August 2013 - FFmpeg-trac - lists.ffmpeg.org

#2667(undetermined:new): can't decode some mp4/h264 streams
by FFmpeg 03 Feb '14

03 Feb '14

#2667: can't decode some mp4/h264 streams -------------------------------------+------------------------------------- Reporter: gjdfgh | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Consider: {{{ ffplay http://video.ted.com/talk/podcast/2013/None/AndrewMcAfee_2013-480p.mp4 }}} This video plays fine at first, but about 130 seconds in it starts spamming these messages: {{{ AVC: nal size 0 nan fd= 2 aq= 46KB vq= 0KB sq= 0B f=0/0 [h264 @ 0xb4a07f40] AVC: nal size 0 [h264 @ 0xb4a07f40] no frame! AVC: nal size 0 nan fd= 2 aq= 46KB vq= 0KB sq= 0B f=0/0 [h264 @ 0xb4ca5940] AVC: nal size 0 [h264 @ 0xb4ca5940] no frame! [h264 @ 0xb4a1f7e0] AVC: nal size 0 Last message repeated 1 times }}} This is with latest git (fa125c5e82fce4). A user reported that a 2 week old Libav actually caused failed with an assertion on this. No, I won't follow the cehoyos ritual, don't even ask. It's easily reproducible. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/2667> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#2710(avformat:new): HTTP redirect 307 of an specified HLS url is not stored after HTTP connexion
by FFmpeg 03 Feb '14

03 Feb '14

#2710: HTTP redirect 307 of an specified HLS url is not stored after HTTP connexion -------------------------------------+------------------------------------- Reporter: fripooy | Type: Status: new | enhancement Component: avformat | Priority: normal Keywords: HLS, HTTP | Version: 1.0.7 redirect 307 | Blocked By: Blocking: | Reproduced by developer: 1 Analyzed by developer: 1 | -------------------------------------+------------------------------------- If there is a HTTP redirect 307 on a specified url, the redirect is followed during the http connexion but is not stored. In my compagny, TV streaming is done through HLS and load balancing. For a channel, the stream URL is forward to a temporary redirect. Ex: wget http://our-server/CH1/CH1.m3u8 --2013-06-25 16:00:59-- http://our-server/CH1/CH1.m3u8 Connecting to our-server:80... connected. HTTP request sent, awaiting response... 307 Temporary Redirect Location: http://our- server:80/CH1/kac3trwi5qcqy5iiartgtbfizjqet2ii/0000150/GlobalHlsManifest.m3u8 [following] --2013-06-25 16:00:59-- http://our- server/CH1/kac3trwi5qcqy5iiartgtbfizjqet2ii/0000150/GlobalHlsManifest.m3u8 Connecting to our-server:80... connected. HTTP request sent, awaiting response... 200 OK In our case, ffplay does not play the m3u8 because the URL sent to the HLS stack is the same as give in parameter but not the real URL (redirected) and the result is 404. /usr/bin/ffplay http://our-server/CH1/CH1.m3u8 avplay version 0.8.5-4:0.8.5-0ubuntu0.12.04.1, Copyright (c) 2003-2012 the Libav developers built on Jan 24 2013 18:01:36 with gcc 4.6.3 [http @ 0x7f702c0043c0] HTTP error 404 Not Found http://our-server/CH1/CH1.m3u8: Input/output error I have fix this bug in the patch. Sidney. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/2710> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#2686(FFmpeg:new): Native AAC encoder collapses at high bitrates on some samples
by FFmpeg 01 Feb '14

