[Libav-user] Adding AMR frames to audio stream of video file
Paul B Mahol
onemda at gmail.com
Thu Jul 2 12:29:22 CEST 2015
On 7/2/15, Adev Dev <androiddevmar11 at 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 <onemda at gmail.com> wrote:
>
>> On 7/2/15, adev dev <androiddevmar11 at 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 <onemda at gmail.com> wrote:
>> >
>> >>
>> >> Dana 2. 7. 2015. 07:58 osoba "adev dev" <androiddevmar11 at 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.fr at 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 <androiddevmar11 at 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 at ffmpeg.org
>> >> >>> http://ffmpeg.org/mailman/listinfo/libav-user
>> >> >>
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> Libav-user mailing list
>> >> >> Libav-user at ffmpeg.org
>> >> >> http://ffmpeg.org/mailman/listinfo/libav-user
>> >> >>
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > Libav-user mailing list
>> >> > Libav-user at ffmpeg.org
>> >> > http://ffmpeg.org/mailman/listinfo/libav-user
>> >> >
>> >>
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>> >>
>> >>
>> >
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