[issue773] AAC/AC3/EAC3/DTS should support channel_layout field

Justin Ruggles justin.rugg...@gmail.com added the comment: I have a working patch for the AC-3 encoder. The decoder is next. I also have a question waiting on some sort of answer on ffmpeg-devel regarding the avcodec_guess_channel_layout() function. -- nosy: +jbr status: new - open

[issue940] 6 channel raw audio input results in 'invalid PCM packet error'

Justin Ruggles justin.rugg...@gmail.com added the comment: It looks like Baptiste's patch got an ok from Michael. Not sure why it was never applied... -- nosy: +jbr FFmpeg issue tracker iss...@roundup.ffmpeg.org https

[issue884] libspeex decoder errors on any file

Justin Ruggles justin.rugg...@gmail.com added the comment: I don't get any errors with the sample you provided. I am using libspeex1.2beta4 from Ubuntu. The output does sound a little odd, but it sounds the same when I use speexdec. So regarding the error messages... which version of libspeex

[issue884] libspeex decoder errors on any file

Justin Ruggles justin.rugg...@gmail.com added the comment: (adding myself to nosy list) FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/roundup/ffmpeg/issue884

[issue1153] ac3dec 5.1 material is 3x softer than 2.0 material

Justin Ruggles justin.rugg...@gmail.com added the comment: I assume you mean 5.1 downmixed to 2.0 is softer than its accompanying 2.0 track. This is normal behavior. It is mentioned in the A/52 spec in section 7.8.2 Downmixing Into Two Channels. All mixing coefficients must be scaled down

[issue1150] FFplay can't seek *.flac file

Justin Ruggles justin.rugg...@gmail.com added the comment: confirmed. I have tried to fix this before, with much frustration, but was not successful. I won't give up yet though. -- assignedto: - jbr nosy: +jbr status: new - open substatus: new - reproduced

[issue1153] ac3dec 5.1 material is 3x softer than 2.0 material

Justin Ruggles justin.rugg...@gmail.com added the comment: closing this issue. will only reopen if a convincing argument is given that the AC3 decoder is doing something wrong. -- status: new - closed _ FFmpeg issue tracker iss

[issue1153] ac3dec 5.1 material is 3x softer than 2.0 material

Justin Ruggles justin.rugg...@gmail.com added the comment: Could you check to see if the dialogue normalization changes between programs? FFmpeg does not apply any dialogue level attenuation. Arbitrarily scaling by 3 is completely wrong though. If the difference you're hearing could indeed

[issue1153] ac3dec 5.1 material is 3x softer than 2.0 material

Justin Ruggles justin.rugg...@gmail.com added the comment: As far as dialogue normalization, it should only reduce, not increase, the output level. So what I'm wondering is if your stereo samples have a significantly higher dialnorm value (and thus should have volume reduced) than your 5.1

[issue884] libspeex decoder errors on any file

Justin Ruggles justin.rugg...@gmail.com added the comment: I'm sorry, but I'm still unable to reproduce. Maybe someone else can. Check the md5sum of the decoded wav file to see if it matches mine. ./ffmpeg -i ~/Desktop/noah-2sec-1600.spx noah.wav FFmpeg version SVN-r19182, Copyright (c) 2000

[issue1026] flv-mp3 streamcopy fails

Justin Ruggles justin.rugg...@gmail.com added the comment: verified with r19266 change status -- substatus: open - reproduced _ FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/roundup/ffmpeg/issue1026 _

[issue1153] ac3dec 5.1 material is 3x softer than 2.0 material

Justin Ruggles justin.rugg...@gmail.com added the comment: IIRC, Michael has said before that it might be an acceptable solution for the decoder to export the source dialogue level. Then the output could be scaled at the user level if the user specifies a target dialogue level