01 Feb '14

#2686: Native AAC encoder collapses at high bitrates on some samples -------------------------------------+------------------------------------- Reporter: Kamedo2 | Type: defect Status: new | Priority: normal Component: FFmpeg | Version: 1.0.7 Keywords: native aac | Blocked By: encoder sound quality 256kbps | Reproduced by developer: 0 320kbps | Blocking: | Analyzed by developer: 0 | -------------------------------------+------------------------------------- '''Summary of the bug:''' FFmpeg native aac encoder outputs horrible sound around 256kbps or more on particular samples. It happens at '''higher''' bitrates. The quality degrades as I '''increase''' the bitrates, and become most degraded at 320-400kbps. '''How to reproduce:''' {{{ ffmpeg -i ffmpeg_aac320k_collapse.flac -vn -c:a aac -strict experimental -b:a 320k ffmpeg_aac320k_collapse.mp4 }}} I couldn't reproduce the results when I trimmed the most problematic sample down to 8 seconds, but by adding 10 seconds of silence before the sample, the bug could be reproduced. So I'm going to upload the sample with 10 seconds of silence attached. The native aac encoder was ok on many music clips at 320kbps, and only some clips exhibit noticeably bad quality aac files, to an extent I'd call it 'bug'. '''Console Output:''' {{{ ffmpeg version N-54096-ge41bf19 Copyright (c) 2000-2013 the FFmpeg developers built on Jun 19 2013 00:20:06 with gcc 4.8.1 (GCC) configuration: --enable-gpl --enable-version3 --enable-libmp3lame --enable-lib vorbis --enable-nonfree --enable-libfdk-aac --enable-libvo_aacenc --enable-libfa ac --extra-ldflags=-static --extra-cflags='-march=nocona -mfpmath=sse' --optflag s=-O2 libavutil 52. 37.101 / 52. 37.101 libavcodec 55. 16.100 / 55. 16.100 libavformat 55. 9.100 / 55. 9.100 libavdevice 55. 2.100 / 55. 2.100 libavfilter 3. 77.101 / 3. 77.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [flac @ 0003f160] max_analyze_duration 5000000 reached at 5015510 microseconds Input #0, flac, from '05-true_my_heart_2m50s.flac': Duration: 00:00:18.01, bitrate: 573 kb/s Stream #0:0: Audio: flac, 44100 Hz, stereo, s16 Output #0, mp4, to '05-true_my_heart_2m50s_320k.mp4': Metadata: encoder : Lavf55.9.100 Stream #0:0: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo, fltp, 32 0 kb/s Stream mapping: Stream #0:0 -> #0:0 (flac -> aac) Press [q] to stop, [?] for help size= 331kB time=00:00:18.01 bitrate= 150.4kbits/s video:0kB audio:327kB subtitle:0 global headers:0kB muxing overhead 1.151111% }}} -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/2686> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#2658(avformat:new): Written mp4 with h264/AAC unplayable (discrepancy with libav)
by FFmpeg 01 Feb '14

01 Feb '14

#2658: Written mp4 with h264/AAC unplayable (discrepancy with libav) ----------------------------------+---------------------------------- Reporter: Lastique | Type: defect Status: new | Priority: normal Component: avformat | Version: 1.2.1 Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+---------------------------------- In our application, we write mp4 files through libav 9.6 using its public API. In one particular case we mux h264 video and AAC audio into the file, so the code is roughly like this: {{{ AVFormatContext* m_av_format_ctx = ...; AVCodec* video_codec = NULL; AVCodec* audio_codec = NULL; CodecID codec_id = AV_CODEC_ID_H264; video_codec = avcodec_find_encoder(codec_id); m_av_format_ctx->oformat->video_codec = codec_id; m_av_format_ctx->video_codec_id = codec_id; AVStream* av_stream = avformat_new_stream(m_av_format_ctx, NULL); av_stream->id = 0; av_stream->r_frame_rate = av_d2q(m_video_info->fps, std::numeric_limits< unsigned short >::max()); av_stream->time_base.num = av_stream->r_frame_rate.den; av_stream->time_base.den = av_stream->r_frame_rate.num; av_video_codec_ctx = av_stream->codec; // av_stream->codec initialization skipped... av_video_codec_ctx->codec_type = AVMEDIA_TYPE_VIDEO; av_video_codec_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER; av_video_codec_ctx->time_base = { 1, 1000 }; codec_id = AV_CODEC_ID_AAC; audio_codec = avcodec_find_encoder(codec_id); m_av_format_ctx->oformat->audio_codec = codec_id; m_av_format_ctx->audio_codec_id = codec_id; AVStream* av_stream = avformat_new_stream(m_av_format_ctx, NULL); av_stream->id = 1; av_audio_codec_ctx = av_stream->codec; av_stream->time_base.num = 1; av_stream->time_base.den = m_audio_info->sample_rate; av_audio_codec_ctx->codec_type = AVMEDIA_TYPE_AUDIO; av_audio_codec_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER; av_audio_codec_ctx->time_base = { 1, 1000 }; }}} (Sorry, I cannot post the complete code) Our internal video and audio timestamps are in milliseconds, but for audio we convert them to the sample rate-based values by calling av_rescale_rnd(). Video timestamps are converted to frame number (since for fixed-fps media we must specify time_base as 1/fps). The files are written ok by libav 9.6 this way, but not by ffmpeg 1.2.1 (our code is unchanged). Files written by ffmpeg are not playable, video frame rate is extremely low (about 1 frame per several seconds). I have attached files written by libav and ffmpeg. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/2658> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#712(FFmpeg:new): m2v: wrong duration
by FFmpeg 31 Jan '14

