<div dir="ltr">I have just updated FFMPEG to latest version 2.7.1. Unfortunately problem still occurs. No progress at all. <div>In console I see now warnings:</div><div>"AVFrame.format is not set" and "AVFrame.width or height is not set".</div><div><br></div><div>Any ideas what is wrong? Thanks for help!<br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On 2 July 2015 at 12:55, Adev Dev <span dir="ltr"><<a href="mailto:androiddevmar11@gmail.com" target="_blank">androiddevmar11@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Sure, please download from GD:<div><br><div><a href="https://drive.google.com/folderview?id=0B7SEEPspZQx1fnZCZGlIVF9fbGVEYmh3UGpnMmxPNVFseUlOZ2xLa010Nk1fZVJLSXlRc2c&usp=sharing" target="_blank">https://drive.google.com/folderview?id=0B7SEEPspZQx1fnZCZGlIVF9fbGVEYmh3UGpnMmxPNVFseUlOZ2xLa010Nk1fZVJLSXlRc2c&usp=sharing</a></div><div><br></div><div>Please also check latest result on youtube: <a href="https://www.youtube.com/watch?v=w0BAyE14xLw" target="_blank">https://www.youtube.com/watch?v=w0BAyE14xLw</a></div><div><br></div><div>Thanks!</div></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On 2 July 2015 at 12:29, Paul B Mahol <span dir="ltr"><<a href="mailto:onemda@gmail.com" target="_blank">onemda@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span>On 7/2/15, Adev Dev <<a href="mailto:androiddevmar11@gmail.com" target="_blank">androiddevmar11@gmail.com</a>> wrote:<br>
> AMR file which is recorded in Android is correct. It can be played both on<br>
> Android and on MAC. After decoding it, reencoding to AAC and adding to<br>
> video file it is damaged. This video which I uploaded to YouTube has sound<br>
> encoded in AAC (reencoded from AMR file).<br>
><br>
> This is really strange because when I record audio file using AAC codec I<br>
> am doing the same steps and it is ok. First decode AAC frame from audio<br>
> file, then encode it and add to audio stream of video file. Maybe some<br>
> other params in codec, or audio stream is not set, or set to wrong value??<br>
><br>
<br>
</span>Could you upload and give a link to AMR file?<br>
<div><div><br>
><br>
><br>
><br>
><br>
><br>
> On 2 July 2015 at 12:12, Paul B Mahol <<a href="mailto:onemda@gmail.com" target="_blank">onemda@gmail.com</a>> wrote:<br>
><br>
>> On 7/2/15, adev dev <<a href="mailto:androiddevmar11@gmail.com" target="_blank">androiddevmar11@gmail.com</a>> wrote:<br>
>> > I was not clear enough. Sound is not bad quality. It is damaged. Please<br>
>> > have a look on video file which I uploaded to YouTube:<br>
>> ><br>
>> > <a href="https://www.youtube.com/watch?v=1UcGQwvtr9s" rel="noreferrer" target="_blank">https://www.youtube.com/watch?v=1UcGQwvtr9s</a><br>
>> ><br>
>> > Video length is 4 seconds. Adding this sound makes it longer to 17<br>
>> seconds.<br>
>> > Looks like some parameters are wrong. Yes, AMR is recorded in mono so<br>
>> > sample format converting is not needed. Thanks for help.<br>
>><br>
>> And sound is damaged when listening straight from recording?<br>
>><br>
>> ><br>
>> ><br>
>> > On 2 July 2015 at 10:14, Paul B Mahol <<a href="mailto:onemda@gmail.com" target="_blank">onemda@gmail.com</a>> wrote:<br>
>> ><br>
>> >><br>
>> >> Dana 2. 7. 2015. 07:58 osoba "adev dev" <<a href="mailto:androiddevmar11@gmail.com" target="_blank">androiddevmar11@gmail.com</a>><br>
