From: Libav-user [mailto:libav-user-boun...@ffmpeg.org] On Behalf Of Adev Dev Sent: Monday, July 6, 2015 11:29 AM To: This list is about using libavcodec, libavformat, libavutil, libavdevice and libavfilter. Subject: [SPAM] Re: [Libav-user] Adding AMR frames to audio stream of video file
I added flush_encoder() function from transcode example but result is the same I would say. I am still doing something wrong. Sound is still about 4 times longer than expected (12 seconds) and voice signal is stretched in the range of these 12 seconds. I have a filling that the problem is something trivial. Maybe I have to merge these frames manually? Encoder still interprets these frames with 320 bytes of data as frames with 1024 bytes like in AAC. Thank you for help. On 4 July 2015 at 15:07, Paul B Mahol <one...@gmail.com<mailto:one...@gmail.com>> wrote: Dana 4. 7. 2015. 13:06 osoba "Adev Dev" <androiddevma...@gmail.com<mailto:androiddevma...@gmail.com>> napisala je: > > I found something that could be the reason of the problem. When I print > frame->nb_samples of AMR sound it is 320. During encoding warning is > generated "Trying to remove 704 more samples than there are in the queue". So > I assume that AAC encoder expects that frame has 1024 samples. > > Encoded AAC sound is about 4 times longer than it should be. When I skipped 3 > framers per 4 frames length is correct but sound is crappy still. > > AAC sound recorded with the same params (sampling rate: 16000, bitrate 23600) > has 1024 samples in frame. Looks that AMR sound has about 4 times more frames > but each frame has about 4 times less samples(320). > > I assume that AAC encoder should handle that situation if it is configured > correctly. Is there anybody who knows what is wrong in codec configuration??? > Thank you for help. > Try feeding encoder with nulls. Read documentation about codec_cap_delay. > > > > > On 3 July 2015 at 13:03, Adev Dev > <androiddevma...@gmail.com<mailto:androiddevma...@gmail.com>> wrote: >> >> Hi all! >> >> I prepared android project which makes encoding from AMR to AAC to better >> show the problem. It takes AMR file from resources and reencode it to >> "/storage/emulated/0/OutSound.aac". >> >> In MainActivity INPUT_AUDIO_NAME constant specifes input file. When set to >> amr.m4a strange problem described in this thread occurs. After changing to >> aac.m4a rencoding is working. >> >> I hope somebody is able to check this project and find the reason. I used >> older FFMPEG library because I do not know why project is not linking with >> latest version. Project is available under link: >> >> https://drive.google.com/folderview?id=0B7SEEPspZQx1fnZCZGlIVF9fbGVEYmh3UGpnMmxPNVFseUlOZ2xLa010Nk1fZVJLSXlRc2c&usp=sharing >> >> Thank you for help. >> >> >> On 2 July 2015 at 20:43, Adev Dev >> <androiddevma...@gmail.com<mailto:androiddevma...@gmail.com>> wrote: >>> >>> I have just updated FFMPEG to latest version 2.7.1. Unfortunately problem >>> still occurs. No progress at all. >>> In console I see now warnings: >>> "AVFrame.format is not set" and "AVFrame.width or height is not set". >>> >>> Any ideas what is wrong? Thanks for help! >>> >>> >>> >>> On 2 July 2015 at 12:55, Adev Dev >>> <androiddevma...@gmail.com<mailto:androiddevma...