31 Jan '14

#712: m2v: wrong duration -------------------------------------+------------------------------------- Reporter: cbsrobot | Type: defect Status: new | Priority: normal Component: FFmpeg | Version: git- Keywords: mpeg2video | master m2v | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- When converting any file to a m2v and checking the resulting file, the duration is not correctly calculated. mplayer shows also wrong duration. Quicktime and Mediainfo show the correct duration. example: ffmpeg -i fate/aletrek-rle.mov -target pal-dvd -f mpeg2video -an aletrek- rle.m2v [...] Duration: 00:00:03.33 ffprobe aletrek-rle.m2v [...] Duration: 00:00:00.76 mediainfo aletrek-rle.m2v [...] Duration : 3s 400ms mplayer -vo null -ao null -frames 0 -identify aletrek-rle.m2v [...] ID_LENGTH=0.77 -- Ticket URL: <http://ffmpeg.org/trac/ffmpeg/ticket/712> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#2628(undetermined:new): when fifo_size is set , memory leaks
by FFmpeg 30 Jan '14

30 Jan '14

#2628: when fifo_size is set , memory leaks -------------------------------------+------------------------------------- Reporter: baituxp | Type: defect Status: new | Priority: normal Component: | Version: 1.2.1 undetermined | Blocked By: Keywords: fifo_size, | Reproduced by developer: 0 memory leak | Blocking: | Analyzed by developer: 1 | -------------------------------------+------------------------------------- Summary of the bug: How to reproduce: {{{ % ./ffmpeg -i 'udp://239.9.9.9:10001?fifo_size=10000000&overrun_nonfatal=1' -b:v 4096k -s "720*576" -vcodec libx264 -r 25 -x264opts vbv-bufsize=163:bitrate=4096 :vbv-maxrate=4096:nal-hrd=cbr -f mpegts udp://10.9.211.4:10023 ffmpeg 1.1.1 linux 5.3 }}} I test this using other ffmpeg versions, it occurs every times. fifo_size is set to a value, such as 10000000, the main memory runs out 30M every minitues. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/2628> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#2911(undetermined:new): Seg Fault
by FFmpeg 30 Jan '14

30 Jan '14

#2911: Seg Fault -------------------------------------+------------------------------------- Reporter: skyride | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Version of FFMpeg: skyride@ns308725:~/ffmpeg_sources/ffmpeg$ ffmpeg --version ffmpeg version git-2013-08-28-129f506 Copyright (c) 2000-2013 the FFmpeg developers built on Aug 28 2013 17:22:50 with gcc 4.7 (Debian 4.7.2-5) configuration: --prefix=/home/skyride/ffmpeg_build --extra- cflags=-I/home/skyride/ffmpeg_build/include --extra- ldflags=-L/home/skyride/ffmpeg_build/lib --bindir=/home/skyride/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab libavutil 52. 42.100 / 52. 42.100 libavcodec 55. 29.100 / 55. 29.100 libavformat 55. 14.102 / 55. 14.102 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 82.102 / 3. 82.102 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Summary of the bug: A segmentation fault occurs when trying to open FFmpeg in listen mode for an RTMP stream, and initiating a connection. How to reproduce: {{{ ffmpeg -rtmp_listen 1 -re -i rtmp://skyride.org/app/asd -vcodec copy -acodec copy rtmp://live-fra.twitch.tv/app/MY_STREAM_KEY }}} The program then runs, and when I attempt to open a connection I receive the following message: "[rtmp @ 0x1b70ea0] Unable to read command string Segmentation fault" -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/2911> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#1087(avcodec:new): support decoding the CineForm codec
by FFmpeg 29 Jan '14

