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  • Why ffmpeg can't convert this .ts to .mp4

    10 janvier 2023, par cpo

    I have ffmpeg running on my Synology NAS, and would like to use it to convert DVR-recorded .ts videos into .mp4 format. I don't fully understand this stuff, and have researched to no avail.

    


    Here is the command I am running with result :

    


    $ ffmpeg -analyzeduration 2147483647 -probesize 2147483647 -i 'video.ts' -acodec -vcodec copy 'video.mp4'

ffmpeg version 4.1.8 Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 8.5.0 (GCC)
  configuration: --prefix=/usr --incdir='${prefix}/include/ffmpeg' --arch=i686 --target-os=linux --cross-prefix=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu- --enable-cross-compile --enable-optimizations --enable-pic --enable-gpl --enable-shared --disable-static --disable-stripping --enable-version3 --enable-encoders --enable-pthreads --disable-protocols --disable-protocol=rtp --enable-protocol=file --enable-protocol=pipe --disable-muxer=image2 --disable-muxer=image2pipe --disable-swscale-alpha --disable-ffplay --disable-ffprobe --disable-doc --disable-devices --disable-bzlib --disable-altivec --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --disable-vaapi --disable-cuvid --disable-nvenc --disable-decoder=aac --disable-decoder=aac_fixed --disable-encoder=aac --disable-decoder=amrnb --disable-decoder=ac3 --disable-decoder=ac3_fixed --disable-encoder=zmbv --disable-encoder=dca --disable-decoder=dca --disable-encoder=ac3 --disable-encoder=ac3_fixed --disable-encoder=eac3 --disable-decoder=eac3 --disable-encoder=truehd --disable-decoder=truehd --disable-encoder=hevc_vaapi --disable-decoder=hevc --disable-muxer=hevc --disable-demuxer=hevc --disable-parser=hevc --disable-bsf=hevc_mp4toannexb --x86asmexe=yasm --cc=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu-wrap-gcc --enable-yasm --enable-libx264 --enable-encoder=libx264
  libavutil      56. 22.100 / 56. 22.100
  libavcodec     58. 35.100 / 58. 35.100
  libavformat    58. 20.100 / 58. 20.100
  libavdevice    58.  5.100 / 58.  5.100
  libavfilter     7. 40.101 /  7. 40.101
  libswscale      5.  3.100 /  5.  3.100
  libswresample   3.  3.100 /  3.  3.100
  libpostproc    55.  3.100 / 55.  3.100
[mpegts @ 0x5591c4feb4c0] Could not find codec parameters for stream 1 (Audio: aac ([15][0][0][0] / 0x000F), 0 channels, 199 kb/s): unspecified sample rate
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, mpegts, from 'video.ts':
  Duration: 02:01:58.77, start: 1.405122, bitrate: 4094 kb/s
  Program 1 
    Metadata:
      service_name    : Service01
      service_provider: FFmpeg
    Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Stream #0:1[0x101]: Audio: aac ([15][0][0][0] / 0x000F), 0 channels, 199 kb/s
[NULL @ 0x5591c6034980] Unable to find a suitable output format for 'copy'
copy: Invalid argument


    


    Here is the information from video.ts as displayed in VLC :

    


    Media information

    


    It's possible I just don't have the right encoders included with my ffmpeg packaged (obtained from https://synocommunity.com/package/ffmpeg). Here are the audio encoders that it appears I have.

    


    A..... adpcm_adx            SEGA CRI ADX ADPCM
 A..... g722                 G.722 ADPCM (codec adpcm_g722)
 A..... g726                 G.726 ADPCM (codec adpcm_g726)
 A..... g726le               G.726 little endian ADPCM ("right-justified") (codec adpcm_g726le)
 A..... adpcm_ima_qt         ADPCM IMA QuickTime
 A..... adpcm_ima_wav        ADPCM IMA WAV
 A..... adpcm_ms             ADPCM Microsoft
 A..... adpcm_swf            ADPCM Shockwave Flash
 A..... adpcm_yamaha         ADPCM Yamaha
 A..... alac                 ALAC (Apple Lossless Audio Codec)
 A..... libopencore_amrnb    OpenCORE AMR-NB (Adaptive Multi-Rate Narrow-Band) (codec amr_nb)
 A..... aptx                 aptX (Audio Processing Technology for Bluetooth)
 A..... aptx_hd              aptX HD (Audio Processing Technology for Bluetooth)
 A..... comfortnoise         RFC 3389 comfort noise generator
 A..... flac                 FLAC (Free Lossless Audio Codec)
 A..... g723_1               G.723.1
 A..X.. mlp                  MLP (Meridian Lossless Packing)
 A..... mp2                  MP2 (MPEG audio layer 2)
 A..... mp2fixed             MP2 fixed point (MPEG audio layer 2) (codec mp2)
 A..... libmp3lame           libmp3lame MP3 (MPEG audio layer 3) (codec mp3)
 A..... nellymoser           Nellymoser Asao
 A..X.. opus                 Opus
 A..... pcm_alaw             PCM A-law / G.711 A-law
 A..... pcm_f32be            PCM 32-bit floating point big-endian
 A..... pcm_f32le            PCM 32-bit floating point little-endian
 A..... pcm_f64be            PCM 64-bit floating point big-endian
 A..... pcm_f64le            PCM 64-bit floating point little-endian
 A..... pcm_mulaw            PCM mu-law / G.711 mu-law
 A..... pcm_s16be            PCM signed 16-bit big-endian
 A..... pcm_s16be_planar     PCM signed 16-bit big-endian planar
 A..... pcm_s16le            PCM signed 16-bit little-endian
 A..... pcm_s16le_planar     PCM signed 16-bit little-endian planar
 A..... pcm_s24be            PCM signed 24-bit big-endian
 A..... pcm_s24daud          PCM D-Cinema audio signed 24-bit
 A..... pcm_s24le            PCM signed 24-bit little-endian
 A..... pcm_s24le_planar     PCM signed 24-bit little-endian planar
 A..... pcm_s32be            PCM signed 32-bit big-endian
 A..... pcm_s32le            PCM signed 32-bit little-endian
 A..... pcm_s32le_planar     PCM signed 32-bit little-endian planar
 A..... pcm_s64be            PCM signed 64-bit big-endian
 A..... pcm_s64le            PCM signed 64-bit little-endian
 A..... pcm_s8               PCM signed 8-bit
 A..... pcm_s8_planar        PCM signed 8-bit planar
 A..... pcm_u16be            PCM unsigned 16-bit big-endian
 A..... pcm_u16le            PCM unsigned 16-bit little-endian
 A..... pcm_u24be            PCM unsigned 24-bit big-endian
 A..... pcm_u24le            PCM unsigned 24-bit little-endian
 A..... pcm_u32be            PCM unsigned 32-bit big-endian
 A..... pcm_u32le            PCM unsigned 32-bit little-endian
 A..... pcm_u8               PCM unsigned 8-bit
 A..... pcm_vidc             PCM Archimedes VIDC
 A..... real_144             RealAudio 1.0 (14.4K) (codec ra_144)
 A..... roq_dpcm             id RoQ DPCM
 A..X.. s302m                SMPTE 302M
 A..... sbc                  SBC (low-complexity subband codec)
 A..X.. sonic                Sonic
 A..X.. sonicls              Sonic lossless
 A..... tta                  TTA (True Audio)
 A..X.. vorbis               Vorbis
 A..... wavpack              WavPack
 A..... wmav1                Windows Media Audio 1
 A..... wmav2                Windows Media Audio 2