[issue1301] support SWF codecs other than MP3

Justin Ruggles justin.rugg...@gmail.com added the comment: Do you have SWF samples with Speex? Full Speex support in FLV is not far off I hope, but I don't know how it is used in SWF. _ FFmpeg issue tracker iss...@roundup.ffmpeg.org https

[issue1301] support SWF codecs other than MP3

Justin Ruggles justin.rugg...@gmail.com added the comment: add myself to nosy list -- nosy: +jbr _ FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/roundup/ffmpeg/issue1301 _

[issue1325] Can't transcode from 5.1 to vorbis

Justin Ruggles justin.rugg...@gmail.com added the comment: It seems the libvorbis wrapper is written to only support mono or stereo, but it doesn't fail otherwise. It needs to be fixed to either support multi-channel input or fail when the input is not mono or stereo. Changing title since

[issue1384] TwinVQ segfault during ff_sine_window_init

Justin Ruggles justin.rugg...@gmail.com added the comment: Yes, your 2nd patch fixes the issue for me. Attaching it here for historical purposes. _ FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/roundup/ffmpeg

[issue786] Support for WMA Voice

Justin Ruggles justin.rugg...@gmail.com added the comment: this is a valid feature request. change status to open/open. -- status: new - open substatus: new - open FFmpeg issue tracker iss...@roundup.ffmpeg.org https

[issue1325] Can't transcode from 5.1 to vorbis

Justin Ruggles justin.rugg...@gmail.com added the comment: Patch attached to allow multichannel input to the libvorbis encoder. Assigning the issue to myself since there is no libvorbis.c maintainer. -- assignedto: - jbr topic: +avcodec

[issue1391] Alias Wavefront - misidentified as adpcm_ea_xas

Justin Ruggles justin.rugg...@gmail.com added the comment: We do not support Alias Wavefront files. With the addition of the low probe score warning message, I would consider this issue fixed. -- status: new - closed substatus: new - fixed

[issue1426] AC3 don't want to decode

Justin Ruggles justin.rugg...@gmail.com added the comment: Benjamin Larsson wrote: Benjamin Larsson ba...@ludd.ltu.se added the comment: Justin Ruggles wrote: Justin Ruggles justin.rugg...@gmail.com added the comment: The file is broken. The AC-3 spec very clearly states that there must

[issue773] AAC/AC3/EAC3/DTS should support channel_layout field

Justin Ruggles justin.rugg...@gmail.com added the comment: I think this is done now for the decoders listed here. FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/roundup/ffmpeg/issue773

[issue1426] AC3 don't want to decode

Justin Ruggles justin.rugg...@gmail.com added the comment: Applied the patch in r20110. -- status: open - closed substatus: analyzed - fixed _ FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/roundup/ffmpeg

[issue589] Incorrect channel routing when decoding multichannel audio

Justin Ruggles justin.rugg...@gmail.com added the comment: The decoders mentioned in this bug report have all been fixed. Closing the issue. -- status: open - closed substatus: reproduced - fixed FFmpeg issue tracker iss

[issue1455] MP3 from AIF puts noise to the end of the file

Justin Ruggles justin.rugg...@gmail.com added the comment: The issue is that the SSND chunk does not have to be the last chunk in the AIFF file, but the AIFF demuxer reads audio data until the end of the file. The attached patch determines the audio data size so that aiff_read_packet() does

[issue1455] AIFF demuxer puts noise to the end of the file

Justin Ruggles justin.rugg...@gmail.com added the comment: change title. update topic list. -- title: MP3 from AIF puts noise to the end of the file - AIFF demuxer puts noise to the end of the file topic: +avformat _ FFmpeg issue

[issue1455] MP3 from AIF puts noise to the end of the file

Justin Ruggles justin.rugg...@gmail.com added the comment: trying to replace the file with new file... -- nosy: -jbr _ FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/roundup/ffmpeg/issue1455

