[Libav-user] Reencoding sound track in video file.

I am working on converting audio track of video files. In these files video is encoded with h263 codec and audio with AMR_WB. I have to convert audio track to be encoded with AAC coded. So the new video should be encoded with h263 and sound with AAC. My general approach is to copy video frame wi

Re: [Libav-user] Problem converting from AV_SAMPLE_FMT_S16 to AV_SAMPLE_FMT_FLTP

It is working after suggested changes!. Thank you. On 5 January 2016 at 22:21, dhenry wrote: > swr_convert() may buffer converted data, so esp. if you are converting > sample > rates, there may not be any data available after the first call. > Also, in your call to utility_init_output_frame(), y

[Libav-user] Problem converting from AV_SAMPLE_FMT_S16 to AV_SAMPLE_FMT_FLTP

My application creates AAC files. The input is feed with frames in AV_SAMPLE_FMT_S16 format. I am doing conversion to AV_SAMPLE_FMT_FLTP and encoding these frames using avcodec_encode_audio2 function. The problem is that function "avcodec_encode_audio2" crashes without error. When I feed input wit

Re: [Libav-user] [SPAM] Re: Adding AMR frames to audio stream of video file

Thank you all for help. Buffering samples is the answer! On 6 July 2015 at 10:41, Mihai Chindea wrote: > > > > > *From:* Libav-user [mailto:libav-user-boun...@ffmpeg.org] *On Behalf Of *Adev > Dev > *Sent:* Monday, July 6, 2015 11:29 AM > *To:* This list is about using

Re: [Libav-user] Adding AMR frames to audio stream of video file

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 wrote: > > Dana 4. 7. 2015. 13:06 osoba "Adev Dev" > napisala

Re: [Libav-user] Adding AMR frames to audio stream of video file

ames 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. On 3 July 2015 at 13:03, Adev Dev wrote: > ​​Hi all! > &

Re: [Libav-user] Adding AMR frames to audio stream of video file

google.com/folderview?id=0B7SEEPspZQx1fnZCZGlIVF9fbGVEYmh3UGpnMmxPNVFseUlOZ2xLa010Nk1fZVJLSXlRc2c&usp=sharing Thank you for help. On 2 July 2015 at 20:43, Adev Dev wrote: > I have just updated FFMPEG to latest version 2.7.1. Unfortunately problem > still occurs. No progress at all. &g

Re: [Libav-user] Adding AMR frames to audio stream of video file

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

Re: [Libav-user] Adding AMR frames to audio stream of video file

ahol wrote: > On 7/2/15, Adev Dev 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 > soun

Re: [Libav-user] Adding AMR frames to audio stream of video file

Mahol wrote: > On 7/2/15, adev dev 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 second

Re: [Libav-user] Adding AMR frames to audio stream of video file

recorded in mono so sample format converting is not needed. Thanks for help. On 2 July 2015 at 10:14, Paul B Mahol wrote: > > Dana 2. 7. 2015. 07:58 osoba "adev dev" > napisala je: > > > > > Hi, > > thanks for answer. > > > > I cannot incr

Re: [Libav-user] Adding AMR frames to audio stream of video file

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->co

Re: [Libav-user] Adding AMR frames to audio stream of video file

sample format. Do I have to resample AMR samples from AV_SAMPLE_FMT_FLT to AV_SAMPLE_FMT_FLTP before encoding to AAC? On 1 July 2015 at 10:40, adev dev wrote: > I am compressing movies from bitmaps and audio files. With AAC files it is > working correctly. But when I have AMR_WB files so

[Libav-user] Adding AMR frames to audio stream of video file

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,

Re: [Libav-user] Once again about seeking in video file

ee_packet(&packet); I would appreciate If somebody could look at it and check if something can be improved. Thanks! On 28 November 2014 at 22:25, adev dev wrote: > I saw many threads about seeking in video files with FFMPEG. But still it > is not clear for me how to seek to specif

[Libav-user] Once again about seeking in video file

I saw many threads about seeking in video files with FFMPEG. But still it is not clear for me how to seek to specific location (in milliseconds) with good accuracy. I know that to seek in video file it is needed to seek than decode frames and check where we are, than seek again decode frames etc. t

[Libav-user] Specific usage of "Volume" filter

I am trying to change volume in the specific way in AAC file. So far as I know "volume" filter allows to change volume on the whole AAC file. What I need is to change volume only on part of file. To be specific, let say I have AAC file which is one minute long: 1. I want to set volume 0.5 in the fi

Re: [Libav-user] Problem with compressing AAC files with sampling rate 16000

I made mistake. FFMPEG2.4 + aac_quality_improvment.patch do not solve the issue with sampling rate 16000. I will wait for patch update. On 20 September 2014 15:10, adev dev wrote: > I do not know why some warnings are treated as errors. I added flag > -disable-werror but still the same.

Re: [Libav-user] Problem with compressing AAC files with sampling rate 16000

20 September 2014 14:14, Claudio Freire wrote: > On Sat, Sep 20, 2014 at 6:33 AM, adev dev > wrote: > > This is information about the file which cannot be compressed: > > Metadata: > > major_brand : isom > > minor_version : 0 >

Re: [Libav-user] Problem with compressing AAC files with sampling rate 16000

is patch problem?? BR, Marcin On 19 September 2014 18:51, Claudio Freire wrote: > On Fri, Sep 19, 2014 at 5:28 AM, adev dev > wrote: > > I am not using command line. It is done in code in Android project. Some > > devices cannot record sound with 192000 and 44100 probably du

Re: [Libav-user] Problem with compressing AAC files with sampling rate 16000

18, 2014 at 12:24 PM, Carl Eugen Hoyos > wrote: > > adev dev writes: > > > >> In this case sound is very fast and ends before > >> end of movie. > > > > (Klaus knows at least a magnitude more about audio > > than I do but I wonder if he misunderstoo

Re: [Libav-user] Problem with compressing AAC files with sampling rate 16000

I will build 2.4 version with patch and write if it helps or not. Thanks! On 18 September 2014 15:41, Claudio Freire wrote: > On Thu, Sep 18, 2014 at 9:43 AM, adev dev > wrote: > > I am compressing movies from bitmaps and AAC files. Normally AAC files > are > > recorded

[Libav-user] Problem with compressing AAC files with sampling rate 16000

I am compressing movies from bitmaps and AAC files. Normally AAC files are recorded with following params: bit rate: 192000, sampling rate 44100. These audio params are set in output context (AVFormatContext) in compression. In happy day scenario it is working correctly. Movie and sound are correct