29 Jan '14

#1087: support decoding the CineForm codec -------------------------------------+------------------------------------- Reporter: dericed | Type: Status: new | enhancement Component: avcodec | Priority: normal Keywords: CFHD, | Version: git- CineForm | master Blocking: | Blocked By: Analyzed by developer: 0 | Reproduced by developer: 0 -------------------------------------+------------------------------------- I found that ffmpeg can't read CineForm CFHD files. There are some samples available here: http://samples.mplayerhq.hu/V-codecs/CFHD/ I may be able to provide more samples if needed. {{{ ffmpeg -y -i MT_BeartoothHighway_1min_Cineform.avi anything.mp4 ffmpeg version N-34835-g4a9f466 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 4 2012 08:53:39 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-libtheora --enable-libvorbis --enable-libvpx --enable-libxvid --enable-libopencore-amrnb --enable- libopencore-amrwb --enable-libass libavutil 51. 41.100 / 51. 41.100 libavcodec 54. 7.100 / 54. 7.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 63.100 / 2. 63.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [avi @ 0x7fb4c201ae00] decoding for stream 0 failed [avi @ 0x7fb4c201ae00] Could not find codec parameters (Video: none (CFHD / 0x44484643), 1280x720) Input #0, avi, from 'MT_BeartoothHighway_1min_Cineform.avi': Duration: 00:01:01.96, start: 0.000000, bitrate: 16472 kb/s Stream #0:0: Video: none (CFHD / 0x44484643), 1280x720, 29.97 fps, 29.97 tbr, 29.97 tbn, 29.97 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s Video pixel format is unknown, stream cannot be encoded }}} -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/1087> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#1636(undetermined:new): FRAPS Video Color Space
by FFmpeg 29 Jan '14

29 Jan '14

#1636: FRAPS Video Color Space -------------------------------------+------------------------------------- Reporter: DJX | Type: defect Status: new | Priority: normal Component: | Version: git- undetermined | master Keywords: FRAPS color | Blocked By: space dark white washed out | Reproduced by developer: 0 Blocking: | Analyzed by developer: 0 | -------------------------------------+------------------------------------- The color space that ffmpeg uses on FRAPS videos seems to be wrong or mishandled. First, running ffplay on a video (color space detected as yuvj420p), the output looks washed out (too white). Second, when transcoding a video to H264 (which keeps the color space as yuvj420p) the output looks too dark. Forcing the color space to yuv420p, the output looks fine. Third, when transcoding a video to MPEG2 (which changes the colorspace to yuv420p), the output looks fine. Here is how ffmpeg detects the original file: {{{ ffmpeg -i "C:\Fraps\Movies\cod2sp_s 2012-08-12 13-39-04-15.avi" ffmpeg version N-43418-g633b90c Copyright (c) 2000-2012 the FFmpeg developers built on Aug 9 2012 23:56:26 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 libavutil 51. 67.100 / 51. 67.100 libavcodec 54. 51.100 / 54. 51.100 libavformat 54. 22.104 / 54. 22.104 libavdevice 54. 2.100 / 54. 2.100 libavfilter 3. 7.100 / 3. 7.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [avi @ 000000000217aa20] non-interleaved AVI Guessed Channel Layout for Input Stream #0.1 : stereo Input #0, avi, from 'C:\Fraps\Movies\orig.avi': Duration: 00:07:26.85, start: 0.000000, bitrate: 530091 kb/s Stream #0:0: Video: fraps (FPS1 / 0x31535046), yuvj420p, 1920x1080, 60 tbr, 60 tbn, 60 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16 , 1536 kb/s At least one output file must be specified }}} Let me know what other info you need. Thanks. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/1636> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#1834(swresample:new): swr_convert() results in integer division by zero exception
by FFmpeg 28 Jan '14