@gmail.com>> wrote: >>>> >>>> Sure, please download from GD: >>>> >>>> https://drive.google.com/folderview?id=0B7SEEPspZQx1fnZCZGlIVF9fbGVEYmh3UGpnMmxPNVFseUlOZ2xLa010Nk1fZVJLSXlRc2c&usp=sharing >>>> >>>> Please also check latest result on youtube: >>>> https://www.youtube.com/watch?v=w0BAyE14xLw >>>> >>>> Thanks! >>>> >>>> On 2 July 2015 at 12:29, Paul B Mahol >>>> <one...@gmail.com<mailto:one...@gmail.com>> wrote: >>>>> >>>>> On 7/2/15, Adev Dev >>>>> <androiddevma...@gmail.com<mailto:androiddevma...@gmail.com>> wrote: >>>>> > AMR file which is recorded in Android is correct. It can be played both >>>>> > on >>>>> > Android and on MAC. After decoding it, reencoding to AAC and adding to >>>>> > video file it is damaged. This video which I uploaded to YouTube has >>>>> > sound >>>>> > encoded in AAC (reencoded from AMR file). >>>>> > >>>>> > This is really strange because when I record audio file using AAC codec >>>>> > I >>>>> > am doing the same steps and it is ok. First decode AAC frame from audio >>>>> > file, then encode it and add to audio stream of video file. Maybe some >>>>> > other params in codec, or audio stream is not set, or set to wrong >>>>> > value?? >>>>> > >>>>> >>>>> Could you upload and give a link to AMR file? >>>>> >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > On 2 July 2015 at 12:12, Paul B Mahol >>>>> > <one...@gmail.com<mailto:one...@gmail.com>> wrote: >>>>> > >>>>> >> On 7/2/15, adev dev >>>>> >> <androiddevma...@gmail.com<mailto:androiddevma...@gmail.com>> wrote: >>>>> >> > I was not clear enough. Sound is not bad quality. It is damaged. >>>>> >> > Please >>>>> >> > have a look on video file which I uploaded to YouTube: >>>>> >> > >>>>> >> > https://www.youtube.com/watch?v=1UcGQwvtr9s >>>>> >> > >>>>> >> > Video length is 4 seconds. Adding this sound makes it longer to 17 >>>>> >> seconds. >>>>> >> > Looks like some parameters are wrong. Yes, AMR is recorded in mono so >>>>> >> > sample format converting is not needed. Thanks for help. >>>>> >> >>>>> >> And sound is damaged when listening straight from recording? >>>>> >> >>>>> >> > >>>>> >> > >>>>> >> > On 2 July 2015 at 10:14, Paul B Mahol >>>>> >> > <one...@gmail.com<mailto:one...@gmail.com>> wrote: >>>>> >> > >>>>> >> >> >>>>> >> >> Dana 2. 7. 2015. 07:58 osoba "adev dev" >>>>> >> >> <androiddevma...@gmail.com<mailto:androiddevma...@gmail.com>> >>>>> >> >> napisala je: >>>>> >> >> >>>>> >> >> > >>>>> >> >> > Hi, >>>>> >> >> > thanks for answer. >>>>> >> >> > >>>>> >> >> > I cannot increase sound bitrate. I am using Android MediaRecorder >>>>> >> >> > and >>>>> >> >> AMR codec for recording audio. AMR is needed because I am doing >>>>> >> >> Chrome >>>>> >> >> version where AAC codec is not working. This AMR codec at least in >>>>> >> >> Android >>>>> >> >> can only record with maximum bitrate 23600. It is not much but sound >>>>> >> >> should >>>>> >> >> be good. Now my result is that sound is totally crappy. There are >>>>> >> strange >>>>> >> >> pulses and if I record speech it is impossible to recognise words. >>>>> >> >> > >>>>> >> >> > I wonder what else could be the problem. When I am adding AAC >>>>> >> >> > files >>>>> >> >> > to >>>>> >> >> output video it is working correctly. Decoding AMR files and >>>>> >> >> encoding >>>>> >> >> them >>>>> >> >> again to AAC is not working. For the