>> >> napisala je:<br>
>> >><br>
>> >> ><br>
>> >> > Hi,<br>
>> >> > thanks for answer.<br>
>> >> ><br>
>> >> > I cannot increase sound bitrate. I am using Android MediaRecorder<br>
>> >> > and<br>
>> >> AMR codec for recording audio. AMR is needed because I am doing Chrome<br>
>> >> version where AAC codec is not working. This AMR codec at least in<br>
>> >> Android<br>
>> >> can only record with maximum bitrate 23600. It is not much but sound<br>
>> >> should<br>
>> >> be good. Now my result is that sound is totally crappy. There are<br>
>> strange<br>
>> >> pulses and if I record speech it is impossible to recognise words.<br>
>> >> ><br>
>> >> > I wonder what else could be the problem. When I am adding AAC files<br>
>> >> > to<br>
>> >> output video it is working correctly. Decoding AMR files and encoding<br>
>> >> them<br>
>> >> again to AAC is not working. For the first glance it looks that AMR<br>
>> >> decoding is not working correctly. Or the frame is in format (not<br>
>> planar)<br>
>> >> and this makes problem. What do you think?<br>
>> >> ><br>
>> >> > This is how I read frames and decode them:<br>
>> >> ><br>
>> >> > static void encodeSoundNext(JNIEnv * env, jobject this) {<br>
>> >> ><br>
>> >> > if (input_context == NULL)<br>
>> >> > return;<br>
>> >> ><br>
>> >> > int samples_size;<br>
>> >> ><br>
>> >> > frameRead = 0;<br>
>> >> > char index = 0;<br>
>> >> ><br>
>> >> > AVFrame *decoded_frame = NULL;<br>
>> >> ><br>
>> >> > int input_audio_stream_index = get_stream_index(input_context,<br>
>> >> AVMEDIA_TYPE_AUDIO);<br>
>> >> ><br>
>> >> > while (frameRead >= 0) {<br>
>> >> ><br>
>> >> > AVPacket in_packet;<br>
>> >> ><br>
>> >> > index++;<br>
>> >> ><br>
>> >> > frameRead = av_read_frame(input_context, &in_packet);<br>
>> >> > if (frameRead < 0) {<br>
>> >> > trackCompressionFinished = 1;<br>
>> >> > avformat_close_input(&input_context);<br>
>> >> ><br>
>> >> > } else {<br>
>> >> ><br>
>> >> > if (decoded_frame == NULL) {<br>
>> >> > if (!(decoded_frame = avcodec_alloc_frame())) {<br>
>> >> > LOGE("out of memory");<br>
>> >> > exit(1);<br>
>> >> > }<br>
>> >> > } else {<br>
>> >> > avcodec_get_frame_defaults(decoded_frame);<br>
>> >> > }<br>
>> >> > int got_frame_ptr;<br>
>> >> > samplesBytes = avcodec_decode_audio4(in_audio_st->codec,<br>
>> >> > decoded_frame, &got_frame_ptr, &in_packet);<br>
>> >> > if (samplesBytes < 0) {<br>
>> >> > LOGE("Error occurred during decoding.");<br>
>> >> > exit(1);<br>
>> >> > break;<br>
>> >> > }<br>
>> >> ><br>
>> >> > write_audio_frame(oc, audio_st, decoded_frame);<br>
>> >> > av_free_packet(&in_packet);<br>
>> >> ><br>
>> >> > }<br>
>> >> > }<br>
>> >> ><br>
>> >> > if (decoded_frame != NULL) {<br>
>> >> > av_free(decoded_frame);<br>
>> >> > decoded_frame = NULL;<br>
>> >> > }<br>
>> >> > }<br>
>> >> ><br>
>> >> ><br>
>> >> > This is how I am encoding sound to AAC:<br>
>> >> ><br>
>> >> ><br>
>> >> > static void write_audio_frame(AVFormatContext *oc, AVStream *st,<br>
>> >> > const AVFrame *frame_to_encode) {<br>
>> >> > AVCodecContext *c;<br>
>> >> > AVPacket pkt;<br>
>> >> > int got_packet_ptr = 0;<br>
>> >> ><br>
>> >> > av_init_packet(&pkt);<br>
>> >> > c = st->codec;<br>
>> >> > pkt.size = 0;<br>
>> >> > pkt.data = NULL;<br>
>> >> > int ret = avcodec_encode_audio2(c, &pkt, frame_to_encode,<br>
>> >> &got_packet_ptr);<br>
