10:04:18 with Apple clang version 3.1 (tags/Apple/clang-318.0.61) (based on LLVM 3.1svn)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/HEAD --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=cc --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid --enable-libfreetype --enable-ffplay --enable-libcaca
  libavutil      52. 26.100 / 52. 26.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  2.100 / 55.  2.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 53.101 /  3. 53.101
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Failed to set value 'ebur128=metadata=1' for option 'af': Option not found

The workaround seems to be to use lavfi as the input:

ffprobe 2>/dev/null -f lavfi amovie=038Robin.wav,ebur128=metadata=1 -show_frames -show_format

However although the latter commands provides the frame metadata that I want, the data provided by -show_format refers to lavfi and not the original file input.

Is it feasible to support -vf and -af in ffprobe to allow something like:

ffprobe -i 038Robin.wav -af ebur128=metadata=1 -show_frames -show_format

          #4175: support phase shift for Dolby Pro Logic II / Dolby matrix downmix   

Summary of the bug:

Hello guys. I am trying to downmix a 5.1 audio to 2.0 (stereo) using the audio resampler filter "matrix_encoding=dplii". The resulting stereo audio presents some audio level bias towards the right speaker when there is more than one active channel on the surround stream. That is, it seems that the audio volume on the right speaker is higher than it should be. This is, of course, using a test signal where all surround channels produce the same sound at the same volume.

How to reproduce:

% ffmpeg -i ChID-BLITS-EBU-Narration441-16b.wav -ac 2 -filter:a "aresample=matrix_encoding=dplii" dplii.wav
ffmpeg version 2.4.4 Copyright (c) 2000-2014 the FFmpeg developers
  built on Dec  4 2014 14:13:11 with Apple LLVM version 6.0 (clang-600.0.56) (based on LLVM 3.5svn)

You may freely obtain the surround test file I used on the command above from Fraunhoffer's website. Direct link to the surround wave file: https://www2.iis.fraunhofer.de/AAC/ChID-BLITS-EBU-Narration441-16b.wav Fraunhoffer's page where that file's URL is located: https://www2.iis.fraunhofer.de/AAC/multichannel.html

If you analyze both the surround input file and the resulting stereo wave file on Audacity, for example, you will notice that by 19 seconds into the audio stream all channels are active with a test signal. The same point in time on the resulting DPL-II downmixed shows a "higher volume" on the right speaker. This effect is even more noticeable by 22 seconds when they change the test signal. It is interesting to note that in the initial seconds of the file, when a male's voice is announcing each channel individually, I could not notice this right speaker bias. The channel downmixing in that segment sounds correct. This effect is very noticeable on sound tracks with very active surround channels, they all seem to be stronger on the right speaker.

Thank you.


          #4998: SanDisk Clip Sport/Jam is buggy, support generating "simpler" mp4 files that can be played by it.   

Hi m4a files created by FFmpeg aren't supported by SanDisk? Clip Sport/Jam?. I've attached 3 files.

1 nero.m4a This plays OK. Created with neroAacEnc version 1.5.4.0 neroAacEnc -if foo.wav -of nero.m4a

2 FFmpeg.m4a This doesn't play. Created with FFmpeg ffmpeg version git-2015-10-28-cceed83 FFmpeg -i foo.wav -c:a aac -strict -2 FFmpeg.m4a

3 nero2.m4a This doesn't play either (nero file re-muxed by FFmpeg). ffmpeg -i nero.m4a -c copy nero2.m4a

So I think that it is not FFmpeg's aac encoder that is causing the problem, but maybe the muxing into m4a by FFmpeg.

My operating system Ubuntu-12.04. The mp3 player is Clip Jam 8GB model number SDMX26-008G-G46K


          #5239: support options for MPEGTS muxer when using RTP_MPEGTS   

I'm trying to produce a MPEG TS over RTP with a constant muxrate. I am able produce a MPEG TS with constant muxrate over UDP using the "mpegts" output format (as shown in the first pasted console output below), but after changing the command to use "rtp_mpegts" and a RTP:// address, the output bit rate simply follows the A/V rate. I.E the -muxrate option is ignored and the null packets are not inserted. Having compared the outputs using a TS analyzer, I've also noticed that the -streamid and -metadata options are also ignored because the PIDs and service names are different. Perhaps all options are being ignored in that case?

COMMAND 1: MPEGTS over UDP

c:\ffmpeg-2-8-3\bin\ffmpeg -loop 1 -re -i HOLPGKO01/bars/NFLbars%03d.jpg -f lavfi -i "amovie=holpgko01/tone/tone.wav:loop=0, asetpts=N/(FRAME_RATE*TB)" -c:v libx264 -profile:v high -pix_fmt yuvj420p -g 5 -b:v 1M -minrate 1M -maxrate 1M -bufsize 512k -b:a 384k -map 0:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -f mpegts -streamid 0:31 -streamid 1:32 -streamid 2:33 -streamid 3:34 -streamid 4:35 -muxrate 8M -mpegts_pmt_start_pid 0x001E -metadata service_provider="Provider" -metadata service_name="Service 1" udp://239.118.0.20:11820?pkt_size=1316

ffmpeg version 2.8.3 Copyright (c) 2000-2015 the FFmpeg developers

built with gcc 5.2.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av

isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca -- enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena ble-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc

--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enabl

e-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --ena ble-decklink --enable-zlib

libavutil 54. 31.100 / 54. 31.100 libavcodec 56. 60.100 / 56. 60.100 libavformat 56. 40.101 / 56. 40.101 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 40.101 / 5. 40.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.101 / 1. 2.101 libpostproc 53. 3.100 / 53. 3.100

[mjpeg @ 0000000000328100] Changeing bps to 8 Input #0, image2, from 'HOLPGKO01/bars/NFLbars%03d.jpg':

Duration: 00:00:12.60, start: 0.000000, bitrate: N/A

Stream #0:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 1920x1080

[SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 25 tbc [Parsed_amovie_0 @ 00000000056cf0a0] Channel layout is not set in output stream 0, guessed channel layout is 'stereo' Input #1, lavfi, from 'amovie=holpgko01/tone/tone.wav:loop=0, asetpts=N/(FRAME _RATE*TB)':

Duration: N/A, bitrate: 1536 kb/s

Stream #1:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s

[libx264 @ 0000000000335ba0] using SAR=1/1 [libx264 @ 0000000000335ba0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2 [libx264 @ 0000000000335ba0] profile High, level 4.0 Output #0, mpegts, to 'udp://239.118.0.20:11820?pkt_size=1316':

Metadata:

service_provider: Provider service_name : Service 1 encoder : Lavf56.40.101 Stream #0:0: Video: h264 (libx264), yuvj420p(pc), 1920x1080 [SAR 1:1 DAR 16:

9], q=-1--1, 1000 kb/s, 25 fps, 90k tbn, 25 tbc

Metadata:

encoder : Lavc56.60.100 libx264

Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 mp2

Stream #0:2: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 mp2

Stream #0:3: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 mp2

Stream #0:4: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 mp2

Stream mapping:

Stream #0:0 -> #0:0 (mjpeg (native) -> h264 (libx264)) Stream #1:0 -> #0:1 (pcm_s16le (native) -> mp2 (native)) Stream #1:0 -> #0:2 (pcm_s16le (native) -> mp2 (native)) Stream #1:0 -> #0:3 (pcm_s16le (native) -> mp2 (native)) Stream #1:0 -> #0:4 (pcm_s16le (native) -> mp2 (native))

Press [q] to stop, ? for help [image2 @ 00000000003251e0] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8) frame= 37 fps= 36 q=33.0 size= 509kB time=00:00:00.52 bitrate=8011.7kbits/ frame= 49 fps= 32 q=33.0 size= 1017kB time=00:00:01.00 bitrate=8330.7kbits/ frame= 62 fps= 30 q=33.0 size= 1485kB time=00:00:01.52 bitrate=8004.8kbits/ frame= 74 fps= 29 q=33.0 size= 1994kB time=00:00:02.00 bitrate=8142.4kbits/ frame= 87 fps= 29 q=33.0 size= 2501kB time=00:00:02.52 bitrate=8128.8kbits/ frame= 100 fps= 28 q=32.0 size= 2973kB time=00:00:03.04 bitrate=8012.3kbits/ frame= 112 fps= 28 q=32.0 size= 3478kB time=00:00:03.52 bitrate=8093.4kbits/ frame= 125 fps= 27 q=33.0 size= 3950kB time=00:00:04.04 bitrate=8008.8kbits/ frame= 137 fps= 27 q=33.0 size= 4454kB time=00:00:04.52 bitrate=8061.4kbits/ frame= 150 fps= 27 q=32.0 size= 4962kB time=00:00:05.04 bitrate=8064.6kbits/ frame= 163 fps= 27 q=30.0 size= 5430kB time=00:00:05.56 bitrate=8001.0kbits/ frame= 175 fps= 27 q=31.0 size= 5938kB time=00:00:06.04 bitrate=8053.9kbits/ frame= 188 fps= 27 q=30.0 size= 6407kB time=00:00:06.56 bitrate=8001.0kbits/ frame= 200 fps= 26 q=31.0 size= 6915kB time=00:00:07.04 bitrate=8039.4kbits/ frame= 213 fps= 26 q=33.0 size= 7439kB time=00:00:07.56 bitrate=8061.3kbits/ frame= 226 fps= 26 q=32.0 size= 7894kB time=00:00:08.08 bitrate=8003.2kbits/ frame= 238 fps= 26 q=32.0 size= 8415kB time=00:00:08.56 bitrate=8053.4kbits/ frame= 251 fps= 26 q=32.0 size= 8870kB time=00:00:09.08 bitrate=8002.8kbits/ frame= 263 fps= 26 q=30.0 size= 9390kB time=00:00:09.56 bitrate=8041.7kbits/ frame= 276 fps= 26 q=33.0 size= 9884kB time=00:00:10.08 bitrate=8032.4kbits/ frame= 288 fps= 26 q=31.0 size= 10368kB time=00:00:10.56 bitrate=8042.7kbits/ frame= 301 fps= 26 q=33.0 size= 10860kB time=00:00:11.08 bitrate=8029.3kbits/ frame= 314 fps= 26 q=32.0 size= 11345kB time=00:00:11.60 bitrate=8012.0kbits/ frame= 326 fps= 26 q=32.0 size= 11836kB time=00:00:12.08 bitrate=8022.9kbits/ frame= 339 fps= 26 q=31.0 size= 12322kB time=00:00:12.60 bitrate=8011.2kbits/ frame= 351 fps= 26 q=31.0 size= 12813kB time=00:00:13.08 bitrate=8024.9kbits/ frame= 364 fps= 26 q=31.0 size= 13300kB time=00:00:13.60 bitrate=8011.5kbits/ frame= 376 fps= 26 q=32.0 size= 13790kB time=00:00:14.08 bitrate=8023.2kbits/ frame= 389 fps= 26 q=32.0 size= 14299kB time=00:00:14.60 bitrate=8019.7kbits/ frame= 402 fps= 26 q=32.0 size= 14769kB time=00:00:15.12 bitrate=8001.6kbits/ frame= 408 fps= 25 q=-1.0 Lsize= 15900kB time=00:00:16.24 bitrate=8020.5kbits /s video:1574kB audio:2880kB subtitle:0kB other streams:0kB global headers:0kB muxi ng overhead: 256.951935% [libx264 @ 0000000000335ba0] frame I:82 Avg QP:20.83 size: 16417 [libx264 @ 0000000000335ba0] frame P:102 Avg QP:25.24 size: 1420 [libx264 @ 0000000000335ba0] frame B:224 Avg QP:38.17 size: 541 [libx264 @ 0000000000335ba0] consecutive B-frames: 25.2% 2.5% 6.6% 65.7% [libx264 @ 0000000000335ba0] mb I I16..4: 62.6% 29.7% 7.7% [libx264 @ 0000000000335ba0] mb P I16..4: 1.5% 0.7% 0.0% P16..4: 3.6% 0.6 % 0.2% 0.0% 0.0% skip:93.4% [libx264 @ 0000000000335ba0] mb B I16..4: 0.1% 0.1% 0.0% B16..8: 3.9% 0.3 % 0.0% direct: 0.0% skip:95.6% L0:39.1% L1:60.7% BI: 0.2% [libx264 @ 0000000000335ba0] 8x8 transform intra:29.7% inter:53.3% [libx264 @ 0000000000335ba0] coded y,uvDC,uvAC intra: 6.4% 2.1% 1.1% inter: 0.2%

0.1% 0.1%

[libx264 @ 0000000000335ba0] i16 v,h,dc,p: 82% 15% 2% 0% [libx264 @ 0000000000335ba0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 36% 19% 42% 1% 0%

0% 0% 0% 1%

[libx264 @ 0000000000335ba0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 39% 26% 11% 8% 3%

3% 3% 4% 3%

[libx264 @ 0000000000335ba0] i8c dc,h,v,p: 83% 11% 6% 0% [libx264 @ 0000000000335ba0] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0000000000335ba0] ref P L0: 94.9% 2.7% 1.7% 0.6% [libx264 @ 0000000000335ba0] ref B L0: 89.7% 10.3% [libx264 @ 0000000000335ba0] ref B L1: 91.2% 8.8% [libx264 @ 0000000000335ba0] kb/s:790.27 Exiting normally, received signal 2.

COMMAND 2: MPEGTS over RTP

c:\ffmpeg-2-8-3\bin\ffmpeg -loop 1 -re -i HOLPGKO01/bars/NFLbars%03d.jpg -f lavfi -i "amovie=holpgko01/tone/tone.wav:loop=0, asetpts=N/(FRAME_RATE*TB)" -c:v libx264 -profile:v high -pix_fmt yuvj420p -g 5 -b:v 1M -minrate 1M -maxrate 1M -bufsize 512k -b:a 384k -map 0:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -f rtp_mpegts -streamid 0:31 -streamid 1:32 -streamid 2:33 -streamid 3:34 -streamid 4:35 -muxrate 8M -mpegts_pmt_start_pid 0x001E -metadata service_provider="Provider" -metadata service_name="Service 1" rtp://239.118.0.20:11820?pkt_size=1316

ffmpeg version 2.8.3 Copyright (c) 2000-2015 the FFmpeg developers

built with gcc 5.2.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av

isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca -- enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena ble-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc

--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enabl

e-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --ena ble-decklink --enable-zlib

libavutil 54. 31.100 / 54. 31.100 libavcodec 56. 60.100 / 56. 60.100 libavformat 56. 40.101 / 56. 40.101 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 40.101 / 5. 40.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.101 / 1. 2.101 libpostproc 53. 3.100 / 53. 3.100

[mjpeg @ 0000000000328100] Changeing bps to 8 Input #0, image2, from 'HOLPGKO01/bars/NFLbars%03d.jpg':

Duration: 00:00:12.60, start: 0.000000, bitrate: N/A

Stream #0:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 1920x1080

[SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 25 tbc [Parsed_amovie_0 @ 00000000057bf0a0] Channel layout is not set in output stream 0, guessed channel layout is 'stereo' Input #1, lavfi, from 'amovie=holpgko01/tone/tone.wav:loop=0, asetpts=N/(FRAME _RATE*TB)':

Duration: N/A, bitrate: 1536 kb/s

Stream #1:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s

[libx264 @ 0000000000335ba0] using SAR=1/1 [libx264 @ 0000000000335ba0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2 [libx264 @ 0000000000335ba0] profile High, level 4.0 Output #0, rtp_mpegts, to 'rtp://239.118.0.20:11820?pkt_size=1316':

Metadata:

service_provider: Provider service_name : Service 1 encoder : Lavf56.40.101 Stream #0:0: Video: h264 (libx264), yuvj420p(pc), 1920x1080 [SAR 1:1 DAR 16:

9], q=-1--1, 1000 kb/s, 25 fps, 90k tbn, 25 tbc

Metadata:

encoder : Lavc56.60.100 libx264

Stream #0:1: Audio: aac (libvo_aacenc), 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 libvo_aacenc

Stream #0:2: Audio: aac (libvo_aacenc), 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 libvo_aacenc

Stream #0:3: Audio: aac (libvo_aacenc), 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 libvo_aacenc

Stream #0:4: Audio: aac (libvo_aacenc), 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 libvo_aacenc

Stream mapping:

Stream #0:0 -> #0:0 (mjpeg (native) -> h264 (libx264)) Stream #1:0 -> #0:1 (pcm_s16le (native) -> aac (libvo_aacenc)) Stream #1:0 -> #0:2 (pcm_s16le (native) -> aac (libvo_aacenc)) Stream #1:0 -> #0:3 (pcm_s16le (native) -> aac (libvo_aacenc)) Stream #1:0 -> #0:4 (pcm_s16le (native) -> aac (libvo_aacenc))

Press [q] to stop, ? for help [image2 @ 00000000003251e0] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8) frame= 36 fps= 35 q=33.0 size= 75kB time=00:00:00.48 bitrate=1273.5kbits/ frame= 48 fps= 31 q=31.0 size= 169kB time=00:00:00.96 bitrate=1444.2kbits/ frame= 61 fps= 30 q=33.0 size= 252kB time=00:00:01.48 bitrate=1392.8kbits/ frame= 74 fps= 29 q=33.0 size= 351kB time=00:00:02.00 bitrate=1436.5kbits/ frame= 86 fps= 28 q=33.0 size= 435kB time=00:00:02.48 bitrate=1438.0kbits/ frame= 99 fps= 28 q=33.0 size= 532kB time=00:00:03.00 bitrate=1453.2kbits/ frame= 111 fps= 27 q=32.0 size= 615kB time=00:00:03.48 bitrate=1446.7kbits/ frame= 124 fps= 27 q=33.0 size= 716kB time=00:00:04.00 bitrate=1466.1kbits/ frame= 137 fps= 27 q=33.0 size= 797kB time=00:00:04.52 bitrate=1444.7kbits/ frame= 149 fps= 27 q=32.0 size= 896kB time=00:00:05.00 bitrate=1468.0kbits/ frame= 162 fps= 27 q=32.0 size= 974kB time=00:00:05.52 bitrate=1445.7kbits/ frame= 174 fps= 26 q=31.0 size= 1070kB time=00:00:06.00 bitrate=1460.8kbits/ frame= 187 fps= 26 q=31.0 size= 1151kB time=00:00:06.52 bitrate=1446.4kbits/ frame= 200 fps= 26 q=31.0 size= 1242kB time=00:00:07.04 bitrate=1445.8kbits/ frame= 212 fps= 26 q=32.0 size= 1334kB time=00:00:07.52 bitrate=1453.0kbits/ frame= 225 fps= 26 q=32.0 size= 1422kB time=00:00:08.04 bitrate=1448.6kbits/ frame= 237 fps= 26 q=32.0 size= 1503kB time=00:00:08.52 bitrate=1444.9kbits/ frame= 250 fps= 26 q=32.0 size= 1603kB time=00:00:09.04 bitrate=1452.8kbits/ frame= 263 fps= 26 q=31.0 size= 1689kB time=00:00:09.56 bitrate=1447.2kbits/ frame= 275 fps= 26 q=33.0 size= 1787kB time=00:00:10.04 bitrate=1458.0kbits/ frame= 288 fps= 26 q=31.0 size= 1868kB time=00:00:10.56 bitrate=1449.2kbits/ frame= 300 fps= 26 q=33.0 size= 1967kB time=00:00:11.04 bitrate=1459.8kbits/ frame= 313 fps= 26 q=30.0 size= 2047kB time=00:00:11.56 bitrate=1450.9kbits/ frame= 325 fps= 26 q=32.0 size= 2142kB time=00:00:12.04 bitrate=1457.3kbits/ frame= 338 fps= 26 q=30.0 size= 2240kB time=00:00:12.56 bitrate=1461.0kbits/ frame= 350 fps= 26 q=31.0 size= 2321kB time=00:00:13.04 bitrate=1458.3kbits/ frame= 363 fps= 26 q=30.0 size= 2420kB time=00:00:13.56 bitrate=1462.2kbits/ frame= 376 fps= 26 q=32.0 size= 2503kB time=00:00:14.08 bitrate=1456.1kbits/ frame= 388 fps= 26 q=30.0 size= 2584kB time=00:00:14.56 bitrate=1453.9kbits/ frame= 401 fps= 26 q=32.0 size= 2686kB time=00:00:15.08 bitrate=1459.4kbits/ frame= 413 fps= 26 q=31.0 size= 2784kB time=00:00:15.56 bitrate=1465.8kbits/ frame= 415 fps= 25 q=-1.0 Lsize= 2889kB time=00:00:16.52 bitrate=1432.6kbits /s video:1595kB audio:982kB subtitle:0kB other streams:0kB global headers:0kB muxin g overhead: 12.116439% [libx264 @ 0000000000335ba0] frame I:83 Avg QP:20.79 size: 16442 [libx264 @ 0000000000335ba0] frame P:105 Avg QP:25.40 size: 1400 [libx264 @ 0000000000335ba0] frame B:227 Avg QP:38.45 size: 537 [libx264 @ 0000000000335ba0] consecutive B-frames: 25.3% 2.9% 7.2% 64.6% [libx264 @ 0000000000335ba0] mb I I16..4: 63.3% 29.0% 7.7% [libx264 @ 0000000000335ba0] mb P I16..4: 1.5% 0.6% 0.0% P16..4: 3.6% 0.5 % 0.2% 0.0% 0.0% skip:93.5% [libx264 @ 0000000000335ba0] mb B I16..4: 0.1% 0.0% 0.0% B16..8: 3.9% 0.2 % 0.0% direct: 0.0% skip:95.7% L0:39.0% L1:60.8% BI: 0.2% [libx264 @ 0000000000335ba0] 8x8 transform intra:29.1% inter:51.0% [libx264 @ 0000000000335ba0] coded y,uvDC,uvAC intra: 6.4% 2.0% 1.1% inter: 0.2%

0.1% 0.1%

[libx264 @ 0000000000335ba0] i16 v,h,dc,p: 82% 15% 2% 0% [libx264 @ 0000000000335ba0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 35% 18% 44% 1% 0%

0% 0% 0% 1%

[libx264 @ 0000000000335ba0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 39% 26% 11% 8% 3%

3% 3% 4% 3%

[libx264 @ 0000000000335ba0] i8c dc,h,v,p: 83% 11% 5% 0% [libx264 @ 0000000000335ba0] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0000000000335ba0] ref P L0: 95.1% 2.6% 1.7% 0.6% [libx264 @ 0000000000335ba0] ref B L0: 89.4% 10.6% [libx264 @ 0000000000335ba0] ref B L1: 91.4% 8.6% [libx264 @ 0000000000335ba0] kb/s:787.29 Exiting normally, received signal 2.


          #3552: Initial warnings shown when playing aac   

http://thread.gmane.org/gmane.comp.video.ffmpeg.user/51341 I will attach (the first 25 seconds of) a sample that shows warnings before the first second is decoded.

$ ffmpeg -i iv_cut.aacp -t 24 out.wav
ffmpeg version N-62354-g671f908 Copyright (c) 2000-2014 the FFmpeg developers
  built on Apr 11 2014 19:43:39 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      52. 75.100 / 52. 75.100
  libavcodec     55. 58.103 / 55. 58.103
  libavformat    55. 37.100 / 55. 37.100
  libavdevice    55. 13.100 / 55. 13.100
  libavfilter     4.  4.100 /  4.  4.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
[aac @ 0x1ef0480] channel element 2.8 is not allocated
[aac @ 0x1eef900] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'iv_cut.aacp':
  Duration: 00:00:25.03, bitrate: 32 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 32 kb/s
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf55.37.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac -> pcm_s16le)
Press [q] to stop, [?] for help
[aac @ 0x1ef0480] channel element 2.8 is not allocated
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0x1ef0480] Reserved SBR extensions is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x1ef0480] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0x1ef0480] Reserved SBR extensions is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x1ef0480] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0x1ef0480] Reserved SBR extensions is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x1ef0480] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0x1ef0480] Reserved SBR extensions is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x1ef0480] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0x1ef0480] Reserved SBR extensions is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x1ef0480] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0x1ef0480] illegal icc
[aac @ 0x1ef0480] Reserved SBR extensions is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x1ef0480] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0x1ef0480] Reserved SBR extensions is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x1ef0480] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
size=    4128kB time=00:00:24.00 bitrate=1408.5kbits/s
video:0kB audio:4128kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.001893%

          #4685: Misleading warning shown for mov file with multiple edit list entries   

FFmpeg shows a warning about missing mp3 frame size for a mov file with multiple edit list entries. The stream can be played and decoded.

