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 <androiddevma...@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 <one...@gmail.com> wrote: > >> On 7/2/15, Adev Dev <androiddevma...@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 <one...@gmail.com> wrote: >> > >> >> On 7/2/15, adev dev <androiddevma...@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 <one...@gmail.com> wrote: >> >> > >> >> >> >> >> >> Dana 2. 7. 2015. 07:58 osoba "adev dev" <androiddevma...@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...@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 <androiddevma...@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@ffmpeg.org >> >> >> >>> http://ffmpeg.org/mailman/listinfo/libav-user >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> Libav-user mailing list >> >> >> >> Libav-user@ffmpeg.org >> >> >> >> http://ffmpeg.org/mailman/listinfo/libav-user >> >> >> >> >> >> >> > >> >> >> > >> >> >> > _______________________________________________ >> >> >> > Libav-user mailing list >> >> >> > Libav-user@ffmpeg.org >> >> >> > http://ffmpeg.org/mailman/listinfo/libav-user >> >> >> > >> >> >> >> >> >> _______________________________________________ >> >> >> Libav-user mailing list >> >> >> Libav-user@ffmpeg.org >> >> >> http://ffmpeg.org/mailman/listinfo/libav-user >> >> >> >> >> >> >> >> > >> >> _______________________________________________ >> >> Libav-user mailing list >> >> Libav-user@ffmpeg.org >> >> http://ffmpeg.org/mailman/listinfo/libav-user >> >> >> > >> _______________________________________________ >> Libav-user mailing list >> Libav-user@ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/libav-user >> > >
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