[issue1455] MP3 from AIF puts noise to the end of the file

Justin Ruggles justin.rugg...@gmail.com added the comment: Baptiste Coudurier wrote: Baptiste Coudurier baptiste.coudur...@gmail.com added the comment: On 10/11/09 1:02 PM, Justin Ruggles wrote: Justin Rugglesjustin.rugg...@gmail.com added the comment: trying to replace the file

[issue1455] MP3 from AIF puts noise to the end of the file

Justin Ruggles justin.rugg...@gmail.com added the comment: fixed in r20219. -- status: open - closed substatus: analyzed - fixed _ FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/roundup/ffmpeg/issue1455 _

[issue1480] Attempting to seek FLAC breaks avf

Justin Ruggles justin.rugg...@gmail.com added the comment: duplicate of Issue1150 though this issue focuses more on lavf rather than just ffplay. -- assignedto: - jbr nosy: +jbr status: new - open substatus: new - duplicate topic: +avformat

[issue1153] ac3dec 5.1 material is 3x softer than 2.0 material

Justin Ruggles justin.rugg...@gmail.com added the comment: I couldn't find the actual bulletin, but I did find the Dolby press release[1] that describes it, and it seems to be saying that for DVB, using RF Mode on the receiver will make the loudness of (E)AC-3 closer to that of MP2. The spec

[issue1290] aac encoder - weird behaviour regarding global_quality

Justin Ruggles justin.rugg...@gmail.com added the comment: I have noticed the same thing. The rate control for aacenc just seems strange. I can't quite figure it out. It should either be intuitive or it should be documented. Also, when bit_rate is high the encoder will hang or crash. I'll

[issue1505] Decode FLAC but get AV_NOPTS_VALUE in pts

Justin Ruggles justin.rugg...@gmail.com added the comment: This is the same issue as 1150. A FLAC parser should fix the pts and also allow generic seeking. -- assignedto: - jbr dependson: +FFplay can't seek *.flac file nosy: +jbr status: new - open substatus: new - open superseder

[issue1511] aac encoding limited to 152kBits

Justin Ruggles justin.rugg...@gmail.com added the comment: It is libfaac. Try the faac commandline encoder and see what you get. faac -b 192 test.wav -o test.aac Freeware Advanced Audio Coder FAAC 1.26.1 (Aug 16 2008) UNSTABLE Average bitrate: 152 kbps Quantization quality: 100 Bandwidth

[issue850] ffmpeg should parse the audio stream as well as the container

Justin Ruggles justin.rugg...@gmail.com added the comment: Setting avctx-channels for each frame decode fixes the issue. That is what the AC-3 decoder does. In this case, ffmpeg still reports it as 5.0 due to the container, but it is changed to 5.1 by the decoder and decodes properly

[issue1595] FLAC decoder fails with large silent chunks.

Justin Ruggles justin.rugg...@gmail.com added the comment: I don't think anything is wrong here. You're getting weird audio output because you're using --sign=unsigned. In unsigned audio, 0 is full negative amplitude. flac -d foo.flac -o foo_flac.wav ffmpeg -i foo.flac foo_lavc.wav cmp

[issue1595] FLAC decoder fails with large silent chunks.

Justin Ruggles justin.rugg...@gmail.com added the comment: Deiz wrote: Deiz ffm...@pwnly.com added the comment: I didn't realize --unsigned was what was causing the output issue (As it turns out, that's dmix mixing the 0 dB sample with other output.) but that's not the underlying issue

[issue1595] FLAC decoder fails with large silent chunks.

Justin Ruggles justin.rugg...@gmail.com added the comment: needs more info. as Carl said, please supply complete output of the ffmpeg command that leads to this failure. -- substatus: - needs_more_info _ FFmpeg issue tracker iss

[issue1595] FLAC decoder fails with large silent chunks.

Justin Ruggles justin.rugg...@gmail.com added the comment: Well, it's hard to tell anything without a sample. Can you at least attach the foo.ana file output from flac -a foo.flac and the output of metaflac --list foo.flac? _ FFmpeg issue

[issue1595] FLAC decoder fails with large silent chunks.