$ ffmpeg -i leopard-intro.mov -map 0:3 out.wav
ffmpeg version N-73240-g55397b0 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 27.100 / 54. 27.100
  libavcodec     56. 45.101 / 56. 45.101
  libavformat    56. 39.100 / 56. 39.100
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 19.100 /  5. 19.100
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.100 /  1.  2.100
  libpostproc    53.  3.100 / 53.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x26a92a0] multiple edit list entries, a/v desync might occur, patch welcome
    Last message repeated 2 times
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x26a92a0] Could not find codec parameters for stream 1 (Video: mjpeg (jpeg / 0x6765706A), none(bt470bg/unknown/unknown), 1008x630, 2055 kb/s): unspecified pixel format
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x26a92a0] Could not find codec parameters for stream 3 (Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, 2 channels, s16p, 173 kb/s): unspecified frame size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Guessed Channel Layout for  Input Stream #0.3 : stereo
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'leopard-intro.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2007-09-23 05:43:34
    timecode        : 00:00:00:00
  Duration: 00:00:46.25, start: 0.000000, bitrate: 7847 kb/s
    Stream #0:0(eng): Video: mjpeg (jpeg / 0x6765706A), yuvj420p(pc, bt470bg/unknown/unknown), 1008x630 [SAR 72:72 DAR 8:5], 9602 kb/s, 22.66 fps, 22.66 tbr, 2266 tbn, 2266 tbc (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      encoder         : Photo - JPEG
    Stream #0:1(eng): Video: mjpeg (jpeg / 0x6765706A), none(bt470bg/unknown/unknown), 1008x630, 2055 kb/s, 24 fps, 24 tbr, 2400 tbn, 2400 tbc (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      encoder         : Photo - JPEG
    Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:00
    Stream #0:3(eng): Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, 2 channels, s16p, 173 kb/s
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
    Stream #0:4(eng): Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, stereo, s16p, 159 kb/s (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
Output #0, wav, to 'out.wav':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    ISMP            : 00:00:00:00
    ISFT            : Lavf56.39.100
    Stream #0:0(eng): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      encoder         : Lavc56.45.101 pcm_s16le
Stream mapping:
  Stream #0:3 -> #0:0 (mp3 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    2097kB time=00:00:46.31 bitrate= 370.9kbits/s
video:0kB audio:2097kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.004564%
$ ffplay -vn -ast 3 leopard-intro.mov -autoexit
ffplay version N-73240-g55397b0 Copyright (c) 2003-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 27.100 / 54. 27.100
  libavcodec     56. 45.101 / 56. 45.101
  libavformat    56. 39.100 / 56. 39.100
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 19.100 /  5. 19.100
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.100 /  1.  2.100
  libpostproc    53.  3.100 / 53.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82a0000920] multiple edit list entries, a/v desync might occur, patch welcome
    Last message repeated 2 times
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82a0000920] Could not find codec parameters for stream 1 (Video: mjpeg (jpeg / 0x6765706A), none(bt470bg/unknown/unknown), 1008x630, 2055 kb/s): unspecified pixel format
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82a0000920] Could not find codec parameters for stream 3 (Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, 2 channels, s16p, 173 kb/s): unspecified frame size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'leopard-intro.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2007-09-23 05:43:34
    timecode        : 00:00:00:00
  Duration: 00:00:46.25, start: 0.000000, bitrate: 7847 kb/s
    Stream #0:0(eng): Video: mjpeg (jpeg / 0x6765706A), yuvj420p(pc, bt470bg/unknown/unknown), 1008x630 [SAR 72:72 DAR 8:5], 9602 kb/s, 22.66 fps, 22.66 tbr, 2266 tbn, 2266 tbc (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      encoder         : Photo - JPEG
    Stream #0:1(eng): Video: mjpeg (jpeg / 0x6765706A), none(bt470bg/unknown/unknown), 1008x630, 2055 kb/s, 24 fps, 24 tbr, 2400 tbn, 2400 tbc (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      encoder         : Photo - JPEG
    Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:00
    Stream #0:3(eng): Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, 2 channels, s16p, 173 kb/s
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
    Stream #0:4(eng): Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, stereo, s16p, 159 kb/s (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
  46.37    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0

          #5573: Incorrect duration when decoding raw G.729   
$ ffmpeg -i fate-suite/act/REC03.act -acodec copy -map 0:0 -f rawvideo out.g729
$ ffmpeg -i out.g729 out.wav
ffmpeg version N-80026-g936751b Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      55. 24.100 / 55. 24.100
  libavcodec     57. 42.100 / 57. 42.100
  libavformat    57. 37.100 / 57. 37.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 45.100 /  6. 45.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[g729 @ 0x2dc93a0] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, g729, from 'out.g729':
  Duration: 00:01:05.79, start: 0.000000, bitrate: 8 kb/s
    Stream #0:0: Audio: g729, 8000 Hz, 1 channels, s16
[wav @ 0x2dcbf40] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.37.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, mono, s16, 128 kb/s
    Metadata:
      encoder         : Lavc57.42.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (g729 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    1028kB time=00:00:00.83 bitrate=10119.2kbits/s speed=6.01x
video:0kB audio:1028kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.007410%

Both input and output file have a duration of 65 seconds.


          #4085: Support PLII upmixing   

http://thread.gmane.org/gmane.comp.video.ffmpeg.user/54545

FFmpeg does support PLII down-mixing (see tickets #709 and #3455) but does not support upmixing afaict. I will attach a sample from the MythTV collection that plays surround sound on my receiver, upmixing to six channels does not produce a similar result.

$ ffmpeg -i channelsPLII.mpg -ac 6 out.wav
ffmpeg version N-67396-g90cdec5 Copyright (c) 2000-2014 the FFmpeg developers
  built on Nov  6 2014 00:16:59 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 11.100 / 54. 11.100
  libavcodec     56. 10.102 / 56. 10.102
  libavformat    56. 12.101 / 56. 12.101
  libavdevice    56.  2.100 / 56.  2.100
  libavfilter     5.  2.103 /  5.  2.103
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Input #0, mpeg, from 'channelsPLII.mpg':
  Duration: 00:00:31.82, start: 0.740000, bitrate: 462 kb/s
    Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv), 480x576 [SAR 8:5 DAR 4:3], max. 2420 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc
    Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16p, 224 kb/s
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf56.12.101
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1, s16, 4608 kb/s
    Metadata:
      encoder         : Lavc56.10.102 pcm_s16le
Stream mapping:
  Stream #0:1 -> #0:0 (mp2 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=   17915kB time=00:00:31.84 bitrate=4608.0kbits/s
video:0kB audio:17914kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000556%

MythTV and VLC contain PLII decoder that support upmixing under GPL.


          #4553: Support G.729 stereo files   

http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/193145 A user uploaded a stereo G.729 file that is currently unsupported.

$ ffmpeg -i g729stereo.wav
ffmpeg version N-72081-gec38a1b Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 23.101 / 54. 23.101
  libavcodec     56. 38.101 / 56. 38.101
  libavformat    56. 33.101 / 56. 33.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 16.101 /  5. 16.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
[g729 @ 0x2897e80] Only mono sound is supported (requested channels: 2).
[wav @ 0x2897460] Failed to open codec in av_find_stream_info
[g729 @ 0x2897e80] Only mono sound is supported (requested channels: 2).
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, wav, from 'g729stereo.wav':
  Duration: 00:00:14.14, bitrate: 8 kb/s
    Stream #0:0: Audio: g729 ([131][0][0][0] / 0x0083), 8000 Hz, 2 channels, 16 kb/s
At least one output file must be specified

          #5700: Read id3v2 in wav   

When i try to obtain the embedded CoverArt? from a WAV files, output is not generated.

Command line: ffmpeg -v 9 -loglevel 99 -i semesta.wav coverwav.jpg -report

ffmpeg version N-80694-g074fdf4 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.4.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 26.100 / 55. 26.100
  libavcodec     57. 47.100 / 57. 47.100
  libavformat    57. 40.101 / 57. 40.101
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 46.102 /  6. 46.102
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
Splitting the commandline.
Reading option '-v' ... matched as option 'v' (set logging level) with argument '9'.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument '99'.
Reading option '-i' ... matched as input file with argument 'semesta.wav'.
Reading option 'coverwav.jpg' ... matched as output file.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set logging level) with argument 9.
Applying option loglevel (set logging level) with argument 99.
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file semesta.wav.
Successfully parsed a group of options.
Opening an input file: semesta.wav.
[file @ 033ca2c0] Setting default whitelist 'file,crypto'
[wav @ 033c9940] Format wav probed with size=2048 and score=99
[wav @ 033c9940] Before avformat_find_stream_info() pos: 44 bytes read:131072 seeks:3 nb_streams:1
[wav @ 033c9940] probing stream 0 pp:32
[wav @ 033c9940] probing stream 0 pp:31
[wav @ 033c9940] probing stream 0 pp:30
[wav @ 033c9940] probing stream 0 pp:29
[wav @ 033c9940] probing stream 0 pp:28
[wav @ 033c9940] probing stream 0 pp:27
[wav @ 033c9940] probing stream 0 pp:26
[wav @ 033c9940] probing stream 0 pp:25
[wav @ 033c9940] probing stream 0 pp:24
[wav @ 033c9940] probing stream 0 pp:23
[wav @ 033c9940] probing stream 0 pp:22
[wav @ 033c9940] probing stream 0 pp:21
[wav @ 033c9940] probing stream 0 pp:20
[wav @ 033c9940] probing stream 0 pp:19
[wav @ 033c9940] probing stream 0 pp:18
[wav @ 033c9940] probing stream 0 pp:17
[wav @ 033c9940] probing stream 0 pp:16
[wav @ 033c9940] probing stream 0 pp:15
[wav @ 033c9940] probing stream 0 pp:14
[wav @ 033c9940] probing stream 0 pp:13
[wav @ 033c9940] probing stream 0 pp:12
[wav @ 033c9940] probing stream 0 pp:11
[wav @ 033c9940] probing stream 0 pp:10
[wav @ 033c9940] probing stream 0 pp:9
[wav @ 033c9940] probing stream 0 pp:8
[wav @ 033c9940] probing stream 0 pp:7
[wav @ 033c9940] probing stream 0 pp:6
[wav @ 033c9940] probing stream 0 pp:5
[wav @ 033c9940] probing stream 0 pp:4
[wav @ 033c9940] probing stream 0 pp:3
[wav @ 033c9940] probing stream 0 pp:2
[wav @ 033c9940] probing stream 0 pp:1
[wav @ 033c9940] probed stream 0
[wav @ 033c9940] parser not found for codec pcm_s16le, packets or times may be invalid.
[wav @ 033c9940] All info found
[wav @ 033c9940] After avformat_find_stream_info() pos: 204844 bytes read:360448 seeks:3 frames:50
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'semesta.wav':
  Duration: 00:04:47.47, bitrate: 1416 kb/s
    Stream #0:0, 50, 1/44100: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s
Successfully opened the file.
Parsing a group of options: output file coverwav.jpg.
Successfully parsed a group of options.
Opening an output file: coverwav.jpg.
Output #0, image2, to 'coverwav.jpg':
Output file #0 does not contain any stream
[AVIOContext @ 033ca400] Statistics: 360448 bytes read, 3 seeks

          #6140: Add support for AMR-WB+ codec   

FFmpeg currently does not have a decoder for AMR-WB+ codec.

A free encoder + decoder for AMR-WB+ can be downloaded from https://portal.3gpp.org/desktopmodules/Specifications/SpecificationDetails.aspx?specificationId=1451. The ZIP file has both binaries (encoder.exe, decoder.exe) and source code.

Here are some usage instructions,

$ wget http://www.music.helsinki.fi/tmt/opetus/uusmedia/esim/a2002011001-e02-16kHz.wav
$ wine encoder.exe -rate 16 -if a2002011001-e02-16kHz.wav -of out.3gp
3gp File Format
Encoding @  16.00kbps
 --- Running ---
$ mediainfo out.3gp
General
Complete name                            : out.3gp
Format                                   : MPEG-4
Format profile                           : 3GPP Media Release 6 Basic
Codec ID                                 : 3gp6 (3gp6/isom)
File size                                : 110 KiB
Duration                                 : 54 s 160 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 16.7 kb/s
Encoded date                             : UTC 2017-02-10 06:10:58
Tagged date                              : UTC 2017-02-10 06:10:58
Audio
ID                                       : 1
Format                                   : sawp
Codec ID                                 : sawp
Duration                                 : 54 s 160 ms
Bit rate mode                            : Constant
Bit rate                                 : 16.2 kb/s
Channel(s)                               : 2 channels
Sampling rate                            : 6 464 Hz
Bit depth                                : 16 bits
Stream size                              : 107 KiB (97%)
Encoded date                             : UTC 2017-02-10 06:10:58
Tagged date                              : UTC 2017-02-10 06:10:58
$ wine decoder.exe  -if out.3gp  -of out.wav
...
File type brand '3gp6' minor version 256
Compatible brands: 'isom' (yes) '3gp4' (no) '3gp5' (no)
Track 1 'sawp'  Duration 54.160s  Time scale 72000
Decoding @  16.00kbps
$ mediainfo out.wav
General
Complete name                            : out.wav
Format                                   : Wave
File size                                : 9.92 MiB
Duration                                 : 54 s 160 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 1 536 kb/s
Audio
Format                                   : PCM
Format settings, Endianness              : Little
Format settings, Sign                    : Signed
Codec ID                                 : 1
Duration                                 : 54 s 160 ms
Bit rate mode                            : Constant
Bit rate                                 : 1 536 kb/s
Channel(s)                               : 2 channels
Sampling rate                            : 48.0 kHz
Bit depth                                : 16 bits
Stream size                              : 9.92 MiB (100%)

It would be great to get support for AMR-WB+ codec in FFmpeg.


          #3815: libcdio only works with -ss 00   

Summary of the bug: The (undocumented) libcdio device does not work directly. How to reproduce:

ffmpeg -f libcdio -i /dev/sr0 test.wav

always end instantly and did not produce any valid file.

A possible workaround is:

ffmpeg -ss 00 -f libcdio -i /dev/sr0 test.wav

I use git version.

(I tried on another computer with ffmpeg stable release and it didn't work, too.)

Here is the output: $ ffmpeg -f libcdio -i /dev/sr0 test.wav ffmpeg version N-65021-gc2ef844 Copyright (c) 2000-2014 the FFmpeg developers

built on Jul 27 2014 14:04:31 with gcc 4.8.2 (Gentoo 4.8.2 p1.3r1, pie-0.5.8r1) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-O2 -pipe -march=native -fomit-frame-pointer' --extra-cflags='-O2 -pipe -march=native -fomit-frame-pointer' --extra-cxxflags='-O2 -pipe -march=native -fomit-frame-pointer' --disable-static --enable-gpl --enable-postproc --enable-avfilter --enable-avresample --disable-stripping --enable-nonfree --enable-version3 --enable-nonfree --disable-indev=v4l2 --disable-outdev=v4l2 --disable-indev=oss --disable-outdev=oss --enable-bzlib --disable-runtime-cpudetect --disable-debug --disable-doc --enable-gnutls --enable-hardcoded-tables --enable-iconv --enable-network --enable-openssl --disable-libsmbclient --enable-ffplay --enable-vaapi --enable-vdpau --enable-xlib --enable-zlib --enable-libvo-aacenc --disable-libvo-amrwbenc --enable-libmp3lame --enable-libaacplus --disable-libfaac --enable-libtheora --disable-libtwolame --enable-libwavpack --disable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libcdio --disable-libiec61883 --disable-libdc1394 --disable-libcaca --disable-openal --enable-opengl --disable-libv4l2 --disable-libpulse --enable-x11grab --disable-libbs2b --disable-libflite --disable-frei0r --disable-libfribidi --disable-fontconfig --enable-ladspa --disable-libass --enable-libfreetype --disable-libsoxr --enable-pthreads --disable-libopencore-amrwb --disable-libopencore-amrnb --disable-libfdk-aac --disable-libopenjpeg --enable-libbluray --enable-libcelt --disable-libgme --disable-libgsm --disable-libmodplug --enable-libopus --enable-libquvi --disable-librtmp --disable-libssh --disable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-libzvbi --disable-armv5te --disable-armv6 --disable-armv6t2 --disable-neon --disable-vfp --disable-mips32r2 --disable-mipsdspr1 --disable-mipsdspr2 --disable-mipsfpu --disable-altivec --disable-amd3dnow --disable-amd3dnowext --disable-avx --disable-avx2 --disable-fma3 --disable-fma4 --disable-sse3 --disable-sse4 --disable-sse42 --disable-xop --cpu=host libavutil 52. 92.101 / 52. 92.101 libavcodec 55. 69.100 / 55. 69.100 libavformat 55. 49.100 / 55. 49.100 libavdevice 55. 13.102 / 55. 13.102 libavfilter 4. 11.102 / 4. 11.102 libavresample 1. 3. 0 / 1. 3. 0 libswscale 2. 6.100 / 2. 6.100 libswresample 0. 19.100 / 0. 19.100 libpostproc 52. 3.100 / 52. 3.100

Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, libcdio, from '/dev/sr0':

Duration: 00:39:21.61, bitrate: N/A

Chapter #0.0: start 0.000000, end 89.360000 Metadata:

title : track 01

Chapter #0.1: start 89.360000, end 100.453333 Metadata:

title : track 02

Chapter #0.2: start 100.453333, end 346.680000 Metadata:

title : track 03

Chapter #0.3: start 346.680000, end 414.746667 Metadata:

title : track 04

Chapter #0.4: start 414.746667, end 576.333333 Metadata:

title : track 05

Chapter #0.5: start 576.333333, end 631.093333 Metadata:

title : track 06

Chapter #0.6: start 631.093333, end 723.906667 Metadata:

title : track 07

Chapter #0.7: start 723.906667, end 771.813333 Metadata:

title : track 08

Chapter #0.8: start 771.813333, end 862.746667 Metadata:

title : track 09

Chapter #0.9: start 862.746667, end 940.346667 Metadata:

title : track 10

Chapter #0.10: start 940.346667, end 1079.186667 Metadata:

title : track 11

Chapter #0.11: start 1079.186667, end 1190.653333 Metadata:

title : track 12

Chapter #0.12: start 1190.653333, end 1238.746667 Metadata:

title : track 13

Chapter #0.13: start 1238.746667, end 1393.866667 Metadata:

title : track 14

Chapter #0.14: start 1393.866667, end 1467.586667 Metadata:

title : track 15

Chapter #0.15: start 1467.586667, end 1548.733333 Metadata:

title : track 16

Chapter #0.16: start 1548.733333, end 1595.493333 Metadata:

title : track 17

Chapter #0.17: start 1595.493333, end 1720.413333 Metadata:

title : track 18

Chapter #0.18: start 1720.413333, end 1801.480000 Metadata:

title : track 19

Chapter #0.19: start 1801.480000, end 1932.146667 Metadata:

title : track 20

Chapter #0.20: start 1932.146667, end 1969.053333 Metadata:

title : track 21

Chapter #0.21: start 1969.053333, end 2135.746667 Metadata:

title : track 22

Chapter #0.22: start 2135.746667, end 2176.933333 Metadata:

title : track 23

Chapter #0.23: start 2176.933333, end 2361.613333 Metadata:

title : track 24

Stream #0:0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s

Output #0, wav, to 'test.wav':

Metadata:

ISFT : Lavf55.49.100 Chapter #0.0: start 0.000000, end 89.360000 Metadata:

title : track 01

Chapter #0.1: start 89.360000, end 100.453333 Metadata:

title : track 02

Chapter #0.2: start 100.453333, end 346.680000 Metadata:

title : track 03

Chapter #0.3: start 346.680000, end 414.746667 Metadata:

title : track 04

Chapter #0.4: start 414.746667, end 576.333333 Metadata:

title : track 05

Chapter #0.5: start 576.333333, end 631.093333 Metadata:

title : track 06

Chapter #0.6: start 631.093333, end 723.906667 Metadata:

title : track 07

Chapter #0.7: start 723.906667, end 771.813333 Metadata:

title : track 08

Chapter #0.8: start 771.813333, end 862.746667 Metadata:

title : track 09

Chapter #0.9: start 862.746667, end 940.346667 Metadata:

title : track 10

Chapter #0.10: start 940.346667, end 1079.186667 Metadata:

title : track 11

Chapter #0.11: start 1079.186667, end 1190.653333 Metadata:

title : track 12

Chapter #0.12: start 1190.653333, end 1238.746667 Metadata:

title : track 13

Chapter #0.13: start 1238.746667, end 1393.866667 Metadata:

title : track 14

Chapter #0.14: start 1393.866667, end 1467.586667 Metadata:

title : track 15

Chapter #0.15: start 1467.586667, end 1548.733333 Metadata:

title : track 16

Chapter #0.16: start 1548.733333, end 1595.493333 Metadata:

title : track 17

Chapter #0.17: start 1595.493333, end 1720.413333 Metadata:

title : track 18

Chapter #0.18: start 1720.413333, end 1801.480000 Metadata:

title : track 19

Chapter #0.19: start 1801.480000, end 1932.146667 Metadata:

title : track 20

Chapter #0.20: start 1932.146667, end 1969.053333 Metadata:

title : track 21

Chapter #0.21: start 1969.053333, end 2135.746667 Metadata:

title : track 22

Chapter #0.22: start 2135.746667, end 2176.933333 Metadata:

title : track 23

Chapter #0.23: start 2176.933333, end 2361.613333 Metadata:

title : track 24

Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s Metadata:

encoder : Lavc55.69.100 pcm_s16le

Stream mapping:

Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))

Press [q] to stop, ? for help size= 0kB time=00:00:00.00 bitrate=N/A video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)


          #5297: mpeg-4 als: "predictor order too large"   

http://www.datafilehost.com/d/e51ea572

C:\>mp4alsRM23.exe -o1023 -MP4 pred.wav
C:\>ffmpeg -i pred.mp4 out.wav
ffmpeg version N-78758-g5156578 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
ble-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --
enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-lib
x265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-d
ecklink --enable-zlib
  libavutil      55. 19.100 / 55. 19.100
  libavcodec     57. 27.100 / 57. 27.100
  libavformat    57. 26.100 / 57. 26.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 37.100 /  6. 37.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'pred.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    creation_time   : 2016-03-06 18:17:21
  Duration: 00:00:30.01, start: 0.000000, bitrate: 916 kb/s
    Stream #0:0(und): Audio: mp4als (mp4a / 0x6134706D), 44100 Hz, 2 channels, s
16, 916 kb/s (default)
    Metadata:
      creation_time   : 2016-03-06 18:17:21
Output #0, wav, to 'out.wav':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    ISFT            : Lavf57.26.100
    Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo
, s16, 1411 kb/s (default)
    Metadata:
      creation_time   : 2016-03-06 18:17:21
      encoder         : Lavc57.27.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (mp4als (als) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[output stream 0:0 @ 038f2740] 100 buffers queued in output stream 0:0, somethin
g may be wrong.
[als @ 038f7020] Predictor order too large.
[als @ 038f7020] Reading frame data failed. Skipping RA unit.
size=    5171kB time=00:00:30.01 bitrate=1411.2kbits/s speed=9.02x
video:0kB audio:5170kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.001473%

          #5671: FFmpeg unable to decode PCM in MXF   

Summary of the bug: FFmpeg is unable to decode PCM which is wrapped in an MXF file. The MXF file is Avid compatible, but was apparently not created with Avid. Sample can be found here: http://dl.flavoursys.com/samples/ffmpeg/TVOKids_6.mxf

How to reproduce:

