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 <[email protected]> wrote: > > Dana 4. 7. 2015. 13:06 osoba "Adev Dev" <[email protected]> > 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 <[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 > >>>> > >>>> > >>> > >> > > > > > > _______________________________________________ > > 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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