first glance it looks that AMR >>>>> >> >> decoding is not working correctly. Or the frame is in format (not >>>>> >> planar) >>>>> >> >> and this makes problem. What do you think? >>>>> >> >> > >>>>> >> >> > This is how I read frames and decode them: >>>>> >> >> > >>>>> >> >> > static void encodeSoundNext(JNIEnv * env, jobject this) { >>>>> >> >> > >>>>> >> >> > if (input_context == NULL) >>>>> >> >> > return; >>>>> >> >> > >>>>> >> >> > int samples_size; >>>>> >> >> > >>>>> >> >> > frameRead = 0; >>>>> >> >> > char index = 0; >>>>> >> >> > >>>>> >> >> > AVFrame *decoded_frame = NULL; >>>>> >> >> > >>>>> >> >> > int input_audio_stream_index = get_stream_index(input_context, >>>>> >> >> AVMEDIA_TYPE_AUDIO); >>>>> >> >> > >>>>> >> >> > while (frameRead >= 0) { >>>>> >> >> > >>>>> >> >> > AVPacket in_packet; >>>>> >> >> > >>>>> >> >> > index++; >>>>> >> >> > >>>>> >> >> > frameRead = av_read_frame(input_context, &in_packet); >>>>> >> >> > if (frameRead < 0) { >>>>> >> >> > trackCompressionFinished = 1; >>>>> >> >> > avformat_close_input(&input_context); >>>>> >> >> > >>>>> >> >> > } else { >>>>> >> >> > >>>>> >> >> > if (decoded_frame == NULL) { >>>>> >> >> > if (!(decoded_frame = avcodec_alloc_frame())) { >>>>> >> >> > LOGE("out of memory"); >>>>> >> >> > exit(1); >>>>> >> >> > } >>>>> >> >> > } else { >>>>> >> >> > avcodec_get_frame_defaults(decoded_frame); >>>>> >> >> > } >>>>> >> >> > int got_frame_ptr; >>>>> >> >> > samplesBytes = avcodec_decode_audio4(in_audio_st->codec, >>>>> >> >> > decoded_frame, &got_frame_ptr, &in_packet); >>>>> >> >> > if (samplesBytes < 0) { >>>>> >> >> > LOGE("Error occurred during decoding."); >>>>> >> >> > exit(1); >>>>> >> >> > break; >>>>> >> >> > } >>>>> >> >> > >>>>> >> >> > write_audio_frame(oc, audio_st, decoded_frame); >>>>> >> >> > av_free_packet(&in_packet); >>>>> >> >> > >>>>> >> >> > } >>>>> >> >> > } >>>>> >> >> > >>>>> >> >> > if (decoded_frame != NULL) { >>>>> >> >> > av_free(decoded_frame); >>>>> >> >> > decoded_frame = NULL; >>>>> >> >> > } >>>>> >> >> > } >>>>> >> >> > >>>>> >> >> > >>>>> >> >> > This is how I am encoding sound to AAC: >>>>> >> >> > >>>>> >> >> > >>>>> >> >> > static void write_audio_frame(AVFormatContext *oc, AVStream *st, >>>>> >> >> > const AVFrame *frame_to_encode) { >>>>> >> >> > AVCodecContext *c; >>>>> >> >> > AVPacket pkt; >>>>> >> >> > int got_packet_ptr = 0; >>>>> >> >> > >>>>> >> >> > av_init_packet(&pkt); >>>>> >> >> > c = st->codec; >>>>> >> >> > pkt.size = 0; >>>>> >> >> > pkt.data = NULL; >>>>> >> >> > int ret = avcodec_encode_audio2(c, &pkt, frame_to_encode, >>>>> >> >> &got_packet_ptr); >>>>> >> >> > if (ret < 0) { >>>>> >> >> > exit(1); >>>>> >> >> > } >>>>> >> >> > if (got_packet_ptr == 1) { >>>>> >> >> > if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE) { >>>>> >> >> > pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base, >>>>> >> >> > st->time_base); >>>>> >> >> > } >>>>> >> >> > pkt.flags |= AV_PKT_FLAG_KEY; >>>>> >> >> > pkt.stream_index = st->index; >>>>> >> >> > // write the compressed frame in the media file >>>>> >> >> > if (av_interleaved_write_frame(oc, &pkt) != 0) { >>>>> >> >> > LOGE("Error while writing audio frame."); >>>>> >> >> > exit(1); >>>>> >> >> > } >>>>> >> >> > } >>>>> >> >> > av_free_packet(&pkt); >>>>> >> >> > } >>>>> >> >> > >>>>> >> >> > >>>>> >> >> > Audio