% $ ffmpeg -i TVOKids_6.mxf -loglevel debug -y out.wav                                                                                                                                                                                                 [17:02:14]
ffmpeg version N-80780-gd693392 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.1 (Ubuntu 5.3.1-14ubuntu2.1) 20160413
  configuration: --extra-libs=-ldl --enable-gpl --enable-nonfree --enable-version3 --enable-libfdk-aac --enable-libx264 --enable-libfreetype
  libavutil      55. 27.100 / 55. 27.100
  libavcodec     57. 48.101 / 57. 48.101
  libavformat    57. 40.101 / 57. 40.101
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 46.102 /  6. 46.102
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
Splitting the commandline.
Reading option '-i' ... matched as input file with argument 'TVOKids_6.mxf'.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option 'out.wav' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file TVOKids_6.mxf.
Successfully parsed a group of options.
Opening an input file: TVOKids_6.mxf.
[file @ 0x362acc0] Setting default whitelist 'file,crypto'
[mxf @ 0x362a380] Format mxf probed with size=2048 and score=100
[mxf @ 0x362a380] Dark key 06.0e.2b.34.02.53.01.01.0d.01.01.01.01.01.23.00
[mxf @ 0x362a380] Dark key 06.0e.2b.34.01.01.01.02.03.01.02.10.01.00.00.00
    Last message repeated 2 times
[mxf @ 0x362a380] Dark key 06.0e.2b.34.02.05.01.01.0d.01.02.01.01.11.01.00
[mxf @ 0x362a380] none: Universal Label: 00000000.0000.0000.00000000.00000000
[mxf @ 0x362a380] Before avformat_find_stream_info() pos: 32744 bytes read:66172 seeks:5 nb_streams:1
[mxf @ 0x362a380] IndexSID 1 segment at 0 missing EditUnitByteCount and IndexEntryArray
[mxf @ 0x362a380] After avformat_find_stream_info() pos: 32744 bytes read:66172 seeks:5 frames:0
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, mxf, from 'TVOKids_6.mxf':
  Metadata:
    uid             : 27023931-e9a3-4549-946d-a5aea01cec3e
    generation_uid  : bd18e86b-ee7a-4f8f-8ceb-7ab5ac3f175f
    company_name    : Matrox Electronic
    product_name    : DSX
    product_version : 4.0
    product_uid     : 060e2b34-0401-0109-0e0e-010101000001
    modification_date: 2016-05-31 17:30:17
    material_package_umid: 0x060A2B340101010501010D20130000004F5296D2A16741ED9D11A972EA7C9C30
    material_package_name: 0004PE
    timecode        : 11:22:33;00
  Duration: 00:00:18.95, start: 0.000000, bitrate: 1166 kb/s
    Stream #0:0, 0, 1/48000: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s
    Metadata:
      file_package_umid: 0x060A2B340101010501010D20130000000B45AF7186674D708C09A0FCF3D5B364
Successfully opened the file.
Parsing a group of options: output file out.wav.
Successfully parsed a group of options.
Opening an output file: out.wav.
[file @ 0x3631540] Setting default whitelist 'file,crypto'
Successfully opened the file.
detected 16 logical cores
[graph 0 input from stream 0:0 @ 0x3643980] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0x3643980] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0x3643980] Setting 'sample_fmt' to value 's32'
[graph 0 input from stream 0:0 @ 0x3643980] Setting 'channel_layout' to value '0x4'
[graph 0 input from stream 0:0 @ 0x3643980] tb:1/48000 samplefmt:s32 samplerate:48000 chlayout:0x4
[audio format for output stream 0:0 @ 0x3644560] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0x3644560] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
[AVFilterGraph @ 0x3631b80] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0x3645d40] [SWR @ 0x3646120] Using fltp internally between filters
[auto-inserted resampler 0 @ 0x3645d40] ch:1 chl:mono fmt:s32 r:48000Hz -> ch:1 chl:mono fmt:s16 r:48000Hz
[wav @ 0x362fa00] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    uid             : 27023931-e9a3-4549-946d-a5aea01cec3e
    generation_uid  : bd18e86b-ee7a-4f8f-8ceb-7ab5ac3f175f
    company_name    : Matrox Electronic
    product_name    : DSX
    product_version : 4.0
    product_uid     : 060e2b34-0401-0109-0e0e-010101000001
    modification_date: 2016-05-31 17:30:17
    material_package_umid: 0x060A2B340101010501010D20130000004F5296D2A16741ED9D11A972EA7C9C30
    material_package_name: 0004PE
    ISMP            : 11:22:33;00
    ISFT            : Lavf57.40.101
    Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16 (24 bit), 768 kb/s
    Metadata:
      file_package_umid: 0x060A2B340101010501010D20130000000B45AF7186674D708C09A0FCF3D5B364
      encoder         : Lavc57.48.101 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s24le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[mxf @ 0x362a380] IndexSID 1 segment at 0 missing EditUnitByteCount and IndexEntryArray
TVOKids_6.mxf: Invalid data found when processing input
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    Last message repeated 1 times
[output stream 0:0 @ 0x36444c0] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
size=       0kB time=00:00:00.00 bitrate=N/A speed=   0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Input file #0 (TVOKids_6.mxf):
  Input stream #0:0 (audio): 0 packets read (0 bytes); 0 frames decoded (0 samples);
  Total: 0 packets (0 bytes) demuxed
Output file #0 (out.wav):
  Output stream #0:0 (audio): 0 frames encoded (0 samples); 0 packets muxed (0 bytes);
  Total: 0 packets (0 bytes) muxed
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x3631640] Statistics: 4 seeks, 3 writeouts
[AVIOContext @ 0x3633020] Statistics: 66172 bytes read, 5 seeks

          #6106: Some pcm wavs detected as eac3   

Summary of the bug:

I use ffmpeg extensively to identify file formats.

There are some pcm wavs that are mis-identified as varieties of ac3 - recompiling with --disable-demuxer=ac3 fixes it, but that's not nice if you also want to be able to deal with ac3 audio.

This was not a case with a very old FFmpeg I have on a different machine (N-34304-gc0dbab9, built Oct 31 2011), but bisecting 5 years seems... unproductive.

How to reproduce:

Compile ffmpeg (git master) with default settings, and run ffmpeg -i with the file I'm supplying.

$ ./ffmpeg -i pcm_wav_misidentified_as_eac3.wav
ffmpeg version N-83248-gb4a13d442a Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  configuration:
  libavutil      55. 45.100 / 55. 45.100
  libavcodec     57. 75.100 / 57. 75.100
  libavformat    57. 65.100 / 57. 65.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 70.100 /  6. 70.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
Input #0, wav, from 'pcm_wav_misidentified_as_eac3.wav':
  Duration: 00:00:05.00, bitrate: 1411 kb/s
    Stream #0:0: Audio: eac3 ([1][0][0][0] / 0x0001), 48000 Hz, 5.0(side), fltp, 64 kb/s (visual impaired)
At least one output file must be specified

Built Jan 25 13:03 CET.

Additional comment:

Based on what I read on http://soundfile.sapp.org/doc/WaveFormat/ the stream shouldn't even have to be interpreted by all different codecs as long as the AudioFormat? field (at bytes 20-21) is 0x0001 ? Just a thought...

Full output:

$ ./ffmpeg -v 9 -loglevel 99 -report -i pcm_wav_misidentified_as_eac3.wav
ffmpeg started on 2017-01-27 at 11:50:13
Report written to "ffmpeg-20170127-115013.log"
ffmpeg version N-83248-gb4a13d442a Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  configuration:
  libavutil      55. 45.100 / 55. 45.100
  libavcodec     57. 75.100 / 57. 75.100
  libavformat    57. 65.100 / 57. 65.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 70.100 /  6. 70.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
Splitting the commandline.
Reading option '-v' ... matched as option 'v' (set logging level) with argument '9'.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument '99'.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Reading option '-i' ... matched as input url with argument 'pcm_wav_misidentified_as_eac3.wav'.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set logging level) with argument 9.
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url pcm_wav_misidentified_as_eac3.wav.
Successfully parsed a group of options.
Opening an input file: pcm_wav_misidentified_as_eac3.wav.
[file @ 0x7fb764d02680] Setting default whitelist 'file,crypto'
Probing mp3 score:1 size:2048
Probing wav score:99 size:2048
[wav @ 0x7fb765800000] Format wav probed with size=2048 and score=99
[wav @ 0x7fb765800000] Before avformat_find_stream_info() pos: 44 bytes read:65580 seeks:1 nb_streams:1
[wav @ 0x7fb765800000] probing stream 0 pp:32
Probing mp3 score:1 size:4096
[wav @ 0x7fb765800000] Probe with size=4096, packets=2469 detected mp3 with score=1
[wav @ 0x7fb765800000] probing stream 0 pp:31
Probing mp3 score:1 size:8192
[wav @ 0x7fb765800000] Probe with size=8192, packets=2470 detected mp3 with score=1
[wav @ 0x7fb765800000] probing stream 0 pp:30
[wav @ 0x7fb765800000] probing stream 0 pp:29
Probing mp3 score:1 size:16384
[wav @ 0x7fb765800000] Probe with size=16384, packets=2472 detected mp3 with score=1
[wav @ 0x7fb765800000] probing stream 0 pp:28
[wav @ 0x7fb765800000] probing stream 0 pp:27
[wav @ 0x7fb765800000] probing stream 0 pp:26
[wav @ 0x7fb765800000] probing stream 0 pp:25
[wav @ 0x7fb765800000] probing stream 0 pp:24
[wav @ 0x7fb765800000] probing stream 0 pp:23
[wav @ 0x7fb765800000] probing stream 0 pp:22
[wav @ 0x7fb765800000] probing stream 0 pp:21
[wav @ 0x7fb765800000] probing stream 0 pp:20
[wav @ 0x7fb765800000] probing stream 0 pp:19
[wav @ 0x7fb765800000] probing stream 0 pp:18
[wav @ 0x7fb765800000] probing stream 0 pp:17
Probing ac3 score:25 size:65536
[wav @ 0x7fb765800000] Probe with size=65536, packets=2484 detected ac3 with score=25
[wav @ 0x7fb765800000] probing stream 0 pp:16
[wav @ 0x7fb765800000] probing stream 0 pp:15
[wav @ 0x7fb765800000] probing stream 0 pp:14
[wav @ 0x7fb765800000] probing stream 0 pp:13
[wav @ 0x7fb765800000] probing stream 0 pp:12
[wav @ 0x7fb765800000] probing stream 0 pp:11
[wav @ 0x7fb765800000] probing stream 0 pp:10
[wav @ 0x7fb765800000] probing stream 0 pp:9
[wav @ 0x7fb765800000] probing stream 0 pp:8
[wav @ 0x7fb765800000] probing stream 0 pp:7
[wav @ 0x7fb765800000] probing stream 0 pp:6
[wav @ 0x7fb765800000] probing stream 0 pp:5
[wav @ 0x7fb765800000] probing stream 0 pp:4
[wav @ 0x7fb765800000] probing stream 0 pp:3
[wav @ 0x7fb765800000] probing stream 0 pp:2
[wav @ 0x7fb765800000] probing stream 0 pp:1
Probing ac3 score:50 size:131072
[wav @ 0x7fb765800000] Probe with size=131072, packets=2500 detected ac3 with score=50
[wav @ 0x7fb765800000] probed stream 0
[wav @ 0x7fb765800000] stream 0: start_time: -209146758205323.719 duration: 5.000
[wav @ 0x7fb765800000] format: start_time: -9223372036854.775 duration: 5.000 bitrate=1411 kb/s
[wav @ 0x7fb765800000] After avformat_find_stream_info() pos: 882044 bytes read:947580 seeks:1 frames:16
Input #0, wav, from 'pcm_wav_misidentified_as_eac3.wav':
  Duration: 00:00:05.00, bitrate: 1411 kb/s
    Stream #0:0, 16, 1/44100: Audio: eac3 ([1][0][0][0] / 0x0001), 48000 Hz, 5.0(side), fltp, 64 kb/s (visual impaired)
Successfully opened the file.
At least one output file must be specified
[AVIOContext @ 0x7fb764d028c0] Statistics: 947580 bytes read, 1 seeks

          #6112: ffmpeg not responding, stuck at converting flac to wav file   

I am trying to convert a flac to a wav file. Unfortunately the ffmpeg is getting stuck in the conversion process. I have tried with multiple ffmpeg versions (including the last one) and the same issue occurs.

I have attached the source flac file (AFN8429768236582_01_11_FLAC.flac)

How to reproduce:

% ffmpeg.exe -i AFN8429768236582_01_11_FLAC.flac -y -ar 44100 -ac 2 -sample_fmt s16 AFN8429768236582_01_11_wav.wav
ffmpeg version N-67063-g282c935 Copyright (c) 2000-2014 the FFmpeg developers
  built on Oct 20 2014 22:01:37 with gcc 4.9.1 (GCC)

Below if the output with the debug log level:

ffmpeg.exe -v 9 -loglevel 99 -i AFN8429768236582_01_11_FLAC.flac -y -ar 44100 -ac 2 -sample_fmt s16 AFN8429768236582_01_11_wav.wav
ffmpeg version N-67063-g282c935 Copyright (c) 2000-2014 the FFmpeg developers
  built on Oct 20 2014 22:01:37 with gcc 4.9.1 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-
libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --e
nable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libw
ebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      54. 10.100 / 54. 10.100
  libavcodec     56.  8.102 / 56.  8.102
  libavformat    56.  9.101 / 56.  9.101
  libavdevice    56.  1.100 / 56.  1.100
  libavfilter     5.  2.100 /  5.  2.100
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Splitting the commandline.
Reading option '-v' ... matched as option 'v' (set logging level) with argument '9'.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument '99'.
Reading option '-i' ... matched as input file with argument 'AFN8429768236582_01_11_FLAC.flac'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '44100'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'.
Reading option '-sample_fmt' ... matched as option 'sample_fmt' (set sample format) with argument 's16'.
Reading option 'AFN8429768236582_01_11_wav.wav' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set logging level) with argument 9.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file AFN8429768236582_01_11_FLAC.flac.
Successfully parsed a group of options.
Opening an input file: AFN8429768236582_01_11_FLAC.flac.
[flac @ 04485720] Format flac probed with size=2048 and score=50
[flac @ 04485720] Before avformat_find_stream_info() pos: 8288 bytes read:32768 seeks:0
[flac @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 14 times
[flac @ 04485720] All info found
[flac @ 04485720] After avformat_find_stream_info() pos: 73728 bytes read:98304 seeks:0 frames:1
Input #0, flac, from 'AFN8429768236582_01_11_FLAC.flac':
  Metadata:
    ENCODER         : Lavf55.12.100
  Duration: 00:01:54.81, start: 0.000000, bitrate: 830 kb/s
    Stream #0:0, 1, 1/44100: Audio: flac, 44100 Hz, stereo, s16
Successfully opened the file.
Parsing a group of options: output file AFN8429768236582_01_11_wav.wav.
Applying option ar (set audio sampling rate (in Hz)) with argument 44100.
Applying option ac (set number of audio channels) with argument 2.
Applying option sample_fmt (set sample format) with argument s16.
Successfully parsed a group of options.
Opening an output file: AFN8429768236582_01_11_wav.wav.
Successfully opened the file.
detected 8 logical cores
[graph 0 input from stream 0:0 @ 0446e820] Setting 'time_base' to value '1/44100'
[graph 0 input from stream 0:0 @ 0446e820] Setting 'sample_rate' to value '44100'
[graph 0 input from stream 0:0 @ 0446e820] Setting 'sample_fmt' to value 's16'
[graph 0 input from stream 0:0 @ 0446e820] Setting 'channel_layout' to value '0x3'
[graph 0 input from stream 0:0 @ 0446e820] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
[audio format for output stream 0:0 @ 0446ed80] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0446ed80] Setting 'sample_rates' to value '44100'
[audio format for output stream 0:0 @ 0446ed80] Setting 'channel_layouts' to value '0x3'
[AVFilterGraph @ 02b3fe80] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed
Output #0, wav, to 'AFN8429768236582_01_11_wav.wav':
  Metadata:
    ISFT            : Lavf56.9.101
    Stream #0:0, 0, 1/44100: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      encoder         : Lavc56.8.102 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (flac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 187 times
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 171 times
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 157 times
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 157 times
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 169 times
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 7 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 70650 (frame 436) to 74071 (frame 243)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 61719 (frame 435) to 74071 (frame 243)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 52626 (frame 434) to 74071 (frame 243)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 62483 (frame 436) to 83256 (frame 244)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 53552 (frame 435) to 83256 (frame 244)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 53909 (frame 436) to 91005 (frame 245)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times

Thanks! Ovidiu


          #6179: ffmpeg cannot recognize an wave file PCM detected as mp1   

Summary of the bug:when i use ffmpeg convert an wave file,the ffmpeg cannot recognize my file and probe result is mp1 How to reproduce:

% ./ffmpeg -i wav_format_recongnize_error.wav out.wav
ffmpeg version is 3.2.4
built with gcc 4.1.2 (GCC) 20070115 (prerelease) (SUSE Linux)

The full report is below:

ffmpeg started on 2017-02-22 at 22:00:00
Report written to "ffmpeg-20170222-220000.log"
Command line:
./ffmpeg -report -v 9 -loglevel 99 -i wav_format_recongnize_error.wav out.wav
ffmpeg version 3.2.4 Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 4.1.2 (GCC) 20070115 (prerelease) (SUSE Linux)
  configuration: --prefix=/data/AutoBuild/src/isd_musicadmin_proj/trunk/biz_pack/lesterkong/ffmpeg_3.2.4_video --disable-yasm --enable-static --disable-shared --enable-libx264 --enable-gpl --extra-cflags=-I/data/AutoBuild/src/isd_musicadmin_proj/trunk/biz_pack/lesterkong/x264_static/include --extra-ldflags=-L/data/AutoBuild/src/isd_musicadmin_proj/trunk/biz_pack/lesterkong/x264_static/lib --extra-libs='-lx264 -ldl' --enable-cross-compile --arch=i486 --target-os=linux
  libavutil      55. 34.101 / 55. 34.101
  libavcodec     57. 64.101 / 57. 64.101
  libavformat    57. 56.101 / 57. 56.101
  libavdevice    57.  1.100 / 57.  1.100
  libavfilter     6. 65.100 /  6. 65.100
  libswscale      4.  2.100 /  4.  2.100
  libswresample   2.  3.100 /  2.  3.100
  libpostproc    54.  1.100 / 54.  1.100
Splitting the commandline.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Reading option '-v' ... matched as option 'v' (set logging level) with argument '9'.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument '99'.
Reading option '-i' ... matched as input url with argument 'wav_format_recongnize_error.wav'.
Reading option 'out.wav' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option report (generate a report) with argument 1.
Applying option v (set logging level) with argument 9.
Applying option loglevel (set logging level) with argument 99.
Successfully parsed a group of options.
Parsing a group of options: input url wav_format_recongnize_error.wav.
Successfully parsed a group of options.
Opening an input file: wav_format_recongnize_error.wav.
[file @ 0x97cfb20] Setting default whitelist 'file,crypto'
[wav @ 0x97cf400] Format wav probed with size=2048 and score=99
[wav @ 0x97cf400] Before avformat_find_stream_info() pos: 44 bytes read:65580 seeks:1 nb_streams:1
[wav @ 0x97cf400] probing stream 0 pp:32
[wav @ 0x97cf400] probing stream 0 pp:31
[wav @ 0x97cf400] probing stream 0 pp:30
[wav @ 0x97cf400] probing stream 0 pp:29
[wav @ 0x97cf400] Probe with size=16384, packets=2472 detected mp3 with score=25
[wav @ 0x97cf400] probing stream 0 pp:28
[wav @ 0x97cf400] probing stream 0 pp:27
[wav @ 0x97cf400] probing stream 0 pp:26
[wav @ 0x97cf400] probing stream 0 pp:25
[wav @ 0x97cf400] Probe with size=32768, packets=2476 detected mp3 with score=25
[wav @ 0x97cf400] probing stream 0 pp:24
[wav @ 0x97cf400] probing stream 0 pp:23
[wav @ 0x97cf400] probing stream 0 pp:22
[wav @ 0x97cf400] probing stream 0 pp:21
[wav @ 0x97cf400] probing stream 0 pp:20
[wav @ 0x97cf400] probing stream 0 pp:19
[wav @ 0x97cf400] probing stream 0 pp:18
[wav @ 0x97cf400] probing stream 0 pp:17
[wav @ 0x97cf400] Probe with size=65536, packets=2484 detected mp3 with score=25
[wav @ 0x97cf400] probing stream 0 pp:16
[wav @ 0x97cf400] probing stream 0 pp:15
[wav @ 0x97cf400] probing stream 0 pp:14
[wav @ 0x97cf400] probing stream 0 pp:13
[wav @ 0x97cf400] probing stream 0 pp:12
[wav @ 0x97cf400] probing stream 0 pp:11
[wav @ 0x97cf400] probing stream 0 pp:10
[wav @ 0x97cf400] probing stream 0 pp:9
[wav @ 0x97cf400] probing stream 0 pp:8
[wav @ 0x97cf400] probing stream 0 pp:7
[wav @ 0x97cf400] probing stream 0 pp:6
[wav @ 0x97cf400] probing stream 0 pp:5
[wav @ 0x97cf400] probing stream 0 pp:4
[wav @ 0x97cf400] probing stream 0 pp:3
[wav @ 0x97cf400] probing stream 0 pp:2
[wav @ 0x97cf400] probing stream 0 pp:1
[wav @ 0x97cf400] Probe with size=131072, packets=2500 detected mp3 with score=50
[wav @ 0x97cf400] probed stream 0
[mp1 @ 0x97d0500] Header missing
[mp1 @ 0x97d0500] Header missing
[wav @ 0x97cf400] All info found
[wav @ 0x97cf400] Estimating duration from bitrate, this may be inaccurate
[wav @ 0x97cf400] After avformat_find_stream_info() pos: 131116 bytes read:196652 seeks:1 frames:50
Input #0, wav, from 'wav_format_recongnize_error.wav':
  Duration: 00:00:58.05, bitrate: 1411 kb/s
    Stream #0:0, 50, 1/44100: Audio: mp1 ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16p, 1411 kb/s
Successfully opened the file.
Parsing a group of options: output url out.wav.
Successfully parsed a group of options.
Opening an output file: out.wav.
[file @ 0x97d3760] Setting default whitelist 'file,crypto'
Successfully opened the file.
detected 6 logical cores
[graph 0 input from stream 0:0 @ 0x980dec0] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0x980dec0] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0x980dec0] Setting 'sample_fmt' to value 's16p'
[graph 0 input from stream 0:0 @ 0x980dec0] Setting 'channel_layout' to value '0x4'
[graph 0 input from stream 0:0 @ 0x980dec0] tb:1/48000 samplefmt:s16p samplerate:48000 chlayout:0x4
[audio format for output stream 0:0 @ 0x97e5700] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0x97e5700] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
[AVFilterGraph @ 0x98036e0] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0x97d0b60] [SWR @ 0x9817580] Using s16p internally between filters
[auto-inserted resampler 0 @ 0x97d0b60] ch:1 chl:mono fmt:s16p r:48000Hz -> ch:1 chl:mono fmt:s16 r:48000Hz
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.56.101
    Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      encoder         : Lavc57.64.101 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (mp1 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[mp1 @ 0x97d31c0] Header missing
Error while decoding stream #0:0: Invalid data found when processing input
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[mp1 @ 0x97d31c0] Header missing
Error while decoding stream #0:0: Invalid data found when processing input
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[mp1 @ 0x97d31c0] Header missing
.....
Error while decoding stream #0:0: Invalid data found when processing input
[mp1 @ 0x97d31c0] Header missing
Error while decoding stream #0:0: Invalid data found when processing input
[output stream 0:0 @ 0x97d2340] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
size=     620kB time=00:00:06.61 bitrate= 767.0kbits/s speed=7.94x
video:0kB audio:620kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.012296%
Input file #0 (wav_format_recongnize_error.wav):
  Input stream #0:0 (audio): 4837 packets read (10239956 bytes); 827 frames decoded (317568 samples);
  Total: 4837 packets (10239956 bytes) demuxed
Output file #0 (out.wav):
  Output stream #0:0 (audio): 826 frames encoded (317184 samples); 826 packets muxed (634368 bytes);
  Total: 826 packets (634368 bytes) muxed
827 frames successfully decoded, 4010 decoding errors
[AVIOContext @ 0x97e0a60] Statistics: 4 seeks, 829 writeouts
[AVIOContext @ 0x97d7cc0] Statistics: 10305536 bytes read, 1 seeks
Conversion failed!}}}