Justin Ruggles justin.rugg...@gmail.com added the comment: That foo.ana file indicates that your source is all silence. So I recreated the file and could not reproduce any issues with ffmpeg or ffplay from SVN. Update to latest SVN and see if you can reproduce any issues with ffmpeg

[issue1595] FLAC decoder fails with large silent chunks.

Justin Ruggles justin.rugg...@gmail.com added the comment: there is already an open issue about flac seeking. closing this issue. -- status: open - closed substatus: open - invalid _ FFmpeg issue tracker iss...@roundup.ffmpeg.org

[issue1692] JetAudio infringes the GPL

Justin Ruggles justin.rugg...@gmail.com added the comment: Vitor wrote: New submission from Vitor vitor1...@gmail.com: Webiste at http://www.jetaudio.com, no source code offer. Binary at http://dn2.cowon.com/JetAudioInc/jetAudio/JAD8002_BASIC.exe . Moreover it looks like they hacked

[issue1709] ac3_decode_frame does not return err on error

Justin Ruggles justin.rugg...@gmail.com added the comment: This is intended behavior. The only time it actually returns an error is when there is a frame sync loss. Partial or invalid frames will trigger error concealment, which repeats the last known good block until the next valid frame

[issue1709] ac3_decode_frame does not return err on error

Justin Ruggles justin.rugg...@gmail.com added the comment: Yes, it would be welcome. Reopening as a feature request. -- priority: normal - minor status: closed - open substatus: invalid - open type: bug - feature_request _ FFmpeg

[issue1720] ffmpeg does not store metadata in converting to ogg, when using vorbis encoder

Justin Ruggles justin.rugg...@gmail.com added the comment: This seems to me to be a feature request to add support for writing of VorbisComment metadata. -- topic: +avformat _ FFmpeg issue tracker iss...@roundup.ffmpeg.org https

[issue1720] ffmpeg does not store metadata in converting to ogg, when using vorbis encoder

Justin Ruggles justin.rugg...@gmail.com added the comment: change type -- type: bug - feature_request _ FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/roundup/ffmpeg/issue1720 _

[issue246] support for CONFIG_MPEGAUDIO_HP in mpegaudioenc.c

Justin Ruggles justin.rugg...@gmail.com added the comment: USE_HIGHPRECISION went away in r16594 in favor of using CONFIG_MPEGAUDIO_HP directly. Part of the issue you mentioned was fixed, in that other files that include mpegaudio.h no longer need an extra definition, only mpegaudioenc.c

[issue491] g726 encoder generates invalid files

Justin Ruggles justin.rugg...@gmail.com added the comment: That is not exactly true. I have been able to generate files that play with the acm codec that are not 32kbps. ffmpeg -i test.wav -acodec g726 -ac 1 -ab 128k -ar 32000 test_g726.wav mplayer -ac g726 test_g726.wav But FFmpeg can

[issue668] ffmpeg cannot remux ac3 from mkv into mp4

Justin Ruggles justin.rugg...@gmail.com added the comment: The data needs to be run through the AC-3 parser to get the required codec parameters. Patch attached. -- topic: +avformat FFmpeg issue tracker iss...@roundup.ffmpeg.org

[issue1850] AC3 transcode (eg to WAV) - bad audio

Justin Ruggles justin.rugg...@gmail.com added the comment: So what happens is that when the stream changes to stereo, the decoder will start outputting stereo. The audio seems correct though. But since the output format is wav, it expects everything to be the same channel layout. So when

[issue1850] AC3 transcode (eg to WAV) - bad audio

Justin Ruggles justin.rugg...@gmail.com added the comment: So right now the application can know that the number of channels changes because avctx-channels changes. But I guess it wouldn't hurt to log a warning in the decoder when the number of channels changes and the decoder goes from