28 Jan '14

#1834: swr_convert() results in integer division by zero exception --------------------------------------+------------------------------------ Reporter: mbradshaw | Owner: michael Type: defect | Status: new Priority: normal | Component: swresample Version: git-master | Keywords: Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 --------------------------------------+------------------------------------ '''Summary of the bug:''' When I call `swr_convert()` I get an integer division by zero exception at 0x707c86ee. The disassembly around 0x707c86ee, should it help, is: {{{ 707C86AB je 707C86E4 707C86AD mov edx,dword ptr [ebx+4E8h] 707C86B3 test edx,edx 707C86B5 jne 707C8743 707C86BB mov dword ptr [esp+0Ch],edi 707C86BF mov dword ptr [esp+8],ebp 707C86C3 mov dword ptr [esp+4],esi 707C86C7 mov dword ptr [esp],eax 707C86CA mov dword ptr [esp+2Ch],ecx 707C86CE call 707C2AF4 707C86D3 mov ecx,dword ptr [esp+2Ch] 707C86D7 mov eax,ecx 707C86D9 add esp,3Ch 707C86DC pop ebx 707C86DD pop esi 707C86DE pop edi 707C86DF pop ebp 707C86E0 ret 707C86E1 lea esi,[esi] 707C86E4 test edi,edi 707C86E6 js 707C873C 707C86E8 mov eax,3FFFFFFFh 707C86ED cdq 707C86EE idiv eax,dword ptr [ebx+1B4h] 707C86F4 cdq 707C86F5 idiv eax,dword ptr [ebx+1B0h] 707C86FB cmp edi,eax 707C86FD jg 707C873C 707C86FF cmp edi,dword ptr [ebx+1B8h] 707C8705 jle 707C8724 707C8707 lea eax,[ebx+12Ch] 707C870D mov edx,edi 707C870F mov dword ptr [esp+2Ch],ecx 707C8713 call 707C77C8 707C8718 test eax,eax 707C871A mov ecx,dword ptr [esp+2Ch] 707C871E jns 707C8724 707C8720 mov ecx,eax 707C8722 jmp 707C86D7 707C8724 mov dword ptr [esp+54h],edi 707C8728 mov dword ptr [esp+50h],ebp 707C872C mov edx,esi 707C872E mov eax,ebx 707C8730 add esp,3Ch 707C8733 pop ebx 707C8734 pop esi 707C8735 pop edi 707C8736 pop ebp 707C8737 jmp 707C80B4 707C873C mov ecx,0FFFFFFEAh }}} This happens with more than just the attached sample file, but I've attached a file just in case. The very first call to `swr_convert()` results in this division by zero exception. '''How to reproduce:''' (some lame sample code just to show when it's happening) {{{ #include <iostream> extern "C" { #include <libavcodec/avcodec.h> #include <libavformat/avformat.h> #include <libswresample/swresample.h> }; int main() { av_register_all(); AVFrame* frame = avcodec_alloc_frame(); if (!frame) { std::cout << "Error allocating the frame" << std::endl; return 1; } AVFormatContext* formatContext = NULL; if (avformat_open_input(&formatContext, "C:/Users/mbradshaw/Desktop/samples/TimeCode.mov", NULL, NULL) != 0) { av_free(frame); std::cout << "Error opening the file" << std::endl; return 1; } if (avformat_find_stream_info(formatContext, NULL) < 0) { av_free(frame); av_close_input_file(formatContext); std::cout << "Error finding the stream info" << std::endl; return 1; } AVStream* audioStream = NULL; for (unsigned int i = 0; i < formatContext->nb_streams; ++i) { if (formatContext->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) { audioStream = formatContext->streams[i]; break; } } if (audioStream == NULL) { av_free(frame); av_close_input_file(formatContext); std::cout << "Could not find any audio stream in the file" << std::endl; return 1; } AVCodecContext* codecContext = audioStream->codec; codecContext->codec = avcodec_find_decoder(codecContext->codec_id); if (codecContext->codec == NULL) { av_free(frame); av_close_input_file(formatContext); std::cout << "Couldn't find a proper decoder" << std::endl; return 1; } else if (avcodec_open2(codecContext, codecContext->codec, NULL) != 0) { av_free(frame); av_close_input_file(formatContext); std::cout << "Couldn't open the context with the decoder" << std::endl; return 1; } SwrContext* swrContext = NULL; swrContext = swr_alloc_set_opts(swrContext, codecContext->channel_layout, // out channel layout codecContext->sample_fmt, // out sample format codecContext->sample_rate, // out sample rate codecContext->channel_layout, // in channel layout codecContext->sample_fmt, // in sample format codecContext->sample_rate, // in sample rate 0, // log offset NULL); // log context if (!swrContext) { av_free(frame); avcodec_close(codecContext); av_close_input_file(formatContext); std::cout << "Couldn't allocate and set the resampling context" << std::endl; return 1; } int bufSize = av_samples_get_buffer_size(NULL, codecContext->channels, codecContext->sample_rate, codecContext->sample_fmt, 0); uint8_t* buf = new uint8_t[bufSize]; AVPacket packet; av_init_packet(&packet); int packetCount = 0; int decodedFrameCount = 0; while (av_read_frame(formatContext, &packet) == 0) { ++packetCount; if (packet.stream_index == audioStream->index) { int frameFinished = 0; avcodec_decode_audio4(codecContext, frame, &frameFinished, &packet); if (frameFinished) { const uint8_t** in = (const uint8_t**)frame->extended_data; uint8_t* out[SWR_CH_MAX] = {buf, NULL}; std::cout << "converted: " << swr_convert(swrContext, out, codecContext->sample_rate, in, frame->nb_samples) << std::endl; } } av_free_packet(&packet); } delete [] buf; swr_free(&swrContext); av_free(frame); avcodec_close(codecContext); av_close_input_file(formatContext); } }}} -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/1834> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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