          #6349: libmp3lame encode segfaults with audio filter   

Summary of the bug:

FFmpeg segfaults with this file when encoding with libmp3lame with an audio filter. The same segfault happens when using one of several other audio filters. FFmpeg was just built from the current git head (dfc4ce5f5ddb7ae0934bb3ca40b99932cf3e1cb0).

ffprobe file info:

Input #0, wav, from 'clip.wav':
  Metadata:
    date            : 2017-01-0
    encoder         : Lavf57.72.101
    encoded_by      : Pro Tools
  Duration: 00:00:15.02, bitrate: 1152 kb/s
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32 (24 bit), 1152 kb/s

How to reproduce:

ffmpeg -i clip.wav -af alimiter -c:a mp3 -b:a 128k -ar 48k -f null -
ffmpeg version N-85642-gdfc4ce5 Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 7.0.2 (clang-700.1.81)
  configuration: --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda --enable-libvpx --enable-libopus
  libavutil      55. 61.100 / 55. 61.100
  libavcodec     57. 93.100 / 57. 93.100
  libavformat    57. 72.101 / 57. 72.101
  libavdevice    57.  7.100 / 57.  7.100
  libavfilter     6. 87.100 /  6. 87.100
  libavresample   3.  6.  0 /  3.  6.  0
  libswscale      4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc    54.  6.100 / 54.  6.100
Input #0, wav, from 'clip.wav':
  Metadata:
    date            : 2017-01-0
    encoder         : Lavf57.72.101
    encoded_by      : Pro Tools
  Duration: 00:00:15.02, bitrate: 1152 kb/s
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s24le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, null, to 'pipe:':
  Metadata:
    date            : 2017-01-0
    encoded_by      : Pro Tools
    encoder         : Lavf57.72.101
    Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, mono, fltp (24 bit), 128 kb/s
    Metadata:
      encoder         : Lavc57.93.100 libmp3lame
zsh: segmentation fault  ffmpeg -i clip.wav -af alimiter -c:a mp3 -b:a 128k -ar 48k -f null -

gdb backtrace:

Program received signal SIGSEGV, Segmentation fault.
0x0000000101ba55cd in av_assert0_fpu () from /usr/local/lib/libavutil.55.dylib
(gdb) backtrace
#0  0x0000000101ba55cd in av_assert0_fpu () from /usr/local/lib/libavutil.55.dylib
#1  0x00000001007bdb10 in av_jni_get_java_vm () from /usr/local/lib/libavcodec.57.dylib
#2  0x00007fff5fbfdc0c in ?? ()
#3  0x0000000102601c80 in ?? ()
#4  0x00000000000001f0 in ?? ()
#5  0x0000000000000006 in ?? ()
#6  0x000000000000000c in ?? ()
#7  0x0000000101ef1000 in ?? ()
#8  0x00007fff5fbfdbb0 in ?? ()
#9  0x00000001026027e0 in ?? ()
#10 0x00007fff5fbfdc0c in ?? ()
#11 0x0000000102601c80 in ?? ()
#12 0x0000000000000000 in ?? ()

MacOS crash report:

Process:               ffmpeg [28218]
Path:                  /usr/local/bin/ffmpeg
Identifier:            ffmpeg
Version:               0
Code Type:             X86-64 (Native)
Parent Process:        zsh [15966]
Responsible:           Terminal [90035]
User ID:               1921560808
Date/Time:             2017-04-24 14:15:16.635 -0500
OS Version:            Mac OS X 10.10.5 (14F2109)
Report Version:        11
Anonymous UUID:        9A022765-A1D6-4133-507D-AF6B8EDF895E
Sleep/Wake UUID:       EE41D361-72E6-423A-86A8-FC2740611B59
Time Awake Since Boot: 530000 seconds
Time Since Wake:       4300 seconds
Crashed Thread:        0  Dispatch queue: com.apple.main-thread
Exception Type:        EXC_BAD_ACCESS (SIGSEGV)
Exception Codes:       EXC_I386_GPFLT
Thread 0 Crashed:: Dispatch queue: com.apple.main-thread
0   libavutil.55.dylib            	0x00000001098f25cd av_assert0_fpu + 1485
Thread 1:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 2:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 3:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 4:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 5:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 6:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 7:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 8:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 9:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 0 crashed with X86 Thread State (64-bit):
  rax: 0x00000000000002d0  rbx: 0x00007fa2b042a780  rcx: 0x00007fa2b0423100  rdx: 0x0000000000000b30
  rdi: 0x00007fa2b080f600  rsi: 0x00007fa2b089a014  rbp: 0x0000000000000000  rsp: 0x00007fff57ebfb68
   r8: 0x00000000000f4240   r9: 0x0000000000000000  r10: 0x000000000007a120  r11: 0x00007fa2b0400000
  r12: 0x00007fa2b0820c00  r13: 0x00007fa2b0429c00  r14: 0x00007fa2b0423b00  r15: 0x00007fa2b0820c00
  rip: 0x00000001098f25cd  rfl: 0x0000000000010202  cr2: 0x0000000109c0b5b0
Logical CPU:     0
Error Code:      0x00000000
Trap Number:     13
Binary Images:
       0x107d3e000 -        0x107d6bfff +ffmpeg (0) <EEABF0E9-23E0-3924-85E0-DC4724D5123E> /usr/local/bin/ffmpeg
       0x107d7e000 -        0x107d86ff7 +libavdevice.57.dylib (0) <3950BA15-C048-3CDB-9EEA-8CAD234DCECE> /usr/local/lib/libavdevice.57.dylib
       0x107d8e000 -        0x107f27fff +libavfilter.6.dylib (0) <EAF7411E-8A52-3FFA-AF89-95BD2B0A9EF5> /usr/local/lib/libavfilter.6.dylib
       0x107f91000 -        0x10812dff7 +libavformat.57.dylib (0) <77435D7B-A942-3BD8-BB97-1D4AAD5254B8> /usr/local/lib/libavformat.57.dylib
       0x108173000 -        0x1090c2f4f +libavcodec.57.dylib (0) <ACA9F35A-557F-311F-B6FD-D24D99EBD1EC> /usr/local/lib/libavcodec.57.dylib
       0x1097ca000 -        0x1097e5ff7 +libavresample.3.dylib (0) <0A355673-F3F3-3989-9CD1-53909B589911> /usr/local/lib/libavresample.3.dylib
       0x1097ec000 -        0x109809ff7 +libpostproc.54.dylib (0) <D4A1CAE3-E011-3C9B-A32B-0FD1E9958590> /usr/local/lib/libpostproc.54.dylib
       0x10980e000 -        0x109826fff +libswresample.2.dylib (0) <80518108-514D-337C-ADBA-27197EB2003D> /usr/local/lib/libswresample.2.dylib
       0x10982c000 -        0x1098afff7 +libswscale.4.dylib (0) <09A2EAE9-8F62-3883-99AF-08B67217C66F> /usr/local/lib/libswscale.4.dylib
       0x1098bd000 -        0x109906ff7 +libavutil.55.dylib (0) <B7957198-75E9-36BF-96CB-21ED3DA10F2E> /usr/local/lib/libavutil.55.dylib
       0x109929000 -        0x1099d2ff7 +libSDL2-2.0.0.dylib (0) <13519299-19D0-3E7B-B282-30793CD07E04> /usr/local/opt/sdl2/lib/libSDL2-2.0.0.dylib
       0x109a0e000 -        0x109a0ffff  com.apple.VideoDecodeAcceleration (1.1 - 8) <B624EA7A-EECA-3A30-9FE9-25C8339535BE> /System/Library/Frameworks/VideoDecodeAcceleration.framework/Versions/A/VideoDecodeAcceleration
       0x109a1a000 -        0x109af9fff +libx264.148.dylib (0) <3F494DDD-A841-3C1A-9B36-2A18A89A7102> /usr/local/opt/x264/lib/libx264.148.dylib
       0x109b8d000 -        0x109bceff7 +libopus.0.dylib (0) <0BFFA36E-D9E6-3ECB-A106-1380FF0E04FE> /usr/local/opt/opus/lib/libopus.0.dylib
       0x109bd6000 -        0x109c0bfff +libmp3lame.0.dylib (0) <2D449F39-3AE5-3C4A-B56D-AF82F6A481F3> /usr/local/opt/lame/lib/libmp3lame.0.dylib
       0x109c48000 -        0x109c4afff  com.apple.ForceFeedback (1.0.6 - 1.0.6) <A411AA46-5DB6-3C6C-B4B4-613EC1EF33C3> /System/Library/Frameworks/ForceFeedback.framework/Versions/A/ForceFeedback
    0x7fff67cb2000 -     0x7fff67ce892f  dyld (353.2.4) <708E385E-50C6-32A3-983C-42E97BEE4346> /usr/lib/dyld
    0x7fff8a0f9000 -     0x7fff8a115ff7  libsystem_malloc.dylib (53.30.1) <DDA8928B-CC0D-3255-BD8A-3FEA0982B890> /usr/lib/system/libsystem_malloc.dylib
    0x7fff8a285000 -     0x7fff8a5a0fcf  com.apple.vImage (8.0 - 8.0) <1183FE6A-FDB6-3B3B-928D-50C7909F2308> /System/Library/Frameworks/Accelerate.framework/Versions/A/Frameworks/vImage.framework/Versions/A/vImage
    0x7fff8a5a1000 -     0x7fff8a5f2ff7  com.apple.AppleVAFramework (5.0.31 - 5.0.31) <17C35538-7C2B-3735-A1B3-E8487EED0668> /System/Library/PrivateFrameworks/AppleVA.framework/Versions/A/AppleVA
    0x7fff8a5f3000 -     0x7fff8a5fcfff  libsystem_pthread.dylib (105.40.1) <ACE90967-ECD0-3251-AEEB-461E3C6414F7> /usr/lib/system/libsystem_pthread.dylib
    0x7fff8a5fd000 -     0x7fff8b17eff7  com.apple.AppKit (6.9 - 1348.17) <E485D56D-3E72-34B7-99BB-BFDEE2D07BF5> /System/Library/Frameworks/AppKit.framework/Versions/C/AppKit
    0x7fff8bace000 -     0x7fff8bb3fffb  com.apple.ApplicationServices.ATS (360 - 375.4) <888FE0B1-835C-30B1-80AD-5066CADB0364> /System/Library/Frameworks/ApplicationServices.framework/Versions/A/Frameworks/ATS.framework/Versions/A/ATS
    0x7fff8bb62000 -     0x7fff8bb6bff7  libsystem_notify.dylib (133.1.1) <61147800-F320-3DAA-850C-BADF33855F29> /usr/lib/system/libsystem_notify.dylib
    0x7fff8bb6c000 -     0x7fff8bb88fff  com.apple.GenerationalStorage (2.0 - 209.11) <9FF8DD11-25FB-3047-A5BF-9415339B3EEC> /System/Library/PrivateFrameworks/GenerationalStorage.framework/Versions/A/GenerationalStorage
    0x7fff8be5c000 -     0x7fff8be76ff3  com.apple.Ubiquity (1.3 - 313) <DF56A657-CC6E-3BE2-86A0-71F07127724C> /System/Library/PrivateFrameworks/Ubiquity.framework/Versions/A/Ubiquity
    0x7fff8be90000 -     0x7fff8bf2fe27  com.apple.AppleJPEG (1.0 - 1) <6627DDD9-A8FE-3968-B23A-B6A29AA3919A> /System/Library/PrivateFrameworks/AppleJPEG.framework/Versions/A/AppleJPEG
    0x7fff8c1cd000 -     0x7fff8c2f5ff7  com.apple.coreui (2.1 - 308.6) <9E0E9C6A-68F5-34C1-A17C-96226D401D4D> /System/Library/PrivateFrameworks/CoreUI.framework/Versions/A/CoreUI
    0x7fff8c2f6000 -     0x7fff8c2f6fff  com.apple.CoreServices (62 - 62) <C69DA8A7-B536-34BF-A93F-1C170E2C6D58> /System/Library/Frameworks/CoreServices.framework/Versions/A/CoreServices
    0x7fff8c2f7000 -     0x7fff8c30eff7  libLinearAlgebra.dylib (1128) <E78CCBAA-A999-3B65-8EC9-06DB15E67C37> /System/Library/Frameworks/Accelerate.framework/Versions/A/Frameworks/vecLib.framework/Versions/A/libLinearAlgebra.dylib
    0x7fff8c30f000 -     0x7fff8c315ff7  libsystem_networkextension.dylib (167.40.3) <BA58B30B-8377-3B0A-8AE3-4F84021D9D4E> /usr/lib/system/libsystem_networkextension.dylib
    0x7fff8c358000 -     0x7fff8c3d9ff7  com.apple.CoreUtils (1.1 - 110.1) <3808CBC5-1DF1-3821-8AAE-57C30D4F514B> /System/Library/PrivateFrameworks/CoreUtils.framework/Versions/A/CoreUtils
    0x7fff8c415000 -     0x7fff8c506fff  libJP2.dylib (1247) <D74A8708-C27E-3D7C-89D0-BE999596E2B1> /System/Library/Frameworks/ImageIO.framework/Versions/A/Resources/libJP2.dylib
    0x7fff8c87f000 -     0x7fff8c8f3ffb  com.apple.securityfoundation (6.0 - 55126) <344401F5-4016-3AEB-BFDC-CE8E5546D39F> /System/Library/Frameworks/SecurityFoundation.framework/Versions/A/SecurityFoundation
    0x7fff8cb53000 -     0x7fff8cbbffff  com.apple.framework.CoreWLAN (5.0 - 500.35.2) <03697149-1CDD-32FF-B564-1C1EF5E9E5C3> /System/Library/Frameworks/CoreWLAN.framework/Versions/A/CoreWLAN
    0x7fff8cbc3000 -     0x7fff8cd73ff3  com.apple.QuartzCore (1.10 - 361.19) <E59000AA-DC30-3F73-A103-0240576C9556> /System/Library/Frameworks/QuartzCore.framework/Versions/A/QuartzCore
    0x7fff8cd74000 -     0x7fff8cda1fff  com.apple.CoreVideo (1.8 - 145.1) <18DB07E0-B927-3260-A234-636F298D1917> /System/Library/Frameworks/CoreVideo.framework/Versions/A/CoreVideo
    0x7fff8cdff000 -     0x7fff8cf2ffff  com.apple.UIFoundation (1.0 - 1) <466BDFA8-0B9F-3AB0-989D-F9779422926A> /System/Library/PrivateFrameworks/UIFoundation.framework/Versions/A/UIFoundation
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    0x7fff97251000 -     0x7fff97251fff  libOpenScriptingUtil.dylib (162.2) <D6A2216D-ADB2-3F24-AD30-F6D00829F545> /usr/lib/libOpenScriptingUtil.dylib
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    0x7fff97285000 -     0x7fff9728aff7  libmacho.dylib (862) <126CA2ED-DE91-308F-8881-B9DAEC3C63B6> /usr/lib/system/libmacho.dylib
    0x7fff9728b000 -     0x7fff9728bfff  com.apple.ApplicationServices (48 - 48) <5BF7910B-C328-3BF8-BA4F-CE52B574CE01> /System/Library/Frameworks/ApplicationServices.framework/Versions/A/ApplicationServices
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    0x7fff98c06000 -     0x7fff98c10ff7  com.apple.NetAuth (5.2 - 5.2) <2BBD749A-8E18-35B8-8E48-A90347C1CCA7> /System/Library/PrivateFrameworks/NetAuth.framework/Versions/A/NetAuth
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External Modification Summary:
  Calls made by other processes targeting this process:
    task_for_pid: 0
    thread_create: 0
    thread_set_state: 0
  Calls made by this process:
    task_for_pid: 0
    thread_create: 0
    thread_set_state: 0
  Calls made by all processes on this machine:
    task_for_pid: 3299541
    thread_create: 0
    thread_set_state: 57
VM Region Summary:
ReadOnly portion of Libraries: Total=213.5M resident=146.4M(69%) swapped_out_or_unallocated=67.1M(31%)
Writable regions: Total=49.3M written=2932K(6%) resident=3192K(6%) swapped_out=0K(0%) unallocated=46.2M(94%)
REGION TYPE                      VIRTUAL
===========                      =======
Kernel Alloc Once                     4K
MALLOC                             28.5M
MALLOC (admin)                       32K
MALLOC_LARGE (reserved)             132K        reserved VM address space (unallocated)
STACK GUARD                        56.0M
Stack                              12.6M
VM_ALLOCATE                           8K
__DATA                             23.7M
__IMAGE                             528K
__LINKEDIT                         71.5M
__TEXT                            142.1M
__UNICODE                           552K
shared memory                         4K
===========                      =======
TOTAL                             335.5M
TOTAL, minus reserved VM space    335.4M

          #1659: g723_1: wrong current duration after seeking   

ffplay displays wrong current duration after seeking in g723_1 (in wav)

C:\>ffmpeg -i g723_1.wav
ffmpeg version N-43678-gceb8773 Copyright (c) 2000-2012 the FFmpeg developers
  built on Aug 18 2012 02:22:03 with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.
0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-w32threads --enable-memalign-hack --enable-runtime-cpudetect
--enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -
lwinmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snap
shots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/s
napshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3
--enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-
libvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-
libopencore-amrwb --enable-libmp3lame --enable-libvpx --disable-decoder=libvpx
  libavutil      51. 70.100 / 51. 70.100
  libavcodec     54. 53.100 / 54. 53.100
  libavformat    54. 25.100 / 54. 25.100
  libavdevice    54.  2.100 / 54.  2.100
  libavfilter     3. 10.100 /  3. 10.100
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
[wav @ 0x1eace20] Estimating duration from bitrate, this may be inaccurate
Input #0, wav, from 'g723_1.wav':
  Duration: 00:00:30.07, bitrate: 6 kb/s
    Stream #0:0: Audio: g723_1 (B[0][0][0] / 0x0042), 8000 Hz, mono, s16, 6 kb/s
At least one output file must be specified

          #1712: Audio precision in lossless and long time to encode.   

Summary of the bug: Hi, first of all, i'm sorry for my english. I've tried the last ffmpeg build and i get a problem with Audio precision in lossless audio file, the audio is delayed about 2 seconds. The audio is passed to another program throught a pipe, but i know that the other program isn't the problem, because with a very old ffmpeg version (SVN 16573, Jan 13 2009) works fine, and i've tried with wav output without pipe and get the same problem.

If you want, here's a test wav files: http://www.mediafire.com/?1ewceliyjhspedk

The command line that i've used is : ffmpeg_old -i "input.ape" -ss 00:59:18.71 -t 00:00:15.55 out_oldffmpeg.wav ffmpeg -i "input.ape" -ss 00:59:18.71 -t 00:00:15.55 out_newffmpeg.wav

How to reproduce:

% ffmpeg -i input -ss 00:45:25 -t 00:05:25 -ac 2 -f wav -
The output is a pipe to another program.
'''ffmpeg version''' N-44080-gdfb4757
'''built on''' Aug 31 2012 16:58:51 with gcc 4.7.1 (GCC)
configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runt
ime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass -
-enable-libcelt --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-l
ibfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenj
peg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheo
ra --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li
bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --ena
ble-zlib

The other thing that i wan to answer is why the program take a long time to convert a portion of a file, for example: ffmpeg -i input.ape -ss 00:45:15.42 -t 00:00:15.25 output.wav

the program stay a long time without do anything and then start to convert. I take about 2 minutes to convert 15s of audio.


          #1841: raac: fail to seek   

ffmpeg fails to seek above 26s

http://www.datafilehost.com/download-733c41c3.html

C:\>ffmpeg -ss 26 -i raac.rm out.wav
ffmpeg version N-45867-g3318d6b Copyright (c) 2000-2012 the FFmpeg developers
  built on Oct 23 2012 02:39:33 with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.
0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-w32threads --enable-memalign-hack --enable-runtime-cpudetect
--enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -
lwinmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snap
shots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/s
napshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3
--enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-
libvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-
libopencore-amrwb --enable-libmp3lame --enable-libvpx --disable-decoder=libvpx
  libavutil      51. 77.100 / 51. 77.100
  libavcodec     54. 68.100 / 54. 68.100
  libavformat    54. 33.100 / 54. 33.100
  libavdevice    54.  3.100 / 54.  3.100
  libavfilter     3. 20.104 /  3. 20.104
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 16.100 /  0. 16.100
  libpostproc    52.  1.100 / 52.  1.100
[rm @ 0x1edbe20] Invalid stream index 1 for index at pos 1233529
Input #0, rm, from 'raac.rm':
  Metadata:
    title           :
    author          :
    copyright       :
    comment         :
    Creation Date   : 4/21/2012 13:08:18
    Generated By    : Helix Producer SDK 13.1 for Windows, Build 13.1.1.3436
    Modification Date: 4/21/2012 13:08:18
  Duration: 00:00:30.07, start: 0.000000, bitrate: 328 kb/s
    Stream #0:0: Audio: aac (raac / 0x63616172), 44100 Hz, stereo, s16, 320 kb/s
Output #0, wav, to 'out.wav':
  Metadata:
    INAM            :
    author          :
    ICOP            :
    ICMT            :
    Creation Date   : 4/21/2012 13:08:18
    Generated By    : Helix Producer SDK 13.1 for Windows, Build 13.1.1.3436
    Modification Date: 4/21/2012 13:08:18
    ISFT            : Lavf54.33.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16
, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac -> pcm_s16le)
Press [q] to stop, [?] for help
[aac @ 0x1edc920] Reserved bit set.
Error while decoding stream #0:0: Invalid data found when processing input
Truncating packet of size 11122 to 7128
[aac @ 0x1edc920] Prediction is not allowed in AAC-LC.
Error while decoding stream #0:0: Invalid data found when processing input
raac.rm: Input/output error
size=       0kB time=00:00:00.00 bitrate=   0.0kbits/s
video:0kB audio:0kB subtitle:0 global headers:0kB muxing overhead inf%
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters i
f used)

          #1958: mp3 24khz (in wav): overreads   

This happens only with 24khz mode.

I can hear some distortions.

C:\>ffmpeg -i 24khz_2ch_160kbps.wav -t 2 -f null -
ffmpeg version N-46755-ge70144c Copyright (c) 2000-2012 the FFmpeg developers
  built on Nov 15 2012 02:38:20 with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.
0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-w32threads --enable-memalign-hack --enable-runtime-cpudetect
--enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -
lwinmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snap
shots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/s
napshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3
--enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-
libvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-
libopencore-amrwb --enable-libmp3lame --enable-libvpx --disable-decoder=libvpx
  libavutil      52.  6.100 / 52.  6.100
  libavcodec     54. 71.100 / 54. 71.100
  libavformat    54. 36.100 / 54. 36.100
  libavdevice    54.  3.100 / 54.  3.100
  libavfilter     3. 23.100 /  3. 23.100
  libswscale      2.  1.102 /  2.  1.102
  libswresample   0. 16.100 /  0. 16.100
  libpostproc    52.  1.100 / 52.  1.100
[wav @ 0x1edc3c0] max_analyze_duration 5000000 reached at 5016000
[wav @ 0x1edc3c0] Estimating duration from bitrate, this may be inaccurate
Input #0, wav, from '24khz_2ch_160kbps.wav':
  Duration: 00:00:33.12, bitrate: 127 kb/s
    Stream #0:0: Audio: mp3 (U[0][0][0] / 0x0055), 24000 Hz, stereo, s16, 128 kb
/s
Output #0, null, to 'pipe:':
  Metadata:
    encoder         : Lavf54.36.100
    Stream #0:0: Audio: pcm_s16le, 24000 Hz, stereo, s16, 768 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 -> pcm_s16le)
Press [q] to stop, [?] for help
[mp3 @ 0x1edcb60] overread, skip -7 enddists: -6 -6
[mp3 @ 0x1edcb60] overread, skip -6 enddists: -5 -5
[mp3 @ 0x1edcb60] overread, skip -9 enddists: -1 -1
[mp3 @ 0x1edcb60] overread, skip -7 enddists: -3 -3
    Last message repeated 1 times
[mp3 @ 0x1edcb60] overread, skip -6 enddists: -2 -2
[mp3 @ 0x1edcb60] overread, skip -6 enddists: -4 -4
[mp3 @ 0x1edcb60] overread, skip -7 enddists: -4 -4
size=       0kB time=00:00:02.01 bitrate=   0.0kbits/s
video:0kB audio:189kB subtitle:0 global headers:0kB muxing overhead -100.000000%

          #2458: raw aac fails with FFmpeg, works fine with faad   

(videolan ticket 8309) I will upload an AAC sample (provided by a vlc user) that decodes fine with faad, FFmpeg fails both auto-detection and decoding.