[issue1709] ac3_decode_frame does not return err on error

Justin Ruggles justin.rugg...@gmail.com added the comment: Joakim Plate wrote: Joakim Plate elu...@ecce.se added the comment: I think what he meant was it returns frame_size even if the supplied amount of data was less than that. Atleast going by the check he added in our codebase

[issue1921] crash in flac encoding with large frame_size

Justin Ruggles justin.rugg...@gmail.com added the comment: I patched this within the last couple weeks for the ALS encoder that Thilo and I are working on. I just haven't had a chance to submit it for review. FFmpeg issue tracker iss

[issue1921] crash in flac encoding with large frame_size

Justin Ruggles justin.rugg...@gmail.com added the comment: I'm not sure how to create a patch for an old git commit... so I'll post the URL here. http://github.com/justinruggles/FFmpeg-alsenc/commit/011c17464f89b188c01627b9c177af665d65b159

[issue1921] crash in flac encoding with large frame_size

Justin Ruggles justin.rugg...@gmail.com added the comment: Fixed in r23570. The FLAC encoder still won't encode large frames, but at least it won't crash anymore. -- status: open - closed substatus: reproduced - fixed FFmpeg issue

[issue2022] Dependent substream eac3 samples are not autodetected as E-AC-3

Justin Ruggles justin.rugg...@gmail.com added the comment: Carl Eugen Hoyos wrote: New submission from Carl Eugen Hoyos ceho...@rainbow.studorg.tuwien.ac.at: Both dependent substream eac3 samples on incoming (blu-ray_eac3 and Dependent_substream_decoding_sample.m2ts) are auto-detected

[issue2071] AC3 audio is not detected in MPEG TS stream

Justin Ruggles justin.rugg...@gmail.com added the comment: Baptiste Coudurier wrote: Baptiste Coudurier baptiste.coudur...@gmail.com added the comment: On 7/5/10 1:20 AM, Carl Eugen Hoyos wrote: Carl Eugen Hoyosceho...@rainbow.studorg.tuwien.ac.at added the comment: [mpegts @ 0x11bf470

[issue2022] Dependent substream eac3 samples are not autodetected as E-AC-3

Justin Ruggles justin.rugg...@gmail.com added the comment: fixed in r24103 -- status: open - closed substatus: open - fixed FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/issue2022

[issue2086] libavformat does not recoginize secondary DTS audio in BDAV

Justin Ruggles justin.rugg...@gmail.com added the comment: Carl Eugen Hoyos wrote: Carl Eugen Hoyos ceho...@rainbow.studorg.tuwien.ac.at added the comment: Secondary E-AC-3 stream works fine (transf.ts in samples). Where is transf.ts? It's not listed in allsamples.txt. -- title

[issue2086] libavformat does not recoginize secondary DTS audio in BDAV

Justin Ruggles justin.rugg...@gmail.com added the comment: Justin Ruggles wrote: Justin Ruggles justin.rugg...@gmail.com added the comment: Carl Eugen Hoyos wrote: Carl Eugen Hoyos ceho...@rainbow.studorg.tuwien.ac.at added the comment: Secondary E-AC-3 stream works fine (transf.ts

[issue810] FLAC files with large blocksize do not decode with ffmpeg

Justin Ruggles justin.rugg...@gmail.com added the comment: This should be fixable for FLAC once the FLAC parser makes it to SVN. The packet duration will be set by the demuxer. Then ffmpeg.c could be modified to allocate enough memory based on sample format, sample rate, channels

[issue2387] MPEG-4 ALS decoder MCC decoding bug (non-lossless output)

Justin Ruggles justin.rugg...@gmail.com added the comment: Sample moved to samples/A-codecs/lossless/als trying to keep it organized. FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/issue2387

[issue810] FLAC files with large blocksize do not decode with ffmpeg

Justin Ruggles justin.rugg...@gmail.com added the comment: Well, the solution I had in mind back in July isn't the best solution. This will be fixed after the audio decoders use get/release_buffer(). Which is something I have been working