>> >> > if (ret < 0) {<br>
>> >> > exit(1);<br>
>> >> > }<br>
>> >> > if (got_packet_ptr == 1) {<br>
>> >> > if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE) {<br>
>> >> > pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base,<br>
>> >> > st->time_base);<br>
>> >> > }<br>
>> >> > pkt.flags |= AV_PKT_FLAG_KEY;<br>
>> >> > pkt.stream_index = st->index;<br>
>> >> > // write the compressed frame in the media file<br>
>> >> > if (av_interleaved_write_frame(oc, &pkt) != 0) {<br>
>> >> > LOGE("Error while writing audio frame.");<br>
>> >> > exit(1);<br>
>> >> > }<br>
>> >> > }<br>
>> >> > av_free_packet(&pkt);<br>
>> >> > }<br>
>> >> ><br>
>> >> ><br>
>> >> > Audio stream is added to video file in this way:<br>
>> >> ><br>
>> >> ><br>
>> >> > static AVStream *add_audio_stream(AVFormatContext *oc, enum<br>
>> >> > AVCodecID<br>
>> >> codec_id) {<br>
>> >> ><br>
>> >> > AVCodecContext *c;<br>
>> >> > AVStream *st;<br>
>> >> ><br>
>> >> > st = avformat_new_stream(oc, NULL);<br>
>> >> ><br>
>> >> > c = st->codec;<br>
>> >> > if (!st) {<br>
>> >> > LOGE("Could not alloc stream.");<br>
>> >> > return NULL;<br>
>> >> > }<br>
>> >> ><br>
>> >> > // AAC is expirimental in FFMPEG2.1<br>
>> >> > c->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;<br>
>> >> ><br>
>> >> > c->codec_id = codec_id;<br>
>> >> > c->codec_type = AVMEDIA_TYPE_AUDIO;<br>
>> >> > c->bit_rate = 23600; // bitrate of the compressed sound (must be<br>
>> higher<br>
>> >> for stereo)<br>
>> >> ><br>
>> >> > c->sample_rate = 16000;<br>
>> >> > c->channels = 1;<br>
>> >> > c->sample_fmt = AV_SAMPLE_FMT_FLT;<br>
>> >> ><br>
>> >> > if (oc->oformat->flags & AVFMT_GLOBALHEADER){<br>
>> >> > c->flags |= CODEC_FLAG_GLOBAL_HEADER;<br>
>> >> > }<br>
>> >> ><br>
>> >> > return st;<br>
>> >> > }<br>
>> >> ><br>
>> >> > What I noticed so far is that when I am decoding AAC files and<br>
>> encoding<br>
>> >> them again to audio stream in video files AAC frames has format<br>
>> >> AV_SAMPLE_FMT_FLTP. AMR frames are in AV_SAMPLE_FMT_FLT format. Do you<br>
>> >> think I have to convert some how from AV_SAMPLE_FMT_FLT to<br>
>> >> AV_SAMPLE_FMT_FLTP?? Thanks for all hints.<br>
>> >> ><br>
>> >><br>
>> >> For mono, single channel, conversion is not needed. If recording is of<br>
>> >> bad<br>
>> >> quality encoding you can only use some other amr encoder.<br>
>> >><br>
>> >> ><br>
>> >> ><br>
>> >> > On 1 July 2015 at 20:57, Talgorn Franc,ois-Xavier <<br>
>> >> <a href="mailto:fxtalgorn-at-yahoo.fr@ffmpeg.org" target="_blank">fxtalgorn-at-yahoo.fr@ffmpeg.org</a>> wrote:<br>
>> >> >><br>
>> >> >> Hi,<br>
>> >> >><br>
>> >> >> I don't know about AMR codec but bitrate definitely impacts on<br>
>> >> >> final<br>
>> >> quality.<br>
>> >> >> Try to increase bitrate value: I had same poor quality problems<br>
>> >> >> with<br>
>> >> MPEG4 encoding until I set the bitrate to width * height * 4.<br>
>> >> >> Keep in mind that poor quality might comes from a wide bunch of<br>
>> >> parameters used to initialize the codec.<br>
>> >> >> As for example, this is how I initialize an MPEG4 codec (A]), for<br>
>> >> clarity, in_ctx is initialized via the code in (B])<br>
>> >> >><br>
>> >> >> Concerning the delay issue: I also faced such a problem. I solved<br>
>> >> >> it<br>
>> >> using av_packet_rescale_ts() which relies on time_base, instead of<br>
