[Libav-user] Adding AMR frames to audio stream of video file
Adev Dev
androiddevmar11 at gmail.com
Fri Jul 3 13:03:01 CEST 2015
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 <androiddevmar11 at 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 <androiddevmar11 at 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 <onemda at gmail.com> wrote:
>>
>>> 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
>>> >> >> >
>>> >> >>
>>> >> >> _______________________________________________
>>> >> >> 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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>>> http://ffmpeg.org/mailman/listinfo/libav-user
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>>
>>
>
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