[issue1505] Decode FLAC but get AV_NOPTS_VALUE in pts

Justin Ruggles justin.rugg...@gmail.com added the comment: The example code is wrong. I verified that ffmpeg.c now gets raw FLAC packets with correct sizes and pts since the FLAC parser was added. -- status: open - closed substatus: open - fixed

[issue2502] ffmpeg crashes for pcm audio with invalid sample_size

Justin Ruggles justin.rugg...@gmail.com added the comment: The heart of the issue seems to be that voc_get_packet() changes the codec_id when reading each packet based on input stream data. If a file is damaged, a random value will most likely make it CODEC_ID_NONE since there are only 8 valid

[issue2502] ffmpeg crashes for pcm audio with invalid sample_size

Justin Ruggles justin.rugg...@gmail.com added the comment: change status -- substatus: open - analyzed FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/issue2502

[issue2560] voc regression

Justin Ruggles justin.rugg...@gmail.com added the comment: On 01/17/2011 03:19 PM, ami_stuff wrote: New submission from ami_stuff ami_st...@o2.pl: the latest voc modification broken decoding of http://samples.mplayerhq.hu/voc/pcm_s16_2/nem.voc Author: cehoyos Date: Tue Jan 11

[issue2505] ffmpeg crashes on ts files with invalid headers

Justin Ruggles justin.rugg...@gmail.com added the comment: Fixed in 1360f07 -- status: open - closed substatus: reproduced - fixed FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/issue2505

[issue2560] voc regression

Justin Ruggles justin.rugg...@gmail.com added the comment: fixed in 8ca8e82 -- status: open - closed substatus: reproduced - fixed FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/issue2560

[issue2562] Common ID3 tags set via -metadata don't show in Windows Explorer

Justin Ruggles justin.rugg...@gmail.com added the comment: Changing this to a feature request. It might be useful to have a per-muxer AVOption for the mp3 muxer to write (or not write) different versions of ID3 tags. -- priority: important - wish status: new - open substatus: new - open

[issue670] Audio at end of clip is cutoff with libfaac encoder

Justin Ruggles justin.rugg...@gmail.com added the comment: file size samples original92160008 23039991 ffmpeg libfaac 92147756 23036928 ffmpeg aacenc 92160044 2304 faac92164140 23041024 What the FFmpeg native AAC encoder does seems to be pretty sensible

[issue1153] ac3dec 5.1 material is 3x softer than 2.0 material

Justin Ruggles justin.rugg...@gmail.com added the comment: changing to feature request. implementation of heavy compression is needed, as well as exporting of dialogue level. -- substatus: needs_changes - open topic: +avcodec type: patch - feature_request

[issue670] Audio at end of clip is cutoff with libfaac encoder

Justin Ruggles justin.rugg...@gmail.com added the comment: fixed in git 243f8241dbf4a451e1197661ccd387c519ae3349 -- status: open - closed substatus: reproduced - fixed FFmpeg issue tracker iss...@roundup.ffmpeg.org https

[issue2580] Complete your registration to FFmpeg issue tracker -- key Ps4rBm7HXyP9fz9ADVr0WmBjAE9kM6H9

Justin Ruggles justin.rugg...@gmail.com added the comment: not an issue -- status: new - closed substatus: new - invalid FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/issue2580

[issue2475] ffmpeg fails assertion on audio files with invalid sample rates

Justin Ruggles justin.rugg...@gmail.com added the comment: patch sent to ffmpeg-devel FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/issue2475

[issue2581] Sound fragments after seeking

New submission from Justin Ruggles justin.rugg...@gmail.com: copy/paste of attached SoundFragmentsAfterSeeking.txt: Hi, while trying to use FFmpeg libs as a developer I found the following: When decoding sound from some compressed audio formats (namely i tested some mp4 files with aac audio