>> >> setting<br>
>> >> timestamps myself manually.<br>
>> >> >><br>
>> >> >> I hope this comments will help put you on the road to success :-)<br>
>> >> >><br>
>> >> >> Good luck.<br>
>> >> >><br>
>> >> >> A]<br>
>> >> >> //codec found, now we param it<br>
>> >> >> o_codec_ctx->codec_id=AV_CODEC_ID_MPEG4;<br>
>> >> >> o_codec_ctx->bit_rate=in_ctx->picture_width *<br>
>> >> in_ctx->picture_height * 4;<br>
>> >> >><br>
>> >><br>
>> o_codec_ctx->width=in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->width;<br>
>> >> >><br>
>> >><br>
>> o_codec_ctx->height=in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->height;<br>
>> >> >> o_codec_ctx->time_base =<br>
>> >> in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->time_base;<br>
>> >> >> o_codec_ctx->ticks_per_frame =<br>
>> >><br>
>> in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->ticks_per_frame;<br>
>> >> >> o_codec_ctx->sample_aspect_ratio =<br>
>> >><br>
>> in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->sample_aspect_ratio;<br>
>> >> >><br>
>> >><br>
>> o_codec_ctx->gop_size=in_ctx->format_ctx->streams[in_ctx->video_stream_idx]->codec->gop_size;<br>
>> >> >> o_codec_ctx->pix_fmt=AV_PIX_FMT_YUV420P;<br>
>> >> >><br>
>> >> >><br>
>> >> >><br>
>> >> >> B]<br>
>> >> >> // register all formats and codecs<br>
>> >> >> av_register_all();<br>
>> >> >> avcodec_register_all();<br>
>> >> >><br>
>> >> >> // open input file, and allocate format context<br>
>> >> >> if (avformat_open_input(&in_fmt_ctx, filename, NULL, NULL) < 0)<br>
>> >> >> {<br>
>> >> >> fprintf(stderr, "Could not open source file %s\n",<br>
>> >> >> filename);<br>
>> >> >> exit(1);<br>
>> >> >> }<br>
>> >> >><br>
>> >> >> // retrieve stream information<br>
>> >> >> if (avformat_find_stream_info(in_fmt_ctx, NULL) < 0)<br>
>> >> >> {<br>
>> >> >> fprintf(stderr, "Could not find stream information\n");<br>
>> >> >> exit(1);<br>
>> >> >> }<br>
>> >> >><br>
>> >> >> if (open_codec_context(&video_stream_idx, in_fmt_ctx,<br>
>> >> AVMEDIA_TYPE_VIDEO, filename) >= 0)<br>
>> >> >> {<br>
>> >> >> video_stream = in_fmt_ctx->streams[video_stream_idx];<br>
>> >> >> video_dec_ctx = video_stream->codec;<br>
>> >> >> }<br>
>> >> >><br>
>> >> >> if (open_codec_context(&audio_stream_idx, in_fmt_ctx,<br>
>> >> AVMEDIA_TYPE_AUDIO, filename) >= 0) {<br>
>> >> >> audio_stream = in_fmt_ctx->streams[audio_stream_idx];<br>
>> >> >> audio_dec_ctx = audio_stream->codec;<br>
>> >> >> }<br>
>> >> >><br>
>> >> >> if (!video_stream) {<br>
>> >> >> fprintf(stderr, "Could not find video stream in the input,<br>
>> >> aborting\n");<br>
>> >> >> avformat_close_input(&in_fmt_ctx);<br>
>> >> >> exit(0);<br>
>> >> >> }<br>
>> >> >><br>
>> >> >> in_video_ctx->format_ctx=in_fmt_ctx;<br>
>> >> >> in_video_ctx->filename=filename;<br>
>> >> >> in_video_ctx->codec_name=(char *)<br>
>> >> in_fmt_ctx->streams[video_stream_idx]->codec->codec->long_name;<br>
>> >> >> in_video_ctx->video_stream_idx=video_stream_idx;<br>
>> >> >> in_video_ctx->audio_stream_idx=audio_stream_idx;<br>
>> >> >><br>
>> >><br>
>> in_video_ctx->picture_width=in_fmt_ctx->streams[video_stream_idx]->codec->width;<br>
>> >> >><br>
>> >><br>
>> in_video_ctx->picture_height=in_fmt_ctx->streams[video_stream_idx]->codec->height;<br>
>> >> >> in_video_ctx->nb_streams=in_fmt_ctx->nb_streams;<br>
>> >> >><br>
>> >> >><br>
>> >> >><br>