stream is added to video file in this way: >>>>> >> >> > >>>>> >> >> > >>>>> >> >> > static AVStream *add_audio_stream(AVFormatContext *oc, enum >>>>> >> >> > AVCodecID >>>>> >> >> codec_id) { >>>>> >> >> > >>>>> >> >> > AVCodecContext *c; >>>>> >> >> > AVStream *st; >>>>> >> >> > >>>>> >> >> > st = avformat_new_stream(oc, NULL); >>>>> >> >> > >>>>> >> >> > c = st->codec; >>>>> >> >> > if (!st) { >>>>> >> >> > LOGE("Could not alloc stream."); >>>>> >> >> > return NULL; >>>>> >> >> > } >>>>> >> >> > >>>>> >> >> > // AAC is expirimental in FFMPEG2.1 >>>>> >> >> > c->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL; >>>>> >> >> > >>>>> >> >> > c->codec_id = codec_id; >>>>> >> >> > c->codec_type = AVMEDIA_TYPE_AUDIO; >>>>> >> >> > c->bit_rate = 23600; // bitrate of the compressed sound (must be >>>>> >> higher >>>>> >> >> for stereo) >>>>> >> >> > >>>>> >> >> > c->sample_rate = 16000; >>>>> >> >> > c->channels = 1; >>>>> >> >> > c->sample_fmt = AV_SAMPLE_FMT_FLT; >>>>> >> >> > >>>>> >> >> > if (oc->oformat->flags & AVFMT_GLOBALHEADER){ >>>>> >> >> > c->flags |= CODEC_FLAG_GLOBAL_HEADER; >>>>> >> >> > } >>>>> >> >> > >>>>> >> >> > return st; >>>>> >> >> > } >>>>> >> >> > >>>>> >> >> > What I noticed so far is that when I am decoding AAC files and >>>>> >> encoding >>>>> >> >> them again to audio stream in video files AAC frames has format >>>>> >> >> AV_SAMPLE_FMT_FLTP. AMR frames are in AV_SAMPLE_FMT_FLT format. Do >>>>> >> >> you >>>>> >> >> think I have to convert some how from AV_SAMPLE_FMT_FLT to >>>>> >> >> AV_SAMPLE_FMT_FLTP?? Thanks for all hints. >>>>> >> >> > >>>>> >> >> >>>>> >> >> For mono, single channel, conversion is not needed. If recording is >>>>> >> >> of >>>>> >> >> bad >>>>> >> >> quality encoding you can only use some other amr encoder. >>>>> >> >> >>>>> >> >> > >>>>> >> >> > >>>>> >> >> > On 1 July 2015 at 20:57, Talgorn Franc,ois-Xavier < >>>>> >> >> fxtalgorn-at-yahoo...@ffmpeg.org<mailto:fxtalgorn-at-yahoo...@ffmpeg.org>> >>>>> >> >> wrote: >>>>> >> >> >> >>>>> >> >> >> Hi, >>>>> >> >> >> >>>>> >> >> >> I don't know about AMR codec but bitrate definitely impacts on >>>>> >> >> >> final >>>>> >> >> quality. >>>>> >> >> >> Try to increase bitrate value: I had same poor quality problems >>>>> >> >> >> with >>>>> >> >> MPEG4 encoding until I set the bitrate to width * height * 4. >>>>> >> >> >> Keep in mind that poor quality might comes from a wide bunch of >>>>> >> >> parameters used to initialize the codec. >>>>> >> >> >> As for example, this is how I initialize an MPEG4 codec (A]), for >>>>> >> >> clarity, in_ctx is initialized via the code in (B]) >>>>> >> >> >> >>>>> >> >> >> Concerning the delay issue: I also faced such a problem. I solved >>>>> >> >> >> it >>>>> >> >> using av_packet_rescale_ts() which relies on time_base, instead of >>>>> >> >> setting >>>>> >> >> timestamps myself manually. >>>>> >> >> >> >>>>> >> >> >> I hope this comments will help put you on the road to success :-) >>>>> >> >> >> >>>>> >> >> >> Good luck. >>>>> >> >> >> >>>>> >> >> >> A] >>>>> >> >> >> //codec found, now we param it >>>>> >> >> >> o_codec_ctx->codec_id=AV_CODEC_ID_MPEG4; >>>>> >> >> >> o_codec_ctx->bit_rate=in_ctx->picture_width * >>>>> >> >> in_ctx->picture_height * 