[Libav-user] [SPAM] Re: Adding AMR frames to audio stream of video file
Adev Dev
androiddevmar11 at gmail.com
Thu Jul 9 15:44:09 CEST 2015
Thank you all for help. Buffering samples is the answer!
On 6 July 2015 at 10:41, Mihai Chindea <mihai.chindea at uti.eu.com> wrote:
>
>
>
>
> *From:* Libav-user [mailto:libav-user-bounces at 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 <onemda at gmail.com> wrote:
>
>
> Dana 4. 7. 2015. 13:06 osoba "Adev Dev" <androiddevmar11 at 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 <androiddevmar11 at 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 <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
> >>>>> >>
> >>>>> >
> >>>>> _______________________________________________
> >>>>> 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
>
>
>
> 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,
>
>
>
> _______________________________________________
> Libav-user mailing list
> Libav-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/libav-user
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://ffmpeg.org/pipermail/libav-user/attachments/20150709/05825317/attachment.html>
More information about the Libav-user
mailing list