$ ffmpeg -probesize 2G -analyzeduration 2G -i trac_8309_raw.aac
ffmpeg version N-51927-geabc0c7 Copyright (c) 2000-2013 the FFmpeg developers
  built on Apr 11 2013 18:23:56 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 26.100 / 52. 26.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  2.100 / 55.  2.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 52.101 /  3. 52.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
trac_8309_raw.aac: Invalid data found when processing input
$ faad trac_8309_raw.aac
 *********** Ahead Software MPEG-4 AAC Decoder V2.7 ******************
 Build: Apr  3 2012
 Copyright 2002-2004: Ahead Software AG
 http://www.audiocoding.com
 Floating point version
 This program is free software; you can redistribute it and/or modify
 it under the terms of the GNU General Public License.
 **************************************************************************
trac_8309_raw.aac file info:
RAW
  ---------------------
 | Config:  2 Ch       |
  ---------------------
 | Ch |    Position    |
  ---------------------
 | 00 | Left front     |
 | 01 | Right front    |
  ---------------------
Decoding trac_8309_raw.aac took:  0.76 sec.  0.00x real-time.
$ ffmpeg -i trac_8309_raw.wav
ffmpeg version N-51927-geabc0c7 Copyright (c) 2000-2013 the FFmpeg developers
  built on Apr 11 2013 18:23:56 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 26.100 / 52. 26.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  2.100 / 55.  2.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 52.101 /  3. 52.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[wav @ 0x1d4ca20] max_analyze_duration 5000000 reached at 5015510 microseconds
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, wav, from 'trac_8309_raw.wav':
  Duration: 00:05:41.22, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
At least one output file must be specified
$ ffmpeg -f aac -i trac_8309_raw.aac
ffmpeg version N-51927-geabc0c7 Copyright (c) 2000-2013 the FFmpeg developers
  built on Apr 11 2013 18:23:56 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 26.100 / 52. 26.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  2.100 / 55.  2.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 52.101 /  3. 52.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[aac @ 0x1ab25a0] get_buffer() failed
[aac @ 0x1ab25a0] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x1ab25a0] Reserved bit set.
[aac @ 0x1ab25a0] channel element 3.1 is not allocated
[aac @ 0x1ab25a0] Number of bands (7) exceeds limit (4).
[aac @ 0x1ab25a0] ms_present = 3 is reserved.
[aac @ 0x1ab25a0] Number of bands (15) exceeds limit (12).
[aac @ 0x1ab25a0] channel element 1.13 is not allocated
[aac @ 0x1ab25a0] Reserved bit set.
[aac @ 0x1ab25a0] channel element 2.5 is not allocated
[aac @ 0x1ab25a0] channel element 3.2 is not allocated
[aac @ 0x1ab25a0] channel element 0.1 is not allocated
[aac @ 0x1ab25a0] Number of bands (6) exceeds limit (2).
[aac @ 0x1ab25a0] channel element 3.8 is not allocated
[aac @ 0x1ab25a0] Reserved bit set.
[aac @ 0x1ab25a0] channel element 1.11 is not allocated
[aac @ 0x1ab25a0] channel element 1.0 is not allocated
[aac @ 0x1ab25a0] Number of bands (10) exceeds limit (9).
[aac @ 0x1ab25a0] Reserved bit set.
[aac @ 0x1ab25a0] Number of bands (17) exceeds limit (12).
[aac @ 0x1ab25a0] Number of bands (20) exceeds limit (4).
[aac @ 0x1ab25a0] channel element 2.7 is not allocated
[aac @ 0x1ab25a0] channel element 1.10 is not allocated
[aac @ 0x1ab25a0] channel element 0.13 is not allocated
[aac @ 0x1ab25a0] channel element 2.1 is not allocated
[aac @ 0x1ab25a0] Reserved bit set.
...
[aac @ 0x1ab25a0] channel element 2.14 is not allocated
[aac @ 0x1ab25a0] Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0x1ab25a0] Inconsistent channel configuration.
[aac @ 0x1ab25a0] get_buffer() failed
[aac @ 0x1ab25a0] Inconsistent channel configuration.
[aac @ 0x1ab25a0] get_buffer() failed
[aac @ 0x1ab25a0] Number of bands (40) exceeds limit (32).
[aac @ 0x1ab25a0] Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0x1ab25a0] Inconsistent channel configuration.
[aac @ 0x1ab25a0] get_buffer() failed
[aac @ 0x1ab25a0] Number of bands (30) exceeds limit (13).
[aac @ 0x1ab25a0] channel element 2.7 is not allocated
[aac @ 0x1ab25a0] Reserved bit set.
[aac @ 0x1ab25a0] Dependent coupling is not supported together with LTP
    Last message repeated 31 times
[aac @ 0x1ab25a0] invalid band type
[aac @ 0x1ab25a0] Number of bands (6) exceeds limit (4).
[aac @ 0x1ab25a0] Dependent coupling is not supported together with LTP
    Last message repeated 31 times
[aac @ 0x1aa9a60] max_analyze_duration 5000000 reached at 5120000 microseconds
[aac @ 0x1aa9a60] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'trac_8309_raw.aac':
  Duration: 00:07:48.76, bitrate: 91 kb/s
    Stream #0:0: Audio: aac, 8000 Hz, 7.1(wide), fltp, 91 kb/s
At least one output file must be specified

MPlayer stops after 15 seconds, fails with -ac faad:

$ mplayer trac_8309_raw.aac -demuxer rawaudio -rawaudio format=0xFF -ac ffaac
MPlayer SVN-r36089-4.7 (C) 2000-2013 MPlayer Team
Playing trac_8309_raw.aac.
rawaudio file format detected.
Load subtitles in ./
==========================================================================
Forced audio codec: ffaac
Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
libavcodec version 55.2.100 (internal)
AUDIO: 44100 Hz, 2 ch, floatle, 1411.2 kbit/50.00% (ratio: 176400->352800)
Selected audio codec: [ffaac] afm: ffmpeg (FFmpeg AAC (MPEG-2/MPEG-4 Audio))
==========================================================================
AO: [alsa] 44100Hz 2ch floatle (4 bytes per sample)
Video: no video
Starting playback...
A:  15.7 (15.6) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]decode_band_types: Input buffer exhausted before END element found
A:  15.7 (15.7) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]channel element 2.7 is not allocated
A:   0.5 (00.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0xf75c1540]Inconsistent channel configuration.
[aac @ 0xf75c1540]get_buffer() failed
A:   1.5 (01.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]channel element 3.7 is not allocated
A:   2.5 (02.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0xf75c1540]Inconsistent channel configuration.
[aac @ 0xf75c1540]get_buffer() failed
A:   3.5 (03.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Number of bands (4) exceeds limit (3).
A:   4.5 (04.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Number of bands (16) exceeds limit (4).
A:   5.5 (05.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Reserved bit set.
A:   6.5 (06.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]channel element 2.7 is not allocated
A:   7.5 (07.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0xf75c1540]Inconsistent channel configuration.
[aac @ 0xf75c1540]get_buffer() failed
A:   8.5 (08.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Reserved bit set.
A:   9.5 (09.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]invalid band type
A:  10.5 (10.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]channel element 2.3 is not allocated
A:  11.5 (11.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Reserved bit set.
A:  12.5 (12.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0xf75c1540]Inconsistent channel configuration.
[aac @ 0xf75c1540]get_buffer() failed
A:  13.5 (13.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Reserved bit set.
A:  14.5 (14.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]channel element 3.4 is not allocated
A:  15.5 (15.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Reserved bit set.
A:  16.5 (16.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]invalid band type
A:  17.5 (17.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]Reserved bit set.
A:  18.5 (18.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]Reserved bit set.
A:  19.5 (19.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 2.6 is not allocated
A:  20.5 (20.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]SBR was found before the first channel element.
[aac @ 0xf75c1540]Reserved bit set.
A:  21.5 (21.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 3.2 is not allocated
A:  22.5 (22.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 2.2 is not allocated
A:  23.5 (23.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 3.0 is not allocated
A:  24.5 (24.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]Number of bands (11) exceeds limit (2).
A:  25.5 (25.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 3.7 is not allocated
A:  26.5 (26.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 3.7 is not allocated
A:  27.5 (27.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 2.15 is not allocated
A:  28.5 (28.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 2.9 is not allocated
A:  29.5 (29.5) of 30.2 (30.2)  0.3%
Exiting... (End of file)

          #3789: shortest command doesnt work when using a filter_complex with audio and video filters   

Summary of the bug: shortest command doesn't work when using a filter_complex with audio and video filters

I uploaded output.mp4 at 8:21 ET on 7/20/14 to upload.ffmpeg.org

The background audio is 4 mins long, but it should be truncated to the video length.

Shortest works if I take the drawtext off the filter_complex

How to reproduce:

ffmpeg -framerate 1/5 -i foo%03d.png -i audio.wav -i speak.wav -filter_complex '[1:a]volume=volume=0.2[aquiet];[aquiet][2:a]amix=inputs=2[a];[0:v]drawtext=fontsize=40:fontcolor=white@0.9:fontfile=FreeSerif.ttf:shadowy=2:shadowx=2:text="I am getting the hang of ffmpeg\!":x=(w-text_w)/2:y=(h-text_h-line_h)/2[v]' -map '[a]' -map '[v]' -shortest -r 32 -c:v libx264 -b:a 192k -pix_fmt yuv420p output.mp4
ffmpeg version 2.2.4-   http://johnvansickle.com/ffmpeg/    Copyright (c) 2000-2014 the FFmpeg developers
  built on Jul  1 2014 22:10:48 with gcc 4.8 (Debian 4.8.3-4)
  configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable-libx264 --enable-libwebp --enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg --enable-libopus --disable-ffserver --enable-libass --enable-gnutls --cc=gcc-4.8
  libavutil      52. 66.100 / 52. 66.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 33.100 / 55. 33.100
  libavdevice    55. 10.100 / 55. 10.100
  libavfilter     4.  2.100 /  4.  2.100
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, image2, from 'foo%03d.png':
  Duration: 00:00:30.00, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: png, rgb24, 640x360 [SAR 2835:2835 DAR 16:9], 0.20 fps, 1 tbr, 0.20 tbn, 0.20 tbc
Input #1, mp3, from 'audio.wav':
  Metadata:
    major_brand     : dash
    minor_version   : 0
    compatible_brands: iso6mp41
    encoder         : Lavf54.29.104
  Duration: 00:04:29.09, start: 0.025057, bitrate: 238 kb/s
    Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 238 kb/s
Guessed Channel Layout for  Input Stream #2.0 : mono
Input #2, wav, from 'speak.wav':
  Duration: 00:00:03.33, bitrate: 256 kb/s
    Stream #2:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s
File 'output.mp4' already exists. Overwrite ? [y/N] y
[libx264 @ 0x2fdf7a0] using SAR=1/1
[libx264 @ 0x2fdf7a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x2fdf7a0] profile High, level 3.0
[libx264 @ 0x2fdf7a0] 264 - core 142 r2 a5831aa - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=18 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'output.mp4':
  Metadata:
    encoder         : Lavf55.33.100
    Stream #0:0: Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, s16, 192 kb/s (default)
    Stream #0:1: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 640x360 [SAR 1:1 DAR 16:9], q=-1--1, 16384 tbn, 32 tbc (default)
Stream mapping:
  Stream #0:0 (png) -> drawtext
  Stream #1:0 (mp3) -> volume
  Stream #2:0 (pcm_s16le) -> amix:input1
  amix -> Stream #0:0 (libvo_aacenc)
  drawtext -> Stream #0:1 (libx264)
Press [q] to stop, [?] for help
frame=  800 fps=116 q=-1.0 Lsize=    6606kB time=00:04:29.05 bitrate= 201.1kbits/s dup=794 drop=0
video:238kB audio:6307kB subtitle:0 data:0 global headers:0kB muxing overhead 0.933773%
[libvo_aacenc @ 0x2fde340] 1 frames left in the queue on closing
[libx264 @ 0x2fdf7a0] frame I:5     Avg QP:17.56  size: 37668
[libx264 @ 0x2fdf7a0] frame P:201   Avg QP:17.22  size:   219
[libx264 @ 0x2fdf7a0] frame B:594   Avg QP:25.96  size:    18
[libx264 @ 0x2fdf7a0] consecutive B-frames:  0.8%  0.2%  1.5% 97.5%
[libx264 @ 0x2fdf7a0] mb I  I16..4:  4.7% 64.5% 30.8%
[libx264 @ 0x2fdf7a0] mb P  I16..4:  0.1%  0.2%  0.3%  P16..4:  0.5%  0.0%  0.0%  0.0%  0.0%    skip:99.0%
[libx264 @ 0x2fdf7a0] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8:  0.1%  0.0%  0.0%  direct: 0.0%  skip:99.9%  L0:16.4% L1:83.6% BI: 0.0%
[libx264 @ 0x2fdf7a0] 8x8 transform intra:58.6% inter:30.7%
[libx264 @ 0x2fdf7a0] coded y,uvDC,uvAC intra: 83.7% 87.7% 75.6% inter: 0.0% 0.1% 0.0%
[libx264 @ 0x2fdf7a0] i16 v,h,dc,p: 50% 11%  7% 32%
[libx264 @ 0x2fdf7a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 15% 19% 21%  6%  6%  5%  8%  6% 13%
[libx264 @ 0x2fdf7a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 24% 22% 11%  6%  7%  6%  7%  5% 12%
[libx264 @ 0x2fdf7a0] i8c dc,h,v,p: 50% 22% 20%  8%
[libx264 @ 0x2fdf7a0] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x2fdf7a0] ref P L0: 94.2%  0.5%  3.9%  1.4%
[libx264 @ 0x2fdf7a0] ref B L0: 53.1% 45.3%  1.6%
[libx264 @ 0x2fdf7a0] ref B L1: 95.1%  4.9%
[libx264 @ 0x2fdf7a0] kb/s:77.80

          #4590: concat filter does not work to loop audio   

As reported by irc user chama The following works as expected:

$ ffmpeg -i fate-suite/svq3/Vertical400kbit.sorenson3.mov -i fate-suite/svq3/Vertical400kbit.sorenson3.mov -i fate-suite/svq3/Vertical400kbit.sorenson3.mov -filter_complex [0:a][1:a][2:a]concat=n=3:v=0:a=1 out.wav

The output file has a correct length of 2:10

The following claims to encode 2:10 but the output file is much shorter:

$ ffmpeg -i fate-suite/svq3/Vertical400kbit.sorenson3.mov -filter_complex [0:a][0:a][0:a]concat=n=3:v=0:a=1 out.wav
ffmpeg version N-72624-g196b885 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 26.101 / 54. 26.101
  libavcodec     56. 41.101 / 56. 41.101
  libavformat    56. 34.100 / 56. 34.100
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 16.101 /  5. 16.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Guessed Channel Layout for  Input Stream #0.1 : mono
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'fate-suite/svq3/Vertical400kbit.sorenson3.mov':
  Metadata:
    creation_time   : 2001-03-20 16:17:18
    title           : Vertical Online SV3 Demo
    title-eng       : Vertical Online SV3 Demo
    artist          : Logan Kelsey
    artist-eng      : Logan Kelsey
    copyright       : (c) Vertical Online 2001
    copyright-eng   : (c) Vertical Online 2001
    encoder         : Sorenson Video 3
    encoder-eng     : Sorenson Video 3
  Duration: 00:00:43.58, start: 0.000000, bitrate: 580 kb/s
    Stream #0:0(eng): Video: svq3 (SVQ3 / 0x33515653), yuvj420p(pc), 320x240, 391 kb/s, 30.02 fps, 30 tbr, 600 tbn, 600 tbc (default)
    Metadata:
      creation_time   : 2001-03-20 16:17:18
      handler_name    : Apple Alias Data Handler
      encoder         : Sorenson Video 3
    Stream #0:1(eng): Audio: adpcm_ima_qt (ima4 / 0x34616D69), 44100 Hz, 1 channels, s16p, 176 kb/s (default)
    Metadata:
      creation_time   : 2001-03-20 16:17:18
      handler_name    : Apple Alias Data Handler
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf56.34.100
    INAM            : Vertical Online SV3 Demo
    title-eng       : Vertical Online SV3 Demo
    IART            : Logan Kelsey
    artist-eng      : Logan Kelsey
    ICOP            : (c) Vertical Online 2001
    copyright-eng   : (c) Vertical Online 2001
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s (default)
    Metadata:
      encoder         : Lavc56.41.101 pcm_s16le
Stream mapping:
  Stream #0:1 (adpcm_ima_qt) -> concat:in0:a0
  Stream #0:1 (adpcm_ima_qt) -> concat:in1:a0
  Stream #0:1 (adpcm_ima_qt) -> concat:in2:a0
  concat -> Stream #0:0 (pcm_s16le)
Press [q] to stop, [?] for help
Multiple frames in a packet from stream 1
[Parsed_concat_0 @ 0x26bed40] Buffer queue overflow, dropping.
    Last message repeated 59539 times
[output stream 0:0 @ 0x26bf600] 100 buffers queued in output stream 0:0, something may be wrong.
size=    3817kB time=00:02:10.72 bitrate= 239.2kbits/s
video:0kB audio:3817kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.004247%
$ ffmpeg -i out.wav
ffmpeg version N-72624-g196b885 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 26.101 / 54. 26.101
  libavcodec     56. 41.101 / 56. 41.101
  libavformat    56. 34.100 / 56. 34.100
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 16.101 /  5. 16.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, wav, from 'out.wav':
  Metadata:
    artist          : Logan Kelsey
    copyright       : (c) Vertical Online 2001
    title           : Vertical Online SV3 Demo
    encoder         : Lavf56.34.100
  Duration: 00:00:44.32, bitrate: 705 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 1 channels, s16, 705 kb/s
At least one output file must be specified

          #5033: Incorrect duration when converting WAV/MP3 files to AAC with -shortest   

Summary of the bug: Incorrect duration of encoded audio files when converting WAV/MP3 (MP3 created with the same WAV file) to AAC. I assume that this is affecting the duration of MP4 videos created with an still image and audio (which is the reason I'm reporting this bug).

How to reproduce: (espeak is a TTS engine. -w in espeak will save the output speech to a wav file)

$ espeak -s 80 -w /tmp/in.wav 'This product is meant for educational purposes only. Any resemblance to real persons, living or dead is purely coincidental. Void where prohibited. Some assembly required. List each check separately by bank number. Batteries not included. Contents may settle during shipment. Use only as directed. No other warranty expressed or implied. Do not use while operating a motor vehicle or heavy equipment. Postage will be paid by addressee. Subject to CARB approval.'
$ ffmpeg -y -i /tmp/in.wav -vn -c:a aac -strict -2 -ab 24k -ar 16000 /tmp/out.aac
ffmpeg version 2.8.2-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 5.2.1 (Debian 5.2.1-23) 20151028
  configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg --enable-libopus --enable-libass --enable-gnutls --enable-libvidstab --enable-libsoxr --enable-frei0r --enable-libfribidi --disable-indev=sndio --disable-outdev=sndio --cc=gcc
  libavutil      54. 31.100 / 54. 31.100
  libavcodec     56. 60.100 / 56. 60.100
  libavformat    56. 40.101 / 56. 40.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 40.101 /  5. 40.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc    53.  3.100 / 53.  3.100
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, wav, from '/tmp/in.wav':
  Duration: 00:01:00.33, bitrate: 352 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, 1 channels, s16, 352 kb/s
Output #0, adts, to '/tmp/out.aac':
  Metadata:
    encoder         : Lavf56.40.101
    Stream #0:0: Audio: aac, 16000 Hz, mono, fltp, 24 kb/s
    Metadata:
      encoder         : Lavc56.60.100 aac
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
size=     195kB time=00:01:00.35 bitrate=  26.5kbits/s
video:0kB audio:189kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3.422557%
$ for i in /tmp/in.wav /tmp/out.aac; do ffprobe -i $i 2>&1 | grep Duration; done
  Duration: 00:01:00.33, bitrate: 352 kb/s
  Duration: 00:03:19.68, bitrate: 8 kb/s

If you encode the WAV files to MP3 and then to AAC, once again, you get different, longer audio streams:

$ ffmpeg -y -i /tmp/in.wav -vn -strict -2 -ab 24k -ar 16000 -f mp3 /tmp/out.mp3
...
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
...
size=     177kB time=00:01:00.33 bitrate=  24.1kbits/s
$ ffmpeg -y -i /tmp/out.mp3 -vn -c:a aac -strict -2 -ab 24k -ar 16000 /tmp/out.aac
$ for i in /tmp/out.mp3 /tmp/out.aac; do ffprobe -i $i 2>&1 | grep Duration; done
  Duration: 00:01:00.41, start: 0.069063, bitrate: 24 kb/s
  Duration: 00:01:06.31, bitrate: 23 kb/s

Interestingly, I ran espeak with a different input ('This is some sample text to test audio encoding. ' repeated 10 times) and it gave me a shorter difference between the duration of the encoded output files.

I tried this with OGG/libvorbis files, the duration of output AAC file didn't seem to be affected.