>> >> >><br>
>> >> >> Le 1 juil. 2015 `a 10:40, adev dev <<a href="mailto:androiddevmar11@gmail.com" target="_blank">androiddevmar11@gmail.com</a>> a<br>
>> ecrit<br>
>> >> >> :<br>
>> >> >><br>
>> >> >>> I am compressing movies from bitmaps and audio files. With AAC<br>
>> >> >>> files<br>
>> >> it is working correctly. But when I have AMR_WB files sound is<br>
>> corrupted.<br>
>> >> I<br>
>> >> can recognise correct words in video file but it is delayed and with<br>
>> very<br>
>> >> bad quality.<br>
>> >> >>><br>
>> >> >>> My AMR files are recorded with parameters:<br>
>> >> >>> - sampling rate: 16000,<br>
>> >> >>> - bitrate: 23000.<br>
>> >> >>><br>
>> >> >>> I am setting this parameters in audio stream which is added to<br>
>> video.<br>
>> >> Sample format is set to AV_SAMPLE_FMT_FLT. When using other formats<br>
>> >> app<br>
>> >> crashes with "Unsupported sample format".<br>
>> >> >>><br>
>> >> >>> What needs to be done to correctly add AMR stream to video file?<br>
>> >> >>> Do<br>
>> I<br>
>> >> have to reencode it to AAC and add as AAC audio stream?? Thank you for<br>
>> >> all<br>
>> >> hints.<br>
>> >> >>> _______________________________________________<br>
>> >> >>> Libav-user mailing list<br>
>> >> >>> <a href="mailto:Libav-user@ffmpeg.org" target="_blank">Libav-user@ffmpeg.org</a><br>
>> >> >>> <a href="http://ffmpeg.org/mailman/listinfo/libav-user" rel="noreferrer" target="_blank">http://ffmpeg.org/mailman/listinfo/libav-user</a><br>
>> >> >><br>
>> >> >><br>
>> >> >><br>
>> >> >> _______________________________________________<br>
>> >> >> Libav-user mailing list<br>
>> >> >> <a href="mailto:Libav-user@ffmpeg.org" target="_blank">Libav-user@ffmpeg.org</a><br>
>> >> >> <a href="http://ffmpeg.org/mailman/listinfo/libav-user" rel="noreferrer" target="_blank">http://ffmpeg.org/mailman/listinfo/libav-user</a><br>
>> >> >><br>
>> >> ><br>
>> >> ><br>
>> >> > _______________________________________________<br>
>> >> > Libav-user mailing list<br>
>> >> > <a href="mailto:Libav-user@ffmpeg.org" target="_blank">Libav-user@ffmpeg.org</a><br>
>> >> > <a href="http://ffmpeg.org/mailman/listinfo/libav-user" rel="noreferrer" target="_blank">http://ffmpeg.org/mailman/listinfo/libav-user</a><br>
>> >> ><br>
>> >><br>
>> >> _______________________________________________<br>
>> >> Libav-user mailing list<br>
>> >> <a href="mailto:Libav-user@ffmpeg.org" target="_blank">Libav-user@ffmpeg.org</a><br>
>> >> <a href="http://ffmpeg.org/mailman/listinfo/libav-user" rel="noreferrer" target="_blank">http://ffmpeg.org/mailman/listinfo/libav-user</a><br>
>> >><br>
>> >><br>
>> ><br>
>> _______________________________________________<br>
>> Libav-user mailing list<br>
>> <a href="mailto:Libav-user@ffmpeg.org" target="_blank">Libav-user@ffmpeg.org</a><br>
>> <a href="http://ffmpeg.org/mailman/listinfo/libav-user" rel="noreferrer" target="_blank">http://ffmpeg.org/mailman/listinfo/libav-user</a><br>
>><br>
><br>
_______________________________________________<br>
Libav-user mailing list<br>
<a href="mailto:Libav-user@ffmpeg.org" target="_blank">Libav-user@ffmpeg.org</a><br>
<a href="http://ffmpeg.org/mailman/listinfo/libav-user" rel="noreferrer" target="_blank">http://ffmpeg.org/mailman/listinfo/libav-user</a><br>
</div></div></blockquote></div><br></div>
</div></div></blockquote></div><br></div>