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. On 3 July 2015 at 13:03, Adev Dev <[email protected]> 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 <[email protected]> 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 <[email protected]> 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 <[email protected]> wrote: >>> >>>> On 7/2/15, Adev Dev <[email protected]> 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 <[email protected]> wrote: >>>> > >>>> >> On 7/2/15, adev dev <[email protected]> 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 <[email protected]> wrote: >>>> >> > >>>> >> >> >>>> >> >> Dana 2. 7. 2015. 07:58 osoba "adev dev" < >>>> [email protected]> >>>> >> >> 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 < >>>> >> >> [email protected]> 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 <[email protected]> >>>> 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 >>>> >> >> >>> [email protected] >>>> >> >> >>> http://ffmpeg.org/mailman/listinfo/libav-user >>>> >> >> >> >>>> >> >> >> >>>> >> >> >> >>>> >> >> >> _______________________________________________ >>>> >> >> >> Libav-user mailing list >>>> >> >> >> [email protected] >>>> >> >> >> http://ffmpeg.org/mailman/listinfo/libav-user >>>> >> >> >> >>>> >> >> > >>>> >> >> > >>>> >> >> > _______________________________________________ >>>> >> >> > Libav-user mailing list >>>> >> >> > [email protected] >>>> >> >> > http://ffmpeg.org/mailman/listinfo/libav-user >>>> >> >> > >>>> >> >> >>>> >> >> _______________________________________________ >>>> >> >> Libav-user mailing list >>>> >> >> [email protected] >>>> >> >> http://ffmpeg.org/mailman/listinfo/libav-user >>>> >> >> >>>> >> >> >>>> >> > >>>> >> _______________________________________________ >>>> >> Libav-user mailing list >>>> >> [email protected] >>>> >> http://ffmpeg.org/mailman/listinfo/libav-user >>>> >> >>>> > >>>> _______________________________________________ >>>> Libav-user mailing list >>>> [email protected] >>>> http://ffmpeg.org/mailman/listinfo/libav-user >>>> >>> >>> >> >
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