For MP4s:

$ ffprobe -i /tmp/in.wav 2>&1 | grep Duration
  Duration: 00:01:00.33, bitrate: 352 kb/s
$ ffmpeg -y -framerate 1 -r 1 -loop 1 -i /tmp/in.jpg -i /tmp/in.wav -c:v libx264 -preset veryfast -tune stillimage -c:a aac -ab 32k -ar 16000 -strict experimental -shortest -pix_fmt yuv420p -movflags faststart -f mp4 /tmp/out.mp4
ffmpeg version 2.8.2-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 5.2.1 (Debian 5.2.1-23) 20151028
  configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg --enable-libopus --enable-libass --enable-gnutls --enable-libvidstab --enable-libsoxr --enable-frei0r --enable-libfribidi --disable-indev=sndio --disable-outdev=sndio --cc=gcc
  libavutil      54. 31.100 / 54. 31.100
  libavcodec     56. 60.100 / 56. 60.100
  libavformat    56. 40.101 / 56. 40.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 40.101 /  5. 40.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc    53.  3.100 / 53.  3.100
[mjpeg @ 0x4b1daa0] Changeing bps to 8
Input #0, image2, from '/tmp/in.jpg':
  Duration: 00:00:01.00, start: 0.000000, bitrate: 272 kb/s
    Stream #0:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 300x300 [SAR 1:1 DAR 1:1], 1 fps, 1 tbr, 1 tbn, 1 tbc
Guessed Channel Layout for  Input Stream #1.0 : mono
Input #1, wav, from '/tmp/in.wav':
  Duration: 00:01:00.33, bitrate: 352 kb/s
    Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, 1 channels, s16, 352 kb/s
[swscaler @ 0x4b59e20] deprecated pixel format used, make sure you did set range correctly
[libx264 @ 0x4b7eba0] using SAR=1/1
[libx264 @ 0x4b7eba0] using cpu capabilities: MMX2 SSE2Fast LZCNT
[libx264 @ 0x4b7eba0] profile High, level 1.2
[libx264 @ 0x4b7eba0] 264 - core 148 r209 7599210 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:-3:-3 analyse=0x3:0x113 me=hex subme=2 psy=1 psy_rd=2.00:0.70 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=6 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=1 keyint=250 keyint_min=1 scenecut=40 intra_refresh=0 rc_lookahead=10 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.20
Output #0, mp4, to '/tmp/out.mp4':
  Metadata:
    encoder         : Lavf56.40.101
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 300x300 [SAR 1:1 DAR 1:1], q=-1--1, 1 fps, 16384 tbn, 1 tbc
    Metadata:
      encoder         : Lavc56.60.100 libx264
    Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 16000 Hz, mono, fltp, 32 kb/s
    Metadata:
      encoder         : Lavc56.60.100 aac
Stream mapping:
  Stream #0:0 -> #0:0 (mjpeg (native) -> h264 (libx264))
  Stream #1:0 -> #0:1 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
[mp4 @ 0x4b38d40] Starting second pass: moving the moov atom to the beginning of the file
frame=   84 fps=0.0 q=-1.0 Lsize=     298kB time=00:01:22.00 bitrate=  29.8kbits/s
video:48kB audio:243kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.420870%
[libx264 @ 0x4b7eba0] frame I:1     Avg QP:13.91  size: 46896
[libx264 @ 0x4b7eba0] frame P:21    Avg QP:11.16  size:    31
[libx264 @ 0x4b7eba0] frame B:62    Avg QP:16.35  size:    15
[libx264 @ 0x4b7eba0] consecutive B-frames:  1.2%  0.0%  3.6% 95.2%
[libx264 @ 0x4b7eba0] mb I  I16..4:  0.6%  5.5% 93.9%
[libx264 @ 0x4b7eba0] mb P  I16..4:  0.0%  0.0%  0.0%  P16..4:  0.7%  0.0%  0.0%  0.0%  0.0%    skip:99.3%
[libx264 @ 0x4b7eba0] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8:  0.0%  0.0%  0.0%  direct: 0.0%  skip:100.0%
[libx264 @ 0x4b7eba0] 8x8 transform intra:5.5% inter:6.7%
[libx264 @ 0x4b7eba0] coded y,uvDC,uvAC intra: 97.9% 97.5% 94.7% inter: 0.0% 0.1% 0.0%
[libx264 @ 0x4b7eba0] i16 v,h,dc,p:  0%  0% 50% 50%
[libx264 @ 0x4b7eba0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu:  8% 51% 14%  4%  2%  1%  9%  5%  6%
[libx264 @ 0x4b7eba0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 14% 22% 16%  6%  8%  5%  8%  7% 13%
[libx264 @ 0x4b7eba0] i8c dc,h,v,p: 41% 27% 18% 14%
[libx264 @ 0x4b7eba0] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x4b7eba0] kb/s:4.62
$ ffprobe -i /tmp/out.mp4 2>&1 | grep Duration
  Duration: 00:01:24.00, start: 0.064000, bitrate: 29 kb/s

I talked in #ffmpeg @ freenode with another user who has the same problem with MP4s:

<user> llogan: http://sprunge.us/JYOS
<user> when i run that with a 4m45s audio stream, the first one has a duration of 4:50, the second one 4:56
<user> s/4:56/4:46/
<user> it's also not specific to aac or mp4
<user> and also according to mediainfo it's the video stream which is too long, not the audio stream
<user> and finally, increasing -r reduces the offset
<user> i assume it's an issue with the image2 muxer, but i don't know for sure
<user> and by for sure i of course mean at all

          #5258: ffmpeg fails to decode opus test vectors correctly   

taken from https://trac.ffmpeg.org/ticket/5238#comment:3

ffmpeg fails to decode opus test vectors correctly, in especially: http://people.xiph.org/~greg/opus_testvectors/correctness_trimming_nobeeps.opus

even using ffmpeg git:

$ ./ffmpeg -i correctness_trimming_nobeeps.opus correctness_trimming_nobeeps.wav
ffmpeg version N-78622-g2b5b001 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.1 (Debian 5.3.1-9) 20160220
  configuration: --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv --enable-version3 --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libvo_amrwbenc
  libavutil      55. 18.100 / 55. 18.100
  libavcodec     57. 24.105 / 57. 24.105
  libavformat    57. 26.100 / 57. 26.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 34.100 /  6. 34.100
  libavresample   3.  0.  0 /  3.  0.  0
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, ogg, from 'correctness_trimming_nobeeps.opus':
  Duration: 00:00:12.72, start: 0.000000, bitrate: 40 kb/s
    Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
    Metadata:
      ENCODER         : Xiph.Org Opus testvectormaker
      TESTDESCRIPTION : Beeps at the beginning and end of this stream MUST be hidden by preskip and end-trimming.
                      : If the beeps are heard the decoder is broken.
Output #0, wav, to 'correctness_trimming_nobeeps.wav':
  Metadata:
    ISFT            : Lavf57.26.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      TESTDESCRIPTION : Beeps at the beginning and end of this stream MUST be hidden by preskip and end-trimming.
                      : If the beeps are heard the decoder is broken.
      encoder         : Lavc57.24.105 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (opus (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    1193kB time=00:00:11.35 bitrate= 860.4kbits/s speed= 211x
video:0kB audio:1193kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.006388%

the resulting WAV files still contains the beep


          #5468: on2avc (0x0501): improve decoding of stereo modes   

https://samples.mplayerhq.hu/V-codecs/VP6/wagner-700.vp6

https://samples.mplayerhq.hu/V-codecs/VP5/BeautifulMind_640vp5-750.vp5

https://samples.mplayerhq.hu/V-codecs/VP6/Xmen-avc-500.vp6

three more samples are here:

https://www.datafilehost.com/d/05f10ede

C:\>ffmpeg -i wagner-700.vp6 out.wav
ffmpeg version N-79546-g13406b6 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-pthreads --enable-memalign-hack --enable-runtime-cpudetect --
enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -lw
inmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snapsh
ots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/sna
pshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3 --
enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-li
bvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-li
bopencore-amrwb --enable-libmp3lame --enable-libfreetype --enable-libvpx --disab
le-decoder=libvpx
  libavutil      55. 22.100 / 55. 22.100
  libavcodec     57. 35.100 / 57. 35.100
  libavformat    57. 34.102 / 57. 34.102
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 44.100 /  6. 44.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[on2avc @ 0x2c47d80] Stereo mode support is not good, patch is welcome
Input #0, avi, from 'wagner-700.vp6':
  Duration: 00:02:49.23, start: 0.000000, bitrate: 816 kb/s
    Stream #0:0: Video: vp6 (VP62 / 0x32365056), yuv420p, 624x352, 700 kb/s, 29.
97 fps, 29.97 tbr, 29.97 tbn
    Stream #0:1: Audio: avc ([1][5][0][0] / 0x0501), 44100 Hz, stereo, fltp, 96
kb/s
[on2avc @ 0x2899020] Stereo mode support is not good, patch is welcome
[wav @ 0x289c020] Using AVStream.codec to pass codec parameters to muxers is dep
recated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.34.102
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16
, 1411 kb/s
    Metadata:
      encoder         : Lavc57.35.100 pcm_s16le
Stream mapping:
  Stream #0:1 -> #0:0 (avc (on2avc) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=   29152kB time=00:02:49.22 bitrate=1411.2kbits/s speed= 285x
video:0kB audio:29152kB subtitle:0kB other streams:0kB global headers:0kB muxing
 overhead: 0.000261%

          #5482: g721 (g726): fix audiable artefacts   

decodes ok with sox

https://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket1955/g721_1ch_44khz.au

also reproducible with g726 sample:

https://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket1955/g726_44khz_2bit.au

C:\>ffmpeg -i C:\g721_1ch_44khz.au out.wav
ffmpeg version N-79632-g3ce1988 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-pthreads --enable-memalign-hack --enable-runtime-cpudetect --
enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -lw
inmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snapsh
ots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/sna
pshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3 --
enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-li
bvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-li
bopencore-amrwb --enable-libmp3lame --enable-libfreetype --enable-libvpx --disab
le-decoder=libvpx
  libavutil      55. 22.101 / 55. 22.101
  libavcodec     57. 38.100 / 57. 38.100
  libavformat    57. 34.103 / 57. 34.103
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 44.100 /  6. 44.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, au, from 'C:\g721_1ch_44khz.au':
  Duration: 00:00:30.01, start: 0.000000, bitrate: 176 kb/s
    Stream #0:0: Audio: adpcm_g726le ([23][0][0][0] / 0x0017), 44100 Hz, mono, s
16, 176 kb/s
[wav @ 0x2c3eb60] Using AVStream.codec to pass codec parameters to muxers is dep
recated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.34.103
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16,
705 kb/s
    Metadata:
      encoder         : Lavc57.38.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_g726le (g726le) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    2585kB time=00:00:30.01 bitrate= 705.6kbits/s speed= 213x
video:0kB audio:2585kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.002946%

          #5942: mpeg-4 als: "quant_cof 237 is out of range" with multichannel 16bit   

no such problem with mono/stereo files

encoded with:

C:\>mp4alsRM23.exe -7 -MP4 D:\16multi.wav
C:\>ffmpeg -i D:\16multi.mp4  D:\out.wav
ffmpeg version N-78758-g5156578 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
ble-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --
enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-lib
x265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-d
ecklink --enable-zlib
  libavutil      55. 19.100 / 55. 19.100
  libavcodec     57. 27.100 / 57. 27.100
  libavformat    57. 26.100 / 57. 26.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 37.100 /  6. 37.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Guessed Channel Layout for  Input Stream #0.0 : 5.1
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'D:\16multi.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    creation_time   : 2016-11-11 17:28:53
  Duration: 00:00:05.00, start: 0.000000, bitrate: 1220 kb/s
    Stream #0:0(und): Audio: mp4als (mp4a / 0x6134706D), 48000 Hz, 6 channels, s
16, 1219 kb/s (default)
    Metadata:
      creation_time   : 2016-11-11 17:28:53
Output #0, wav, to 'D:\out.wav':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    ISFT            : Lavf57.26.100
    Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1, s
16, 4608 kb/s (default)
    Metadata:
      creation_time   : 2016-11-11 17:28:53
      encoder         : Lavc57.27.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (mp4als (als) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[als @ 038f0d60] quant_cof 237 is out of range.
[als @ 038f0d60] Reading frame data failed. Skipping RA unit.
size=     720kB time=00:00:01.28 bitrate=4608.6kbits/s speed=N/A
video:0kB audio:720kB subtitle:0kB other streams:0kB global headers:0kB muxing o
verhead: 0.013835%

          #5943: mpeg-4 als: "Overread 8" with 8bit file   

encoded with:

C:\>mp4alsRM23.exe -MP4 -7 -i d:\8mono_7_i.wav
C:\>ffmpeg -i D:\8mono_7_i.mp4 D:\out.wav
ffmpeg version N-78758-g5156578 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
ble-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --
enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-lib
x265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-d
ecklink --enable-zlib
  libavutil      55. 19.100 / 55. 19.100
  libavcodec     57. 27.100 / 57. 27.100
  libavformat    57. 26.100 / 57. 26.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 37.100 /  6. 37.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'D:\8mono_7_i.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    creation_time   : 2016-11-12 14:06:09
  Duration: 00:00:05.00, start: 0.000000, bitrate: 45 kb/s
    Stream #0:0(und): Audio: mp4als (mp4a / 0x6134706D), 48000 Hz, 1 channels, s
16 (8 bit), 44 kb/s (default)
    Metadata:
      creation_time   : 2016-11-12 14:06:09
Output #0, wav, to 'D:\out.wav':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    ISFT            : Lavf57.26.100
    Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono,
s16 (8 bit), 768 kb/s (default)
    Metadata:
      creation_time   : 2016-11-12 14:06:09
      encoder         : Lavc57.27.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (mp4als (als) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[als @ 03906020] Overread 8
[als @ 03906020] Reading frame data failed. Skipping RA unit.
size=     509kB time=00:00:05.42 bitrate= 768.1kbits/s speed= 174x
video:0kB audio:509kB subtitle:0kB other streams:0kB global headers:0kB muxing o
verhead: 0.014972%

          #6035: wmavoice: files decodes incorrectly   

4kbps 8khz and 5kbps 8khz modes decodes ok, the rest don't

https://trac.ffmpeg.org/raw-attachment/ticket/5426/wmavoice.7z

C:\>ffmpeg.exe -i C:\2_16kbps_16khz.wma out.wav
ffmpeg version 3.2.git Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.6.2 (GCC)
  configuration: --disable-pthreads --enable-gpl
  libavutil      55. 43.100 / 55. 43.100
  libavcodec     57. 68.100 / 57. 68.100
  libavformat    57. 61.100 / 57. 61.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 68.100 /  6. 68.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
[wmavoice @ 0224d5c0] Warning: not compiled with thread support, using thread em
ulation
Input #0, asf, from 'C:\2_16kbps_16khz.wma':
  Metadata:
    WMFSDKNeeded    : 0.0.0.0000
    DeviceConformanceTemplate: S1
    WMFSDKVersion   : 11.0.5721.5275
    IsVBR           : 0
  Duration: 00:00:37.98, start: 0.000000, bitrate: 17 kb/s
    Stream #0:0(pol): Audio: wmavoice ([10][0][0][0] / 0x000A), 16000 Hz, mono,
flt, 16 kb/s
[wmavoice @ 025ffa20] Warning: not compiled with thread support, using thread em
ulation
[pcm_s16le @ 02247e00] Warning: not compiled with thread support, using thread e
mulation
Output #0, wav, to 'out.wav':
  Metadata:
    WMFSDKNeeded    : 0.0.0.0000
    DeviceConformanceTemplate: S1
    WMFSDKVersion   : 11.0.5721.5275
    IsVBR           : 0
    ISFT            : Lavf57.61.100
    Stream #0:0(pol): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono,
s16, 256 kb/s
    Metadata:
      encoder         : Lavc57.68.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (wmavoice (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Error while decoding stream #0:0: Function not implementeded=13.5x
    Last message repeated 3 times
size=    1188kB time=00:00:37.97 bitrate= 256.2kbits/s speed=13.4x
video:0kB audio:1187kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.006415%

          #6036: sipr: improve decoding of 16khz 16kbps mode   

this bug deserves it's own ticket

there are metalic artifacts between 21s-32s

C:\>ffmpeg -i sipr_16k_could_be_better.rm out.wav
ffmpeg version 3.2.git Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.6.2 (GCC)
  configuration: --disable-pthreads --enable-gpl
  libavutil      55. 43.100 / 55. 43.100
  libavcodec     57. 68.100 / 57. 68.100
  libavformat    57. 61.100 / 57. 61.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 68.100 /  6. 68.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
[rm @ 02243d60] Invalid stream index 1 for index at pos 81173
[sipr @ 0224d380] Warning: not compiled with thread support, using thread emulat
ion
Input #0, rm, from 'sipr_16k_could_be_better.
rm':
  Metadata:
    Generated By    : RealProducer 6.1.0.153 Windows
    Content Rating  : (PICS-1.1 "http://www.classify.org/safesurf" labels commen
t "RealProducer 6.1.0.153 Windows" ratings (SS~~000 1))
    Target Audiences: 28K Modem (20 Kbps);
    Audio Format    : Voice with Background Music
    Video Quality   : No Video
    Creation Date   : 12/19/2016 3:26:44
    Modification Date: 12/19/2016 3:26:44
    File ID         : 5001d8cc-b0c8-5861-36e0-07a19838af8b
    Keywords        :
    Abstract        :
    title           : Inp
    author          :
    copyright       : ę2016
    comment         :
  Duration: 00:00:38.31, start: 0.000000, bitrate: 16 kb/s
    Stream #0:0: Audio: sipr (sipr / 0x72706973), 16000 Hz, mono, flt, 16 kb/s
[sipr @ 025d9c00] Warning: not compiled with thread support, using thread emulat
ion
[pcm_s16le @ 02247140] Warning: not compiled with thread support, using thread e
mulation
Output #0, wav, to 'out.wav':
  Metadata:
    Generated By    : RealProducer 6.1.0.153 Windows
    Content Rating  : (PICS-1.1 "http://www.classify.org/safesurf" labels commen
t "RealProducer 6.1.0.153 Windows" ratings (SS~~000 1))
    Target Audiences: 28K Modem (20 Kbps);
    Audio Format    : Voice with Background Music
    Video Quality   : No Video
    Creation Date   : 12/19/2016 3:26:44
    Modification Date: 12/19/2016 3:26:44
    File ID         : 5001d8cc-b0c8-5861-36e0-07a19838af8b
    Keywords        :
    Abstract        :
    INAM            : Inp
    author          :
    ICOP            : ę2016
    ICMT            :
    ISFT            : Lavf57.61.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16,
256 kb/s
    Metadata:
      encoder         : Lavc57.68.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (sipr (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    1200kB time=00:00:38.31 bitrate= 256.6kbits/s speed= 189x
video:0kB audio:1200kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.008464%

          #6046: mpc8: "maxband 19 too large" / "overread 8"   
C:\>ffmpeg -i q1.mpc out.wav
ffmpeg version 3.2.git Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.6.2 (GCC)
  configuration: --disable-pthreads --enable-gpl
  libavutil      55. 43.100 / 55. 43.100
  libavcodec     57. 68.100 / 57. 68.100
  libavformat    57. 61.100 / 57. 61.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 68.100 /  6. 68.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
[mpc8 @ 0224d2e0] Warning: not compiled with thread support, using thread emulat
ion
Input #0, mpc8, from 'q1.mpc':
  Duration: 00:00:58.51, start: 0.000000, bitrate: 17 kb/s
    Stream #0:0: Audio: musepack8, 44100 Hz, mono, s16p
[mpc8 @ 025db260] Warning: not compiled with thread support, using thread emulat
ion
[pcm_s16le @ 025dc9c0] Warning: not compiled with thread support, using thread e
mulation
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.61.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16,
705 kb/s
    Metadata:
      encoder         : Lavc57.68.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (musepack8 (mpc8) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[mpc8 @ 025db260] Multiple frames in a packet.
[mpc8 @ 025db260] maxband 19 too large
Error while decoding stream #0:0: Invalid data found when processing input
q1.mpc: I/O error
size=    5171kB time=00:01:00.02 bitrate= 705.6kbits/s speed= 384x
video:0kB audio:5170kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.001473%
C:\>ffmpeg -i q10.mpc out.wav
ffmpeg version 3.2.git Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.6.2 (GCC)
  configuration: --disable-pthreads --enable-gpl
  libavutil      55. 43.100 / 55. 43.100
  libavcodec     57. 68.100 / 57. 68.100
  libavformat    57. 61.100 / 57. 61.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 68.100 /  6. 68.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
[mpc8 @ 0224d300] Warning: not compiled with thread support, using thread emulat
ion
Input #0, mpc8, from 'q10.mpc':
  Duration: 00:00:58.51, start: 0.000000, bitrate: 123 kb/s
    Stream #0:0: Audio: musepack8, 44100 Hz, mono, s16p
[mpc8 @ 0224c040] Warning: not compiled with thread support, using thread emulat
ion
[pcm_s16le @ 025fd600] Warning: not compiled with thread support, using thread e
mulation
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.61.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16,
705 kb/s
    Metadata:
      encoder         : Lavc57.68.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (musepack8 (mpc8) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[mpc8 @ 0224c040] Multiple frames in a packet.
[mpc8 @ 0224c040] Overread 8
q10.mpc: I/O error
size=    5173kB time=00:01:00.05 bitrate= 705.6kbits/s speed= 384x
video:0kB audio:5173kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.001473%

          #667: Error when decoding TrueHD 1.0 track that also contains AC-3   

I am trying to use ffmpeg to decode a TrueHD 1.0 track to WAV for encoding to FLAC. Originally I was using eac3to to accomplish this, and received an error. I was told this had to do with the outdated ffmpeg used by eac3to, and that I should run the following command line with the latest ffmpeg build:

ffmpeg -i japanese.thd -ac 1 -acodec pcm_s24le -f wav japanese.wav

This command produces the following error message:

[truehd @ 0000000001C80290] mlpparse: Parity check failed.

Last message repeated 228 times

[truehd @ 000000000029D570] max_analyze_duration 5000000 reached at 5000000 [truehd @ 000000000029D570] Estimating duration from bitrate, this may be inaccurate Input #0, truehd, from 'japanese.thd':

Duration: N/A, start: 0.000000, bitrate: N/A

Stream #0:0: Audio: truehd, 48000 Hz, mono, s32

Output #0, wav, to 'japanese.wav':

Metadata:

encoder : Lavf53.20.0 Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels, s32, 1152 kb/s

Stream mapping:

Stream #0:0 -> #0:0 (truehd -> pcm_s24le)

Press [q] to stop, ? for help Input stream #0:0 frame changed from rate:48000 fmt:s32 ch:1 to rate:48000 fmt:s32 ch:2 [SWR @ 0000000001CA18D0] Input channel layout has a different number of channels than the number of used channels, ignoring layout [truehd @ 0000000001C80290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C80290] Lossless check failed - expected 05, calculated 00. [truehd @ 0000000001C80290] Lossless check failed - expected 05, calculated 0f. [truehd @ 0000000001C80290] Lossless check failed - expected 09, calculated 06. [truehd @ 0000000001C80290] Lossless check failed - expected 0c, calculated 12. [truehd @ 0000000001C80290] Lossless check failed - expected 00, calculated 0a. [truehd @ 0000000001C80290] Invalid nonrestart_substr. Error while decoding stream #0:0

This error continues until the program halts and stops processing the file. According to the error above, this track is a mix of 1.0 and 2.0, so I tried running the same command with -ac 2 instead and received the same error message.

Here is the output produced by ffmpeg -v 9 -loglevel 99 -i japanese.thd -f wav japanese.wav:

ffmpeg version N-34906-g4e7b3ef, Copyright (c) 2000-2011 the FFmpeg developers

built on Nov 16 2011 12:39:54 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 25. 0 / 51. 25. 0 libavcodec 53. 34. 0 / 53. 34. 0 libavformat 53. 20. 0 / 53. 20. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 48. 1 / 2. 48. 1 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0

[truehd @ 00000000003AD540] Format truehd probed with size=2048 and score=50 [truehd @ 0000000001C10290] err{or,}_recognition separate: 1; 1 [truehd @ 0000000001C10290] err{or,}_recognition combined: 1; 1 [truehd @ 0000000001C10290] Unsupported bit depth: 0 [truehd @ 0000000001C10290] mlpparse: Parity check failed.

Last message repeated 228 times

[truehd @ 00000000003AD540] max_analyze_duration 5000000 reached at 5000000 [truehd @ 00000000003AD540] Estimating duration from bitrate, this may be inaccurate Input #0, truehd, from 'japanese.thd':

Duration: N/A, start: 0.000000, bitrate: N/A

Stream #0:0, 6002, 1/90000: Audio: truehd, 48000 Hz, mono, s32

Incompatible sample format 's32' for codec 'pcm_s16le', auto-selecting format 's16' [pcm_s16le @ 0000000003B05820] err{or,}_recognition separate: 1; 1 [pcm_s16le @ 0000000003B05820] err{or,}_recognition combined: 1; 1 [truehd @ 0000000001C10290] err{or,}_recognition separate: 1; 1 [truehd @ 0000000001C10290] err{or,}_recognition combined: 1; 1 [truehd @ 0000000001C10290] Unsupported bit depth: 24 Output #0, wav, to 'japanese.wav':

Metadata:

encoder : Lavf53.20.0 Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s

Stream mapping:

Stream #0:0 -> #0:0 (truehd -> pcm_s16le)

Press [q] to stop, ? for help [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. Input stream #0:0 frame changed from rate:48000 fmt:s32 ch:1 to rate:48000 fmt:s32 ch:2 [SWR @ 0000000001C318D0] Input channel layout has a different number of channels than the number of used channels, ignoring layout 0.500000 0.500000 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped.

Last message repeated 2 times

[truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 00. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 0f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped.

Last message repeated 1 times

[truehd @ 0000000001C10290] Lossless check failed - expected 09, calculated 06. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0c, calculated 12. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 00, calculated 0a. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 03. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 35, calculated 1d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 96, calculated 1b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 18, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 21, calculated e8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected dd, calculated 8e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 99, calculated bb. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d2, calculated a5. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 69, calculated a9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d2, calculated e8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2d, calculated 11. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected db, calculated a9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6a, calculated c9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6a, calculated 1e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 8d, calculated ee. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a9, calculated 4e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f3, calculated b8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b1, calculated 21. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 93, calculated 06. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6a, calculated 2b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped.