4; >>>>> >> >> >> >>>>> >> >> >>>>> >> o_codec_ctx->width=in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->width; >>>>> >> >> >> >>>>> >> >> >>>>> >> o_codec_ctx->height=in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->height; >>>>> >> >> >> o_codec_ctx->time_base = >>>>> >> >> in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->time_base; >>>>> >> >> >> o_codec_ctx->ticks_per_frame = >>>>> >> >> >>>>> >> in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->ticks_per_frame; >>>>> >> >> >> o_codec_ctx->sample_aspect_ratio = >>>>> >> >> >>>>> >> in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->sample_aspect_ratio; >>>>> >> >> >> >>>>> >> >> >>>>> >> o_codec_ctx->gop_size=in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->gop_size; >>>>> >> >> >> o_codec_ctx->pix_fmt=AV_PIX_FMT_YUV420P; >>>>> >> >> >> >>>>> >> >> >> >>>>> >> >> >> >>>>> >> >> >> B] >>>>> >> >> >> // register all formats and codecs >>>>> >> >> >> av_register_all(); >>>>> >> >> >> avcodec_register_all(); >>>>> >> >> >> >>>>> >> >> >> // open input file, and allocate format context >>>>> >> >> >> if (avformat_open_input(&in_fmt_ctx, filename, NULL, NULL) < >>>>> >> >> >> 0) >>>>> >> >> >> { >>>>> >> >> >> fprintf(stderr, "Could not open source file %s\n", >>>>> >> >> >> filename); >>>>> >> >> >> exit(1); >>>>> >> >> >> } >>>>> >> >> >> >>>>> >> >> >> // retrieve stream information >>>>> >> >> >> if (avformat_find_stream_info(in_fmt_ctx, NULL) < 0) >>>>> >> >> >> { >>>>> >> >> >> fprintf(stderr, "Could not find stream information\n"); >>>>> >> >> >> exit(1); >>>>> >> >> >> } >>>>> >> >> >> >>>>> >> >> >> if (open_codec_context(&video_stream_idx, in_fmt_ctx, >>>>> >> >> AVMEDIA_TYPE_VIDEO, filename) >= 0) >>>>> >> >> >> { >>>>> >> >> >> video_stream = in_fmt_ctx->streams[video_stream_idx]; >>>>> >> >> >> video_dec_ctx = video_stream->codec; >>>>> >> >> >> } >>>>> >> >> >> >>>>> >> >> >> if (open_codec_context(&audio_stream_idx, in_fmt_ctx, >>>>> >> >> AVMEDIA_TYPE_AUDIO, filename) >= 0) { >>>>> >> >> >> audio_stream = in_fmt_ctx->streams[audio_stream_idx]; >>>>> >> >> >> audio_dec_ctx = audio_stream->codec; >>>>> >> >> >> } >>>>> >> >> >> >>>>> >> >> >> if (!video_stream) { >>>>> >> >> >> fprintf(stderr, "Could not find video stream in the >>>>> >> >> >> input, >>>>> >> >> aborting\n"); >>>>> >> >> >> avformat_close_input(&in_fmt_ctx); >>>>> >> >> >> exit(0); >>>>> >> >> >> } >>>>> >> >> >> >>>>> >> >> >> in_video_ctx->format_ctx=in_fmt_ctx; >>>>> >> >> >> in_video_ctx->filename=filename; >>>>> >> >> >> in_video_ctx->codec_name=(char *) >>>>> >> >> in_fmt_ctx->streams[video_stream_idx]->codec->codec->long_name; >>>>> >> >> >> in_video_ctx->video_stream_idx=video_stream_idx; >>>>> >> >> >> in_video_ctx->audio_stream_idx=audio_stream_idx; >>>>> >> >> >> >>>>> >> >> >>>>> >> in_video_ctx->picture_width=in_fmt_ctx->streams[video_stream_idx]->codec->width; >>>>> >> >> >> >>>>> >> >> >>>>> >> in_video_ctx->picture_height=in_fmt_ctx->streams[video_stream_idx]->codec->height; >>>>> >> >> >> in_video_ctx->nb_streams=in_fmt_ctx->nb_streams; >>>>> >> >> >> >>>>> >> >> >> >>>>> >> >> >> >>>>> >> >> >> >>>>> >> >> >> Le 1 juil. 2015 `a 10:40, adev dev >>>>> >> >> >> <androiddevma...@gmail.com<mailto:androiddevma...