[issue442] parameter for getting information about files

Justin Ruggles justin.rugg...@gmail.com added the comment: Should this issue be closed now that ffprobe is in FFmpeg? FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/issue442

[issue2483] Mix Multiple Audio Inputs

Justin Ruggles justin.rugg...@gmail.com added the comment: It's not clear what HoboZero wants to do by that commandline, but the mail referred to is regarding mixing multiple mono streams into a single multichannel stream. That seems like a valid feature request and probably a good candidate

[issue2556] Can't detect AC3 stream properly in WTV

Justin Ruggles justin.rugg...@gmail.com added the comment: Our wtv demuxer isn't handling something correctly. There seem to be 2 streams with the same stream id of 267. The first is AC-3, and it uses a stream2 GUID. The second is MP2, and it uses a stream GUID. The 2nd seems

[issue2556] Can't detect AC3 stream properly in WTV

Justin Ruggles justin.rugg...@gmail.com added the comment: It seems that the mp2 and ac3 are interleaved in the same stream, with each packet having an mp2 frame concatenated with an ac3 frame. Our demuxer skips the ac3 frames. The data for ac3 is using stream id 266, but that is not detected

[issue2556] Can't detect AC3 stream properly in WTV

Justin Ruggles justin.rugg...@gmail.com added the comment: I mean... they're not interleaved in the same stream. they have 2 separate stream id's (266 267) but the setup is not detecting the AC3 as a separate stream id. they're both detected as 267 and the mp2 info overrides the ac3 info

[issue2556] Can't detect AC3 stream properly in WTV

Justin Ruggles justin.rugg...@gmail.com added the comment: I have a fix. I'll send it to ffmpeg-devel after a bit more testing. Basically, stream2_guid chunks can initialize new streams as well. Existing demuxer only parses stream2_guid info if the stream already exists but no data has been

[issue2556] Can't detect AC3 stream properly in WTV

Justin Ruggles justin.rugg...@gmail.com added the comment: Out of the 3 samples streams (this one + 2 in mphq samples), all of them have extra streams using other stream ids that are skipped because they're initialized with stream2_guid, not stream_guid. But this is the only sample of the 3

[issue2556] Can't detect AC3 stream properly in WTV

Justin Ruggles justin.rugg...@gmail.com added the comment: Actually the stereo mp2 stream seems to be descriptive audio for the visually impaired, while the ac3 stream is the main program. FFmpeg issue tracker iss...@roundup.ffmpeg.org https

[issue2587] AC3 encoder has been changed to take float but accepts S16

Justin Ruggles justin.rugg...@gmail.com added the comment: Many of the audio encoders do not check sample format explicitly. Instead they have a list of supported sample formats. The application is supposed to check AVCodecContext-codec-sample_fmts. -- status: new - closed substatus

[issue2556] Can't detect AC3 stream properly in WTV

Justin Ruggles justin.rugg...@gmail.com added the comment: fixed by Peter Ross in e4f85b849913794395bb03dfc09546cd41b10882 -- status: open - closed substatus: analyzed - fixed FFmpeg issue tracker iss...@roundup.ffmpeg.org https

[issue2585] Error when execute ffmpeg

Justin Ruggles justin.rugg...@gmail.com added the comment: Please don't change to important until the issue is verified opened. -- priority: important - normal FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org

[issue2604] ffmpeg crashes when muxing video file with smaller audio file

Justin Ruggles justin.rugg...@gmail.com added the comment: missing necessary information. please see http://www.ffmpeg.org/bugreports.html -- substatus: new - needs_more_info FFmpeg issue tracker iss...@roundup.ffmpeg.org https

[issue2605] WARNING: Library configuration mismatch

Justin Ruggles justin.rugg...@gmail.com added the comment: I don't consider this a bug. It seems like a completely valid warning to me, and pretty obvious since it prints out the configure string for each library that doesn't match the configure string of the ffmpeg program. In the example