Last message repeated 1 times

[truehd @ 0000000001C10290] Lossless check failed - expected 21, calculated 56. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5c, calculated b2. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 65, calculated 7e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated d8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated a9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated 00. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c5, calculated d4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a6, calculated a5. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 4b, calculated 3c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected fc, calculated 99. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5a, calculated 1b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected af, calculated 6a. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Stream parameters not seen; skipping frame. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d2, calculated c5. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 9a, calculated 39. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0a, calculated 17. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 2b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f5, calculated e2. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 39, calculated 06. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 21, calculated ee. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 81, calculated 6a. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated a5. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1e, calculated 12. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a6, calculated a9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 06, calculated d2. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 36, calculated 69. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7d, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f9, calculated e4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated 2b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 59, calculated 56. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 48, calculated dd. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 24, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3c, calculated ed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f3, calculated b1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 17, calculated 28. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 82, calculated 7e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c6, calculated 9a. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 33, calculated e8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e7, calculated 87. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7e, calculated 56. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Stream parameters not seen; skipping frame. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c3, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 99, calculated 95. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 35, calculated bb. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a3, calculated 8d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected cc, calculated 59. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated 5f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 50, calculated e8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 95, calculated d7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 78, calculated 71. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ca, calculated 72. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected bb, calculated 9c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7d, calculated c6. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6a, calculated aa. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d1, calculated a6. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated d1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 9c, calculated 39. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected dd, calculated 24. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 28, calculated 4e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e7, calculated 41. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected bb, calculated b7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b7, calculated 88. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 69, calculated db. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5a, calculated 44. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7e, calculated 5f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0c, calculated e8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 93, calculated 5f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected af, calculated d2. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 11, calculated cc. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 03, calculated ed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e7, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 03, calculated cc. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3c, calculated d4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e1, calculated bb. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 41, calculated 50. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f9, calculated 05. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected fc, calculated cc. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected fc, calculated e4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 22, calculated 0a. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b8, calculated db. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated 39. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c3, calculated 8b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 90, calculated 41. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 18, calculated e2. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 8e, calculated ff. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 0f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected aa, calculated 0a. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c3, calculated 8d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 41, calculated 90. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c5, calculated 28. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected bb, calculated 2b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2b, calculated 09. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c0, calculated 7d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e7, calculated 69. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e7, calculated 7e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected dd, calculated 6f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 99, calculated f0. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ff, calculated e4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6c, calculated 4e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3a, calculated 14. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2b, calculated bd. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e8, calculated 50. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b2, calculated 5c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 55, calculated b4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 33, calculated 8b. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b1, calculated 81. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated f3. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 12, calculated b1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 47, calculated cf. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7e, calculated 39. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3f, calculated 33. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c6, calculated b1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c5, calculated 56. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected db, calculated 47. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c5, calculated a9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 88, calculated 2d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 9f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 59, calculated 6c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d1, calculated 27. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2b, calculated 66. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated 56. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 48, calculated 05. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected be, calculated e7. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0c, calculated 28. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 47, calculated af. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2d, calculated 59. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 84, calculated ac. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 56, calculated d4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b1, calculated 4b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 33, calculated 6c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected fa, calculated 2b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 12, calculated 59. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b8, calculated 5c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ee, calculated 66. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3a, calculated e7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated e1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1e, calculated 5f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e2, calculated 11. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 96, calculated 11. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1e, calculated 05. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1d, calculated 7d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 06, calculated e7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 22, calculated 95. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 77, calculated 71. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 4b, calculated 95. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated e1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e1, calculated bd. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b4, calculated aa. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b8, calculated 6a. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 17, calculated 03. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 82, calculated 2b. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected be, calculated 4d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1b, calculated 9c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3f, calculated 6f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 30, calculated eb. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a3, calculated a5. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5f, calculated 17. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ca, calculated 05. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a5, calculated b7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a9, calculated 87. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 82, calculated a0. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a6, calculated de. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 71, calculated cf. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 4d, calculated 9c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a5, calculated e7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0f, calculated c3. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ed, calculated a6. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated 42. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 82, calculated 72. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated 8b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2e, calculated db. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected dd, calculated fc. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f5, calculated 1b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 39, calculated 47. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7b, calculated 30. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Stream parameters not seen; skipping frame. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0c, calculated 93. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 66, calculated 1e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 12, calculated de. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 4d, calculated bd. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c6, calculated 17. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected fc, calculated db. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3f, calculated 11. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b4, calculated 0c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated b2. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 56, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7e, calculated b7. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d1, calculated 7b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 99, calculated db. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 81, calculated ff. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated 5f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 47, calculated b4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3a, calculated 4b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 71, calculated 9f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated 03. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1b, calculated b8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ac, calculated b8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated b1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ee, calculated f6. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c3, calculated cc. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f5, calculated 93. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 12, calculated 5f. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6a, calculated f6. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 00, calculated 0f. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 90, calculated cc. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ac, calculated 18. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5c, calculated 81. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0a, calculated be. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 4d, calculated 5c. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 77, calculated bb. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6c, calculated a0. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 30, calculated 66. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5c, calculated a0. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c0, calculated e2. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f5, calculated 3f. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 35, calculated 0a. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated 78. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated b7. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ac, calculated 56. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0a, calculated ff. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 14, calculated d1. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3a, calculated 95. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 8e, calculated 56. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f9, calculated 53. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2e, calculated aa. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 77, calculated 03. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ee, calculated 12. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c9, calculated 1b. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f9, calculated 87. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b2, calculated 55. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c6, calculated 17. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 14, calculated 18. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 06, calculated aa. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 53, calculated 95. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated 1d. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 71, calculated d8. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ff, calculated 66. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected dd, calculated d2. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated 4b. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d8, calculated 8e. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected eb, calculated 95. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7d, calculated 55. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 65, calculated f9. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 90, calculated ee. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3c, calculated 33. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c6, calculated e8. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7d, calculated 44. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6c, calculated cc. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 09, calculated 24. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 66, calculated ed. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected eb, calculated 3a. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 99, calculated 4d. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ca, calculated e8. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 8e, calculated 30. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 84, calculated 1d. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e4, calculated 48. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 95, calculated 78. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Loss

          #2459: mpc8: wrong duration   

ffmpeg reports 28s, mpcdec 30s

1323648 / 44100 gives the correct duration

http://ffmpeg.org/trac/ffmpeg/raw-attachment/ticket/1254/musepack8.mpc

$ mpcdec -i musepack8.mpc
mpcdec - Musepack (MPC) decoder v1.0.0 (C) 2006-2009 MDT
Built Feb 26 2009 20:20:48
file: musepack8.mpc
stream version 8
encoder: --Stable-- 1.30.0
profile: 'Standard' (q=5.00)
PNS: off
mid/side stereo: on
gapless: on
average bitrate:  176.5 kbps
samplerate: 44100 Hz
channels: 2
length: 0:30 (1323648 samples)
file size: 662373 Bytes
track peak: 0.00 dB
track gain: 0.00 dB / 0.00 dB
album peak: 0.00 dB
album gain: 0.00 dB / 0.00 dB
C:\>ffmpeg -i musepack8.mpc out.wav
ffmpeg version N-51683-g9dc88ac Copyright (c) 2000-2013 the FFmpeg developers
  built on Apr  8 2013 21:19:21 with gcc 4.8.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrn
b --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libr
tmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheo
ra --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li
bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --ena
ble-zlib
  libavutil      52. 25.100 / 52. 25.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  1.100 / 55.  1.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 49.101 /  3. 49.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[mpc8 @ 026e4020] max_analyze_duration 5000000 reached at 5015510 microseconds
Input #0, mpc8, from 'musepack8.mpc':
  Duration: 00:00:28.42, start: 0.000000, bitrate: 186 kb/s
    Stream #0:0: Audio: musepack8, 44100 Hz, stereo, s16p
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf55.1.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16
, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mpc8 -> pcm_s16le)
Press [q] to stop, [?] for help
Multiple frames in a packet from stream 0
musepack8.mpc: Input/output error
size=    5175kB time=00:00:30.04 bitrate=1411.2kbits/s
video:0kB audio:5175kB subtitle:0 global headers:0kB muxing overhead 0.001510%

          #5448: aac: wrong calculation of duration   

it used to be correct long time ago:

C:\>ffmpeg -i 1.aac
ffmpeg version 2.1.3 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jan 21 2014 18:55:26 with gcc 4.8.2 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa
cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp
ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 48.101 / 52. 48.101
  libavcodec     55. 39.101 / 55. 39.101
  libavformat    55. 19.104 / 55. 19.104
  libavdevice    55.  5.100 / 55.  5.100
  libavfilter     3. 90.100 /  3. 90.100
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100
[aac @ 02a7a980] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from '1.aac':
  Duration: 00:00:29.58, bitrate: 73 kb/s
    Stream #0:0: Audio: aac, 88200 Hz, mono, fltp, 73 kb/s
At least one output file must be specified
C:\>ffmpeg -i 1.aac out.wav
ffmpeg version N-79468-ga676c29 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-pthreads --enable-memalign-hack --enable-runtime-cpudetect --
enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -lw
inmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snapsh
ots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/sna
pshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3 --
enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-li
bvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-li
bopencore-amrwb --enable-libmp3lame --enable-libfreetype --enable-libvpx --disab
le-decoder=libvpx
  libavutil      55. 22.100 / 55. 22.100
  libavcodec     57. 34.102 / 57. 34.102
  libavformat    57. 34.101 / 57. 34.101
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 42.100 /  6. 42.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[aac @ 0x2894160] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from '1.aac':
  Duration: 00:04:47.72, bitrate: 7 kb/s
    Stream #0:0: Audio: aac (LC), 88200 Hz, mono, fltp, 7 kb/s
File 'out.wav' already exists. Overwrite ? [y/N] y
[wav @ 0x2897420] Using AVStream.codec to pass codec parameters to muxers is dep
recated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.34.101
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 88200 Hz, mono, s16,
1411 kb/s
    Metadata:
      encoder         : Lavc57.34.102 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 0; changing to 88057. This may result in incorrect timestamps in the output
 file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 1024; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 2048; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 3072; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 4096; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 5120; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 6144; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 7168; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 8192; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 9216; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 10240; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 11264; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 12288; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 13312; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 14336; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 15360; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 16384; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 17408; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 18432; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 19456; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 20480; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 21504; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 22528; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 23552; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 24576; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 25600; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 26624; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 27648; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 28672; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 29696; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 30720; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 31744; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 32768; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 33792; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 34816; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 35840; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 36864; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 37888; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 38912; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 39936; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 40960; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 41984; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 43008; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 44032; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 45056; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 46080; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 47104; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 48128; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 49152; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 50176; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 51200; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 52224; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 53248; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 54272; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 55296; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 56320; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 57344; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 58368; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 59392; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 60416; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 61440; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 62464; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 63488; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 64512; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 65536; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 66560; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 67584; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 68608; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 69632; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 70656; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 71680; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 72704; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 73728; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 74752; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 75776; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 76800; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 77824; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 78848; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 79872; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 80896; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 81920; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 82944; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 83968; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 84992; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 86016; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 87040; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
size=    5188kB time=00:00:29.10 bitrate=1460.2kbits/s speed= 133x
video:0kB audio:5188kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.001920%

          #5501: EAC3 decoding   

Problem decoding EAC3 from TS, if track is repackaged with "-copy_unkown", then the stream is a FFmpeg decodable AC3 384k...

Any way to decode original file without the temp file?

  1. FAILED TO DECODE

ffmpeg -i part.ts -map 0:1 -t 10 -f null -

ffmpeg version N-79000-g66edd86 Copyright (c) 2000-2016 the FFmpeg developers

built with gcc 5.3.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib libavutil 55. 19.100 / 55. 19.100 libavcodec 57. 28.100 / 57. 28.100 libavformat 57. 28.100 / 57. 28.100 libavdevice 57. 0.101 / 57. 0.101 libavfilter 6. 39.102 / 6. 39.102 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100

[mpegts @ 00000000007a2700] probed stream 1 failed [mpegts @ 00000000007a2700] Could not find codec parameters for stream 1 (Unknown: none (EAC3 / 0x33434145)): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mpegts, from 'part.ts':

Duration: N/A, start: 1.083400, bitrate: N/A Program 2

Stream #0:0[0x1e1]: Video: h264 (High 4:2:2) ([27][0][0][0] / 0x001B), yuv422p10le(tv, bt709), 3840x2160 [SAR 1:1 DAR 16:9],

Closed Captions, 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc

Stream #0:1[0x1e2]: Unknown: none (EAC3 / 0x33434145)

Cannot map stream #0:1 - unsupported type. If you want unsupported types ignored instead of failing, please use the -ignore_unknown option If you want them copied, please use -copy_unknown

  1. COPIED TRACK TO TS with "-copy_unknown"

ffmpeg -i 108GB_9.ts -map 0:1 -t 10 -copy_unknown -c copy audio_test.ts

ffmpeg version N-79000-g66edd86 Copyright (c) 2000-2016 the FFmpeg developers

built with gcc 5.3.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib libavutil 55. 19.100 / 55. 19.100 libavcodec 57. 28.100 / 57. 28.100 libavformat 57. 28.100 / 57. 28.100 libavdevice 57. 0.101 / 57. 0.101 libavfilter 6. 39.102 / 6. 39.102 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100

[mpegts @ 0000000000662820] probed stream 1 failed [mpegts @ 0000000000662820] Could not find codec parameters for stream 1 (Unknown: none (EAC3 / 0x33434145)): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mpegts, from '108GB_9.ts':

Duration: N/A, start: 1.083400, bitrate: N/A Program 2

Stream #0:0[0x1e1]: Video: h264 (High 4:2:2) ([27][0][0][0] / 0x001B), yuv422p10le(tv, bt709), 3840x2160 [SAR 1:1 DAR 16:9],

Closed Captions, 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc

Stream #0:1[0x1e2]: Unknown: none (EAC3 / 0x33434145)

Output #0, mpegts, to 'audio_test.ts':

Metadata:

encoder : Lavf57.28.100 Stream #0:0: Unknown: none (EAC3 / 0x33434145)

Stream mapping:

Stream #0:1 -> #0:0 (copy)

Press [q] to stop, ? for help size= 540kB time=00:00:09.98 bitrate= 443.5kbits/s speed= 35x video:0kB audio:0kB subtitle:0kB other streams:473kB global headers:0kB muxing overhead: 14.242310%

  1. CONVERT TEST AUDIO FILE to WAV - OK!

ffmpeg -i audio_test.ts 1.wav

ffmpeg version N-79000-g66edd86 Copyright (c) 2000-2016 the FFmpeg developers

built with gcc 5.3.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib libavutil 55. 19.100 / 55. 19.100 libavcodec 57. 28.100 / 57. 28.100 libavformat 57. 28.100 / 57. 28.100 libavdevice 57. 0.101 / 57. 0.101 libavfilter 6. 39.102 / 6. 39.102 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100

Input #0, mpegts, from 'audio_test.ts':

Duration: 00:00:09.98, start: 1.400000, bitrate: 443 kb/s Program 1

Metadata:

service_name : Service01 service_provider: FFmpeg

Stream #0:0[0x100]: Audio: eac3 ([6][0][0][0] / 0x0006), 48000 Hz, 5.1(side), fltp, 384 kb/s

Output #0, wav, to '1.wav':

Metadata:

ISFT : Lavf57.28.100 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1(side), s16, 4608 kb/s Metadata:

encoder : Lavc57.28.100 pcm_s16le

Stream mapping:

Stream #0:0 -> #0:0 (eac3 (native) -> pcm_s16le (native))

Press [q] to stop, ? for help [eac3 @ 000000000078ef20] incomplete frame size= 5652kB time=00:00:10.04 bitrate=4608.1kbits/s speed= 141x video:0kB audio:5652kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.001762%


          #5828: Duration mismatch on opus to pcm conversion   

Summary of the bug:

There is a random non negligible duration mismatch when converting from opus to PCM

How to reproduce:

% ffmpeg -i test.opus out.wav
ffmpeg version N-81528-g39f7564-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.4.1 (Debian 5.4.1-1) 20160803
libopus version 1.1.3
% ffprobe test.opus
Input #0, ogg, from 'test.opus':
  Duration: 00:00:22.81, start: 1.346000, bitrate: 27 kb/s
    Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp, delay 48, padding 0
    Metadata:
      LANGUAGE        : eng
      ENCODER         : Lavf57.48.101
% ffprobe output.wav
Input #0, wav, from 'output.wav':
  Metadata:
    encoder         : Lavf57.48.101
  Duration: 00:00:21.48, bitrate: 768 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels, s16, 768 kb/s

          #5931: showwavespic, showspectrumpic innacurate length of resulting image   

Summary of the bug:showwavespic, showspectrumpic innacurate length of resulting image How to reproduce:

ffmpeg -i voice.wav -lavfi  showwavespic=scale=log:s=1000x180 waveform.png

In this example, the waveform is rendered correctly (end of waveform on image1.png). But if we increase the with to 10000, the end of waveform have an extra gap (end of waveform on image2.png).

ffmpeg -i voice.wav -lavfi  showwavespic=scale=log:s=10000x180 waveform.png

          #6500: Fails to playback a certain .WAV audio file   

Summary of the bug: ffplay fails to playback a certain .WAV audio file. VLC and other non-ffmpeg-based players can play it correctly; I'd assume the audio file is malformed in some way because it seems to be detected as a MP3 file. How to reproduce:

% ffplay -i FF_massfowl.wav
FF_massfowl.wav: Invalid data found when processing input0B f=0/0

          #357: Trying to transcode a vbk file to wav   

Hi Team, I was trying to transcode a vbk file to a wav file. But I encoutered an error log pasted below. Also I verified that the file is not corrupt by playing the same in CallPilot? Player and it plays fine.

C:\Users\himanshu.bhardwaj\Desktop\FFMpegTest>ffmpeg -y -i C:\Users\himanshu.bhardwaj\Desktop\FFMpegTest\v060102.vbk -acodec pcm_mulaw C:\Users\himanshu.bhardwaj\Desktop\FFMpegTest\output.wav ffmpeg version N-31548-g78accb8, Copyright (c) 2000-2011 the FFmpeg developers built on Jul 17 2011 22:41:27 with gcc 4.6.1 configuration: --enable-gpl --enable-version3 --enable-memalign-hack --enable- runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib

libavutil 51. 11. 0 / 51. 11. 0 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 6. 0 / 53. 6. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 27. 0 / 2. 27. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0

C:\Users\himanshu.bhardwaj\Desktop\FFMpegTest\v060102.vbk: Invalid data found when processing input


          #1158: support iXML riff chunk   

This sample file contains metadata in a bext chunk and iXML chunk.

http://www.gallery.co.uk/ixml/examples/usesEntireiXMLSpec.WAV

With the flac utility there is an option called --keep-foreign-metadata that allows these chunks to be copied to the flac output. When I use ffmpeg to transcode the wav to a flac file there is no means to keep the bext chunk or ixml chunk of source file.

ffmpeg -i usesEntireiXMLSpec.WAV usesEntireiXMLSpec.flac
ffmpeg version N-35765-gf9143d2 Copyright (c) 2000-2012 the FFmpeg developers
  built on Mar 31 2012 12:44:33 with llvm_gcc 4.2.1 (Based on Apple Inc. build 5658) (LLVM build 2335.15.00)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/HEAD --enable-shared --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-libfreetype --cc=/usr/bin/llvm-gcc --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libass --disable-ffplay
  libavutil      51. 44.100 / 51. 44.100
  libavcodec     54. 12.100 / 54. 12.100
  libavformat    54.  3.100 / 54.  3.100
  libavdevice    53.  4.100 / 53.  4.100
  libavfilter     2. 66.101 /  2. 66.101
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 10.100 /  0. 10.100
  libpostproc    52.  0.100 / 52.  0.100
Input #0, wav, from 'usesEntireiXMLSpec.WAV':
  Metadata:
    comment         : gSCENE=66a
                    : gTAKE=002
                    : gTAPE=007
                    : gNOTE=Circle
                    : gUBITS=00000000
                    :
    encoded_by      : Gallery Metacorder
    date            : 2005:08:04
    creation_time   : 16:55:54
    time_reference  : 4090608000
    coding_history  :
  Duration: 00:00:02.18, bitrate: 781 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels, s16, 768 kb/s
Output #0, flac, to 'usesEntireiXMLSpec.flac':
  Metadata:
    comment         : gSCENE=66a
                    : gTAKE=002
                    : gTAPE=007
                    : gNOTE=Circle
                    : gUBITS=00000000
                    :
    encoded_by      : Gallery Metacorder
    date            : 2005:08:04
    creation_time   : 16:55:54
    time_reference  : 4090608000
    coding_history  :
    encoder         : Lavf54.3.100
    Stream #0:0: Audio: flac, 48000 Hz, 1 channels, s16, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> flac)
Press [q] to stop, [?] for help
size=      42kB time=00:00:02.20 bitrate= 155.8kbits/s
video:0kB audio:34kB global headers:0kB muxing overhead 24.469009%

          #1325: Different results win7 vs. linux on pcm_dvd --> pcm_s24le   

Version : ffmpeg version N-40824-g31dfe20 Job : pcm_dvd --> pcm_s24le Args[0] : -i E:\VIDEO_TS\VTS_01_1.VOB test.wav Args[1] : -acodec pcm_s24le -i E:\VIDEO_TS\VTS_01_1.VOB test.wav Result[Win7] : pcm_s16le Result[Ubu] : pcm_s32le

Desc : Sourceformat (24Bit 96kHz) detected, but ffmpeg@windows switch to pcm_s16le for output. FFmpeg@ubuntu create the wav-output with 24Bit. Set the codes pcm_s24le explicit don't help:

Microsoft Windows [Version 6.1.7601]
Copyright (c) 2009 Microsoft Corporation. Alle Rechte vorbehalten.
Z:\Music\Ripp\DVD-A>ffmpeg -acodec pcm_s24le -i E:\VIDEO_TS\VTS_01_1.VOB test.wav
ffmpeg version N-40824-g31dfe20 Copyright (c) 2000-2012 the FFmpeg developers
  built on May 19 2012 00:49:25 with gcc 4.6.3
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enab
le-bzlib --enable-frei0r --enable-libass --enable-libcelt --enable-libopencore-amrnb --enable-libopencore-amrwb --enable
-libfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-librtmp --enable-libschro
edinger --enable-libspeex --enable-libtheora --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-
libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      51. 53.100 / 51. 53.100
  libavcodec     54. 21.101 / 54. 21.101
  libavformat    54.  5.100 / 54.  5.100
  libavdevice    53.  4.100 / 53.  4.100
  libavfilter     2. 74.101 /  2. 74.101
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 12.100 /  0. 12.100
  libpostproc    52.  0.100 / 52.  0.100
[mpeg @ 0000000001e6eee0] max_analyze_duration 5000000 reached at 5001044
Guessed Channel Layout for  Input Stream #0.1 : stereo
Input #0, mpeg, from 'E:\VIDEO_TS\VTS_01_1.VOB':
  Duration: 00:10:59.06, start: 0.262200, bitrate: 13030 kb/s
    Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 8:9 DAR 4:3], 7500 kb/s, 29.97 tbr, 90k tbn, 59.
94 tbc
    Stream #0:1[0xa0]: Audio: '''pcm_s24le, 96000 Hz, stereo, s32''', 4608 kb/s
[abuffer @ 000000000a0fbf20] format:s32 layout:stereo rate:96000
[aformat @ 000000000ec50d00] auto-inserting filter 'auto-inserted resampler 0' between the filter 'src' and the filter '
aformat'
[aresample @ 000000000dd40da0] r:96000Hz -> r:96000Hz
Output #0, wav, to 'test.wav':
  Metadata:
    encoder         : Lavf54.5.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 96000 Hz, stereo, s16, 3072 kb/s
Stream mapping:
  Stream #0:1 -> #0:0 ('''pcm_s24le -> pcm_s16le''')
Press [q] to stop, [?] for help
size=   85217kB time=00:03:47.24 bitrate=3072.0kbits/s
video:0kB audio:85217kB global headers:0kB muxing overhead 0.000078%
Received signal 2: terminating.