@gmail.com>> a >>>>> >> ecrit >>>>> >> >> >> : >>>>> >> >> >> >>>>> >> >> >>> I am compressing movies from bitmaps and audio files. With AAC >>>>> >> >> >>> files >>>>> >> >> it is working correctly. But when I have AMR_WB files sound is >>>>> >> corrupted. >>>>> >> >> I >>>>> >> >> can recognise correct words in video file but it is delayed and with >>>>> >> very >>>>> >> >> bad quality. >>>>> >> >> >>> >>>>> >> >> >>> My AMR files are recorded with parameters: >>>>> >> >> >>> - sampling rate: 16000, >>>>> >> >> >>> - bitrate: 23000. >>>>> >> >> >>> >>>>> >> >> >>> I am setting this parameters in audio stream which is added to >>>>> >> video. >>>>> >> >> Sample format is set to AV_SAMPLE_FMT_FLT. When using other formats >>>>> >> >> app >>>>> >> >> crashes with "Unsupported sample format". >>>>> >> >> >>> >>>>> >> >> >>> What needs to be done to correctly add AMR stream to video file? >>>>> >> >> >>> Do >>>>> >> I >>>>> >> >> have to reencode it to AAC and add as AAC audio stream?? Thank you >>>>> >> >> for >>>>> >> >> all >>>>> >> >> hints. >>>>> >> >> >>> _______________________________________________ >>>>> >> >> >>> Libav-user mailing list >>>>> >> >> >>> Libav-user@ffmpeg.org<mailto:Libav-user@ffmpeg.org> >>>>> >> >> >>> http://ffmpeg.org/mailman/listinfo/libav-user >>>>> >> >> >> >>>>> >> >> >> >>>>> >> >> >> >>>>> >> >> >> _______________________________________________ >>>>> >> >> >> Libav-user mailing list >>>>> >> >> >> Libav-user@ffmpeg.org<mailto:Libav-user@ffmpeg.org> >>>>> >> >> >> http://ffmpeg.org/mailman/listinfo/libav-user >>>>> >> >> >> >>>>> >> >> > >>>>> >> >> > >>>>> >> >> > _______________________________________________ >>>>> >> >> > Libav-user mailing list >>>>> >> >> > Libav-user@ffmpeg.org<mailto:Libav-user@ffmpeg.org> >>>>> >> >> > http://ffmpeg.org/mailman/listinfo/libav-user >>>>> >> >> > >>>>> >> >> >>>>> >> >> _______________________________________________ >>>>> >> >> Libav-user mailing list >>>>> >> >> Libav-user@ffmpeg.org<mailto:Libav-user@ffmpeg.org> >>>>> >> >> http://ffmpeg.org/mailman/listinfo/libav-user >>>>> >> >> >>>>> >> >> >>>>> >> > >>>>> >> _______________________________________________ >>>>> >> Libav-user mailing list >>>>> >> Libav-user@ffmpeg.org<mailto:Libav-user@ffmpeg.org> >>>>> >> http://ffmpeg.org/mailman/listinfo/libav-user >>>>> >> >>>>> > >>>>> _______________________________________________ >>>>> Libav-user mailing list >>>>> Libav-user@ffmpeg.org<mailto:Libav-user@ffmpeg.org> >>>>> http://ffmpeg.org/mailman/listinfo/libav-user >>>> >>>> >>> >> > > > _______________________________________________ > Libav-user mailing list > Libav-user@ffmpeg.org<mailto:Libav-user@ffmpeg.org> > http://ffmpeg.org/mailman/listinfo/libav-user > _______________________________________________ Libav-user mailing list Libav-user@ffmpeg.org<mailto:Libav-user@ffmpeg.org> http://ffmpeg.org/mailman/listinfo/libav-user If the number of samples the decoder outputs differs from number of samples the encoder expects ( specified in AVCodecContext->frame_size ) you are supposed to cache the samples up until you have at least “AVCodecContext->frame_size” samples, from experience the encoder will pad your input sample array up to it’s desired size, and that is the noise you are hearing in output. Have a look at AVAudioFifo Regards,
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