[issue2622] Recent change to BBC HD encoder causes un-playable TS file

Justin Ruggles justin.rugg...@gmail.com added the comment: Missing required information for a bug report. Please see http://www.ffmpeg.org/bugreports.html The video plays fine for me with current ffplay. -- substatus: new - needs_more_info

[issue2617] ffmpeg stalls and finally fails on handbrake-generated mp4

Justin Ruggles justin.rugg...@gmail.com added the comment: confirmed that this issue exists. it's not a demuxer/decoder problem since transcoding to ffv1/nut works just fine with this sample. probably an issue with ffmpeg, libx264 wrapper, or mp4 muxer. -- status: new - open substatus

[issue2576] AVCHD lite MTS file conversion loses first seconds

Justin Ruggles justin.rugg...@gmail.com added the comment: changing priority to normal. needs more info and/or reproducing to determine what the cause of the problem is. -- priority: important - normal status: new - open substatus: new - needs_more_info

[issue421] Support for WMA lossless

Justin Ruggles justin.rugg...@gmail.com added the comment: feature requests are not priority:important except maybe in some extreme circumstances. maybe we can add a topic tag for bounty? -- priority: important - minor FFmpeg issue

[issue2618] alac files created with ffmpeg do not play through iTunes sharing

Justin Ruggles justin.rugg...@gmail.com added the comment: Thanks for the bug report. Can you please provide a sample of the Apple-generated file before and after passing it through ffmpeg with -acodec copy? That will allow us to compare the mp4 structures to see where the problem might

[issue2635] symbol av_get_sample_fmt_string not defined

Justin Ruggles justin.rugg...@gmail.com added the comment: Changing priority back to normal until verified. -- priority: critical - normal topic: -git FFmpeg issue tracker iss...@roundup.ffmpeg.org https://roundup.ffmpeg.org/issue2635

[issue2570] configure incorrectly enables #define HAVE_AMD3DNOW 1 in config.h

Justin Ruggles justin.rugg...@gmail.com added the comment: unlike in libavcodec and libavutil, runtime cpu detection for libswscale is optional. maybe this should be changed to at least default to 'on' instead of 'off' to avoid unwanted surprises like this. -- status: new - open

[issue2639] frame copy of flac audio produces broken file

Justin Ruggles justin.rugg...@gmail.com added the comment: All I can confirm is that it does work fine with vlc and ffplay. Maybe there is something about the mkv files that mpc-hc (or whatever mkv demuxer is being used with it) doesn't like for some reason. -- status: new - open

[issue2639] frame copy of flac audio produces broken file

Justin Ruggles justin.rugg...@gmail.com added the comment: attached: mkvinfo of file generated by ffmpeg File 'ffmkv.txt' not attached - you can download it from https://roundup.ffmpeg.org/file1358. FFmpeg issue tracker iss...@roundup.ffmpeg.org

[issue2639] frame copy of flac audio produces broken file

Justin Ruggles justin.rugg...@gmail.com added the comment: attached: mkvinfo of file produced by mkvmerge File 'mkvmerge.txt' not attached - you can download it from https://roundup.ffmpeg.org/file1359. FFmpeg issue tracker iss

[issue2640] ff_fmt_convert_init_ppc not defined for powerpc only defined for altivec

Justin Ruggles justin.rugg...@gmail.com added the comment: working on a fix. thanks for the bug report. broken build is priority:important -- assignedto: - jbr nosy: +jbr priority: normal - important status: new - open substatus: new - analyzed topic: +avcodec, build system

[issue2647] AC-3 files with a 16-byte header

Justin Ruggles justin.rugg...@gmail.com added the comment: all except bytes 8 to 11 seem to be the same for every frame. 01 10 00 00 00 00 00 00 XX XX XX XX 00 14 22 06 bytes 8 to 11 could be some sort of timestamp. first two frames are 00 00 00 00 subsequent frames increment by FF C0 any

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