          #1864: support 0x350 TwoCC (Micronas speech ADPCM codec)   

it's "Micronas InterMetall? Speech MI-SC4 ADPCM", decodes with noise for example with "-acodec adpcm_ct"

http://wiki.multimedia.cx/index.php?title=Micronas_SC4

C:\>ffmpeg -i 32khz_stereo.wav
ffmpeg version N-45867-g3318d6b Copyright (c) 2000-2012 the FFmpeg developers
  built on Oct 23 2012 02:39:33 with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.
0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-w32threads --enable-memalign-hack --enable-runtime-cpudetect
--enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -
lwinmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snap
shots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/s
napshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3
--enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-
libvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-
libopencore-amrwb --enable-libmp3lame --enable-libvpx --disable-decoder=libvpx
  libavutil      51. 77.100 / 51. 77.100
  libavcodec     54. 68.100 / 54. 68.100
  libavformat    54. 33.100 / 54. 33.100
  libavdevice    54.  3.100 / 54.  3.100
  libavfilter     3. 20.104 /  3. 20.104
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 16.100 /  0. 16.100
  libpostproc    52.  1.100 / 52.  1.100
[wav @ 0x1edbe00] Could not find codec parameters for stream 0 (Audio: none (P[3
][0][0] / 0x0350), 32000 Hz, 2 channels, 264 kb/s): unknown codec
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[wav @ 0x1edbe00] Estimating duration from bitrate, this may be inaccurate
32khz_stereo.wav: could not find codec parameters

          #2837: Unsupported tak codec   

(bug 546) A user uploaded a tak sample (cut with dd) that cannot be decoded with current git head.

$ ffmpeg -i 01.tak out.wav
ffmpeg version N-55249-ga7bb12a Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug  6 2013 08:47:17 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 20.100 / 55. 20.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, tak, from '01.tak':
  Duration: 00:51:26.69, start: 0.000000, bitrate: 2 kb/s
    Stream #0:0: Audio: tak, 44100 Hz, stereo, s16p
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf55.13.102
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (tak -> pcm_s16le)
Press [q] to stop, [?] for help
[tak @ 0x2357cc0] unsupported codec: 0
[tak @ 0x2358140] unsupported codec: 0
[tak @ 0x23694e0] unsupported codec: 0
[tak @ 0x2369d40] unsupported codec: 0
[tak @ 0x2341540] unsupported codec: 0
[tak @ 0x2343200] unsupported codec: 0
[tak @ 0x2344ee0] unsupported codec: 0
[tak @ 0x2346bc0] unsupported codec: 0
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
[tak @ 0x23488a0] unsupported codec: 0
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
...
[tak @ 0x2346bc0] unsupported codec: 0
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
[tak @ 0x23488a0] unsupported codec: 0
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
[tak @ 0x2357cc0] unsupported codec: 0
size=       0kB time=00:00:00.00 bitrate=N/A
video:0kB audio:0kB subtitle:0 global headers:0kB muxing overhead inf%
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)

          #3718: ffmpeg does not correctly read input text file.   

Summary of the bug: How to reproduce:

> ffmpeg -f concat -i t
mp.txt -c copy output.wav
ffmpeg version N-60592-gfd982f2 Copyright (c) 2000-2014 the FFmpeg developers
  built on Feb 13 2014 22:01:02 with gcc 4.8.2 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfi
g --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopenco
re-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li
bvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 63.101 / 52. 63.101
  libavcodec     55. 52.101 / 55. 52.101
  libavformat    55. 32.101 / 55. 32.101
  libavdevice    55.  9.100 / 55.  9.100
  libavfilter     4.  1.102 /  4.  1.102
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100
[concat @ 003b36e0] Line 1: unknown keyword 'file'
tmp.txt: Invalid data found when processing input

This is the Windows Zeranoe Build downloaded from here: http://ffmpeg.zeranoe.com/builds/win32/static/ffmpeg-20140612-git-3a1c895-win32-static.7z

The file is written from a self-made Visual Basic program using the method described here: http://msdn.microsoft.com/en-us/library/ms128035(v=vs.110).aspx. As you can see under the Remarks section, it uses the UTF-8 encoding.

Turns out, that method also writes 3 extra chars to the file, ef bb bf. It seems like this throws off FFMPEG and it gives the error above.


          #4140: wav: Demux cover art   

Summary of the bug:

ffplay does not show cover picture in it's gui window. Attached file contains picture described by id3v2.2 tags. File is wav. From libavformat/utils.c I found that id3 parsing happens only with mp3, aac, tta formats. If add wav condition parsing done well. Is it correct to add wav id3 parsing functionality?

How to reproduce: % ffplay test.wav

ffplay version N-68018-g8b8cb30 Copyright (c) 2003-2014 the FFmpeg developers

built on Nov 26 2014 16:52:25 with gcc 4.6 (Ubuntu/Linaro? 4.6.3-1ubuntu5) configuration: libavutil 54. 15.100 / 54. 15.100 libavcodec 56. 13.100 / 56. 13.100 libavformat 56. 15.100 / 56. 15.100 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100

Input #0, wav, from 'test.wav': 0KB vq= 0KB sq= 0B f=0/0

Metadata:

title : The Wailing Wall artist : Inner Sanctum album : CBS, Tue 11-06-45 date : 1945

Duration: 00:54:50.70, bitrate: 12 kb/s

Stream #0:0: Audio: mp3 (U[0][0][0] / 0x0055), 22050 Hz, mono, s16p, 24 kb/s

3.29 M-A: 0.000 fd= 0 aq= 1KB vq= 0KB sq= 0B f=0/0

ffmpeg version: master (8b8cb30d11885b406ce0dd896b42992386d98594)

built on: Ubuntu 12.04 LTS 64bit


          #4799: audio codec conversion codec 0x0135 sipro lab kelvin   

I'm trying to convert codec 0x0135 sipro lab kelvin to mp3. I think ffmpeg currently doesn't support this conversion.

commandline - ffmpeg -i sipro_codec_wav.wav test.mp3 http://www.datafilehost.com/d/cef9a4d9 http://www.datafilehost.com/d/d87133da http://www.datafilehost.com/d/05a4dc9e - sample wav


          #5313: mpeg-4 als: add support for adaptive RLS-LMS prediction   
C:\>mp4alsRM23.exe -[z1/z2/z3] -MP4 [z1/z2/z3].wav

http://www.datafilehost.com/d/220f9dee

C:\>ffmpeg -i z1.mp4
ffmpeg version N-78758-g5156578 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
ble-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --
enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-lib
x265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-d
ecklink --enable-zlib
  libavutil      55. 19.100 / 55. 19.100
  libavcodec     57. 27.100 / 57. 27.100
  libavformat    57. 26.100 / 57. 26.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 37.100 /  6. 37.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[als @ 038ef940] Adaptive RLS-LMS prediction is not implemented. Update your FFm
peg version to the newest one from Git. If the problem still occurs, it means th
at your file has a feature which has not been implemented.
[mov,mp4,m4a,3gp,3g2,mj2 @ 038de520] Failed to open codec in av_find_stream_info
[als @ 038ef940] Adaptive RLS-LMS prediction is not implemented. Update your FFm
peg version to the newest one from Git. If the problem still occurs, it means th
at your file has a feature which has not been implemented.
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'z1.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    creation_time   : 2016-03-08 19:47:15
  Duration: 00:00:30.01, start: 0.000000, bitrate: 811 kb/s
    Stream #0:0(und): Audio: mp4als (mp4a / 0x6134706D), 44100 Hz, 2 channels, 8
10 kb/s (default)
    Metadata:
      creation_time   : 2016-03-08 19:47:15
At least one output file must be specified

          #5407: WavPack correction file (hybrid mode)   

As far as I can tell ffmpeg is currently not able to use WavePack? correction file when decoding a WavePack? track.

It would be great if this could be implemented.

Relevant excerpt from WavePack? user documentation:

If the -c option is specified (in addition to the -b option), then WavPack? will generate an additional file that will contain the information that was discarded to generate the lossy .wv file. This file will have the same name as the .wv file but will have the extension .wvc (the 'c' is for "correction"). When WvUnpack? is used to restore the .wav file, it will attempt to find the .wvc file in the same directory as the .wv file. If it is found then the decompression will be lossless and will be reported as such, otherwise lossy decompression will be reported (assuming that any information was actually discarded during the original compression).


          #5483: g721: support stereo modes   

decodes ok with winamp

decodes not so good with sox (probably buggy support)

https://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket1955/g721_2ch_44khz.au

C:\>ffmpeg -i C:\g721_2ch_44khz.au out.wav
ffmpeg version N-79632-g3ce1988 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-pthreads --enable-memalign-hack --enable-runtime-cpudetect --
enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -lw
inmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snapsh
ots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/sna
pshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3 --
enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-li
bvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-li
bopencore-amrwb --enable-libmp3lame --enable-libfreetype --enable-libvpx --disab
le-decoder=libvpx
  libavutil      55. 22.101 / 55. 22.101
  libavcodec     57. 38.100 / 57. 38.100
  libavformat    57. 34.103 / 57. 34.103
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 44.100 /  6. 44.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[g726le @ 0x28ad4a0] Decoding more than one channel is not implemented. Update y
our FFmpeg version to the newest one from Git. If the problem still occurs, it m
eans that your file has a feature which has not been implemented.
[g726le @ 0x28ad4a0] If you want to help, upload a sample of this file to ftp://
upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing list. (ffmpeg-d
evel@ffmpeg.org)
[au @ 0x28a4020] Failed to open codec in av_find_stream_info
[g726le @ 0x28ad4a0] Decoding more than one channel is not implemented. Update y
our FFmpeg version to the newest one from Git. If the problem still occurs, it m
eans that your file has a feature which has not been implemented.
[g726le @ 0x28ad4a0] If you want to help, upload a sample of this file to ftp://
upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing list. (ffmpeg-d
evel@ffmpeg.org)
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, au, from 'C:\g721_2ch_44khz.au':
  Duration: 00:00:30.01, start: 0.000000, bitrate: 352 kb/s
    Stream #0:0: Audio: adpcm_g726le ([23][0][0][0] / 0x0017), 44100 Hz, 2 chann
els, 352 kb/s
File 'out.wav' already exists. Overwrite ? [y/N] y
[abuffer @ 0x2c49700] Unable to parse option value "(null)" as sample format
    Last message repeated 1 times
[abuffer @ 0x2c49700] Error setting option sample_fmt to value (null).
[graph 0 input from stream 0:0 @ 0x2c49640] Error applying options to the filter
.
Error opening filters!

          #2786: last packet of TTA stream in Matroska isn't decoded correctly   

Using inside.tta from the FATE test suit as an example:

Testing MD5 of the decoded stream:

$ ./ffmpeg -i ../samples/lossless-audio/inside.tta -f md5 -
ffmpeg version N-54738-g04b9836 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 14 2013 17:57:33 with gcc 4.7.3 (GCC)
  configuration: --enable-nonfree --enable-gpl --enable-libfdk-aac --enable-libm
p3lame --enable-libvorbis --cross-prefix=x86_64-w64-mingw32- --target-os=mingw32
 --arch=x86_64 --cpu=amdfam10 --prefix=/mingw64
  libavutil      52. 39.100 / 52. 39.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 81.101 /  3. 81.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Input #0, tta, from '../samples/lossless-audio/inside.tta':
  Duration: 00:00:11.89, start: 0.000000, bitrate: 814 kb/s
    Stream #0:0: Audio: tta, 44100 Hz, stereo, s16
Output #0, md5, to 'pipe:':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (tta -> pcm_s16le)
Press [q] to stop, [?] for help
MD5=d0beb768d860b4776358077dd9fcb1e9
size=       0kB time=00:00:11.88 bitrate=   0.0kbits/s
video:0kB audio:2048kB subtitle:0 global headers:0kB muxing overhead -99.998236%

Remuxing into Matroska:

$ ./ffmpeg -i ../samples/lossless-audio/inside.tta -c:a copy inside.mka
ffmpeg version N-54738-g04b9836 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 14 2013 17:57:33 with gcc 4.7.3 (GCC)
  configuration: --enable-nonfree --enable-gpl --enable-libfdk-aac --enable-libm
p3lame --enable-libvorbis --cross-prefix=x86_64-w64-mingw32- --target-os=mingw32
 --arch=x86_64 --cpu=amdfam10 --prefix=/mingw64
[...]
Input #0, tta, from '../samples/lossless-audio/inside.tta':
  Duration: 00:00:11.89, start: 0.000000, bitrate: 814 kb/s
    Stream #0:0: Audio: tta, 44100 Hz, stereo, s16
Output #0, matroska, to 'inside.mka':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: tta, 44100 Hz, stereo
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=    1182kB time=00:00:12.53 bitrate= 772.4kbits/s
video:0kB audio:1182kB subtitle:0 global headers:0kB muxing overhead 0.053638%

Testing MD5 of the decoded stream, this time from the Matroska file:

$ ./ffmpeg -i inside.mka -f md5 -
ffmpeg version N-54738-g04b9836 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 14 2013 17:57:33 with gcc 4.7.3 (GCC)
  configuration: --enable-nonfree --enable-gpl --enable-libfdk-aac --enable-libm
p3lame --enable-libvorbis --cross-prefix=x86_64-w64-mingw32- --target-os=mingw32
 --arch=x86_64 --cpu=amdfam10 --prefix=/mingw64
[...]
Input #0, matroska,webm, from 'inside.mka':
  Metadata:
    ENCODER         : Lavf55.12.102
  Duration: 00:00:11.89, start: 0.000000, bitrate: 814 kb/s
    Stream #0:0: Audio: tta, 44100 Hz, stereo, s16 (default)
Output #0, md5, to 'pipe:':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s (default)
Stream mapping:
  Stream #0:0 -> #0:0 (tta -> pcm_s16le)
Press [q] to stop, [?] for help
MD5=eb96f9b80e0ce6bc5ddd841a8843620a
size=       0kB time=00:00:11.49 bitrate=   0.0kbits/s
video:0kB audio:1980kB subtitle:0 global headers:0kB muxing overhead -99.998175%

Remuxing inside.tta using mkvtoolnix, then testing MD5 of the decoded stream:

$ ./ffmpeg -i inside_mkvtoolnix.mka -f md5 -
ffmpeg version N-54738-g04b9836 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 14 2013 17:57:33 with gcc 4.7.3 (GCC)
  configuration: --enable-nonfree --enable-gpl --enable-libfdk-aac --enable-libm
p3lame --enable-libvorbis --cross-prefix=x86_64-w64-mingw32- --target-os=mingw32
 --arch=x86_64 --cpu=amdfam10 --prefix=/mingw64
[...]
Input #0, matroska,webm, from 'inside_mkvtoolnix.mka':
  Metadata:
    creation_time   : 2013-07-15 07:41:51
  Duration: 00:00:11.89, start: 0.000000, bitrate: 818 kb/s
    Stream #0:0: Audio: tta, 44100 Hz, stereo, s16 (default)
Output #0, md5, to 'pipe:':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s (default)
Stream mapping:
  Stream #0:0 -> #0:0 (tta -> pcm_s16le)
Press [q] to stop, [?] for help
MD5=eb96f9b80e0ce6bc5ddd841a8843620a
size=       0kB time=00:00:11.49 bitrate=   0.0kbits/s
video:0kB audio:1980kB subtitle:0 global headers:0kB muxing overhead -99.998175%

To make sure I tried the bit-compare plugin for foobar2000, and unlike with avformat, both inside.tta and any of the Matroska files are the same:

All tracks decoded fine, no differences found.
Comparing:
"D:\MinGW\msys\1.0\ffmpeg\samples\lossless-audio\inside.tta"
"D:\MinGW\msys\1.0\ffmpeg\build\inside.mka"
No differences in decoded data found.
All tracks decoded fine, no differences found.
Comparing:
"D:\MinGW\msys\1.0\ffmpeg\samples\lossless-audio\inside.tta"
"D:\MinGW\msys\1.0\ffmpeg\build\inside_mkvtoolnix.mka"
No differences in decoded data found.

After converting inside.tta into WAV pcm_s16le with "./ffmpeg -i ../samples/lossless-audio/inside.tta inside_tta.wav":

All tracks decoded fine, no differences found.
Comparing:
"D:\MinGW\msys\1.0\ffmpeg\samples\lossless-audio\inside.tta"
"D:\MinGW\msys\1.0\ffmpeg\build\inside_tta.wav"
No differences in decoded data found.

After converting inside.mka into WAV pcm_s16le with "./ffmpeg -i inside.mka inside_mka.wav":

Differences found in 1 out of 1 track pairs.
Comparing:
"D:\MinGW\msys\1.0\ffmpeg\samples\lossless-audio\inside.tta"
"D:\MinGW\msys\1.0\ffmpeg\build\inside_mka.wav"
Length mismatch : 0:11.888367 vs 0:11.493878, 524277 vs 506880 samples

Looks like the Matroska demuxer is discarding data at the end of the stream or something, without giving an error or warning.


          #5238: gapless playback doesn't work (with at least Opus)   

Hi.

It seems that ffmpeg has quite some issues with respect to gapless playback at several formats. I'll show it here for Opus but it probably doesn't work for at least AAC either.

The following is basically from https://github.com/mpv-player/mpv/issues/2823, which AFAIU uses ffmpeg for opus playback:


The approach is basically: take and two tracks that play gapless (or take a single track and split it at a position with audio into two WAVs), encode them with opusenc, play it back with mpv... and one will hear a really very tiny gap (which sounds therefore more like a click) between the two.

Encoding settings I've used were: opusenc --bitrate 128 --vbr in.wav out.opus. Further, I decoded the opus files again with opusdec, opened the 6 files in audactiy (which AFAICS also uses ffmpeg to import Opus, just as mpv) and concatenated the corresponding files:

1st track: original WAV 2nd track: ffmpeg imported Opus 3rd track: WAV, gained by decoding the Opus files with opusdec

The beginning looks like: screenshot from 2016-02-14 23-26-40 The approach is basically: take and two tracks that play gapless (or take a single track and split it at a position with audio into two WAVs), encode them with opusenc, play it back with mpv... and one will hear a really very tiny gap (which sounds therefore more like a click) between the two.

Encoding settings I've used were: opusenc --bitrate 128 --vbr in.wav out.opus. Further, I decoded the opus files again with opusdec, opened the 6 files in audactiy (which AFAICS also uses ffmpeg to import Opus, just as mpv) and concatenated the corresponding files:

1st track: original WAV 2nd track: ffmpeg imported Opus 3rd track: WAV, gained by decoding the Opus files with opusdec

The beginning looks like beginning.png.

The join section looks like intersection.png

And the end like end.png

One can quite clearly see that 1st and 3rd track are pretty fine... but the directly read Opus is not only shifted, but also badly broken at the intersection.


I have basically the same problem with AAC, more detailed described here: https://github.com/mpv-player/mpv/issues/2284 and https://github.com/mstorsjo/fdk-aac/issues/35 https://github.com/nu774/fdkaac/issues/19

Interestingly, mpv does perfectly playback MP3s encoded gaplessly with lame (which confusingly does mean NOT to use the --gapless* options, which are broken for a long time).

mplayer OTOH, fails to badly with playing back anything gapless (including WAVs)

Cheers, Chris.


          #4916: AAC decoder generates noise if compiled on OSX 10.10.3   

We've a custom audio player built upon libav. We discovered that if we compile libav on osx 10.10.3 without changing a single line of code, then any AAC file decoding (see below) generates noise (you can't hear the audio at all, just noise). However, if we compile ffmpeg on osx 10.9.5 and we move the .dylib to 10.10.3, then AAC decoding works perfectly.

FFmpeg version: 2.8 (we also tested 2.3.6 and they issue is there too)

How we compile the libraries:

./configure \

--enable-shared \ --disable-static \ --enable-gpl \ --enable-version3 \ --enable-nonfree \ --disable-runtime-cpudetect \ --disable-all \ --disable-doc \ --enable-avcodec \ --enable-avformat \ --enable-avutil \ --enable-swresample \ --disable-w32threads \ --disable-os2threads \ --disable-network \ --disable-dxva2 \ --disable-vaapi \ --disable-vda \ --disable-vdpau \ --enable-protocol="file" \ --enable-decoder="aac" \ --enable-decoder="cook" \ --enable-decoder="flac" \ --enable-decoder="mp3" \ --enable-decoder="mp3adu" \ --enable-decoder="mp3adufloat" \ --enable-decoder="mp3float" \ --enable-decoder="mp3on4" \ --enable-decoder="mp3on4float" \ --enable-decoder="pcm_alaw" \ --enable-decoder="pcm_bluray" \ --enable-decoder="pcm_dvd" \ --enable-decoder="pcm_f32be" \ --enable-decoder="pcm_f32le" \ --enable-decoder="pcm_f64be" \ --enable-decoder="pcm_f64le" \ --enable-decoder="pcm_lxf" \ --enable-decoder="pcm_mulaw" \ --enable-decoder="pcm_s16be" \ --enable-decoder="pcm_s16be_planar" \ --enable-decoder="pcm_s16le" \ --enable-decoder="pcm_s16le_planar" \ --enable-decoder="pcm_s24be" \ --enable-decoder="pcm_s24daud" \ --enable-decoder="pcm_s24le" \ --enable-decoder="pcm_s24le_planar" \ --enable-decoder="pcm_s32be" \ --enable-decoder="pcm_s32le" \ --enable-decoder="pcm_s32le_planar" \ --enable-decoder="pcm_s8" \ --enable-decoder="pcm_s8_planar" \ --enable-decoder="pcm_u16be" \ --enable-decoder="pcm_u16le" \ --enable-decoder="pcm_u24be" \ --enable-decoder="pcm_u24le" \ --enable-decoder="pcm_u32be" \ --enable-decoder="pcm_u32le" \ --enable-decoder="pcm_u8" \ --enable-decoder="pcm_zork" \ --enable-decoder="ra_144" \ --enable-decoder="ra_288" \ --enable-decoder="ralf" \ --enable-decoder="vorbis" \ --enable-decoder="wmav1" \ --enable-decoder="wmav2" \ --enable-decoder="wmavoice" \ --enable-parser="aac" \ --enable-parser="flac" \ --enable-parser="mpegaudio" \ --enable-parser="vorbis" \ --enable-demuxer="aac" \ --enable-demuxer="asf" \ --enable-demuxer="flac" \ --enable-demuxer="ogg" \ --enable-demuxer="mp3" \ --enable-demuxer="mov" \ --enable-demuxer="pcm_alaw" \ --enable-demuxer="pcm_f32be" \ --enable-demuxer="pcm_f32le" \ --enable-demuxer="pcm_f64be" \ --enable-demuxer="pcm_f64le" \ --enable-demuxer="pcm_mulaw" \ --enable-demuxer="pcm_s16be" \ --enable-demuxer="pcm_s16le" \ --enable-demuxer="pcm_s24be" \ --enable-demuxer="pcm_s24le" \ --enable-demuxer="pcm_s32be" \ --enable-demuxer="pcm_s32le" \ --enable-demuxer="pcm_s8" \ --enable-demuxer="pcm_u16be" \ --enable-demuxer="pcm_u16le" \ --enable-demuxer="pcm_u24be" \ --enable-demuxer="pcm_u24le" \ --enable-demuxer="pcm_u32be" \ --enable-demuxer="pcm_u32le" \ --enable-demuxer="rm" \ --enable-demuxer="wav" \ --disable-debug \ --enable-stripping

make -j4 && make install

AAC test files: All the files listed here generate noise: http://download.wavetlan.com/SVV/Media/HTTP/http-aac.htm

Feel free to ask me any further information required to debug it. Thanks!


          #3803: Calculate actual bitrate of first frame when reading cbr dts   

Summary of the bug:

When adding a WAV header to a dts stream, FFmpeg writes the value read from the dts "target bitrate" field, which almost-always is not the actual bitrate of the stream. DTS in a WAV container is accepted by VirtualDub?'s AVI muxer, and so far I have to hex-edit the WAVved DTS files created through FFmpeg.

How to reproduce:

1) Create a dts file from an Audio CD track;

dcaenc -i CD_Track_01.wav -o track_01.dts -b 441

2) Mux with FFmpeg;

ffmpeg -i track_01.dts -acodec copy track_01.WAV

The RIFF header will report "448kbps", instead of *441kbps*, which is the correct value.


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