    


  • Please help me figure out why ffmpeg can't convert this .ts to .mp4

    10 janvier 2023, par cpo

    I have ffmpeg running on my Sinology NAS, and would like to use it to convert DVR-recorded .ts videos into .mp4 format. I don't fully understand this stuff, and have researched to no avail. Perhaps someone who has some expertise in this area can quickly help me get up and running. Thanks !

    


    Here is the command I am running with result :

    


    $ ffmpeg -analyzeduration 2147483647 -probesize 2147483647 -i 'video.ts' -acodec -vcodec copy 'video.mp4'

ffmpeg version 4.1.8 Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 8.5.0 (GCC)
  configuration: --prefix=/usr --incdir='${prefix}/include/ffmpeg' --arch=i686 --target-os=linux --cross-prefix=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu- --enable-cross-compile --enable-optimizations --enable-pic --enable-gpl --enable-shared --disable-static --disable-stripping --enable-version3 --enable-encoders --enable-pthreads --disable-protocols --disable-protocol=rtp --enable-protocol=file --enable-protocol=pipe --disable-muxer=image2 --disable-muxer=image2pipe --disable-swscale-alpha --disable-ffplay --disable-ffprobe --disable-doc --disable-devices --disable-bzlib --disable-altivec --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --disable-vaapi --disable-cuvid --disable-nvenc --disable-decoder=aac --disable-decoder=aac_fixed --disable-encoder=aac --disable-decoder=amrnb --disable-decoder=ac3 --disable-decoder=ac3_fixed --disable-encoder=zmbv --disable-encoder=dca --disable-decoder=dca --disable-encoder=ac3 --disable-encoder=ac3_fixed --disable-encoder=eac3 --disable-decoder=eac3 --disable-encoder=truehd --disable-decoder=truehd --disable-encoder=hevc_vaapi --disable-decoder=hevc --disable-muxer=hevc --disable-demuxer=hevc --disable-parser=hevc --disable-bsf=hevc_mp4toannexb --x86asmexe=yasm --cc=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu-wrap-gcc --enable-yasm --enable-libx264 --enable-encoder=libx264
  libavutil      56. 22.100 / 56. 22.100
  libavcodec     58. 35.100 / 58. 35.100
  libavformat    58. 20.100 / 58. 20.100
  libavdevice    58.  5.100 / 58.  5.100
  libavfilter     7. 40.101 /  7. 40.101
  libswscale      5.  3.100 /  5.  3.100
  libswresample   3.  3.100 /  3.  3.100
  libpostproc    55.  3.100 / 55.  3.100
[mpegts @ 0x5591c4feb4c0] Could not find codec parameters for stream 1 (Audio: aac ([15][0][0][0] / 0x000F), 0 channels, 199 kb/s): unspecified sample rate
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, mpegts, from 'video.ts':
  Duration: 02:01:58.77, start: 1.405122, bitrate: 4094 kb/s
  Program 1 
    Metadata:
      service_name    : Service01
      service_provider: FFmpeg
    Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Stream #0:1[0x101]: Audio: aac ([15][0][0][0] / 0x000F), 0 channels, 199 kb/s
[NULL @ 0x5591c6034980] Unable to find a suitable output format for 'copy'
copy: Invalid argument


    


    Here is the information from video.ts as displayed in VLC :

    


    Media information

    


    It's possible I just don't have the right encoders included with my ffmpeg packaged (obtained from https://synocommunity.com/package/ffmpeg). Here are the audio encoders that it appears I have.

    


    A..... adpcm_adx            SEGA CRI ADX ADPCM
 A..... g722                 G.722 ADPCM (codec adpcm_g722)
 A..... g726                 G.726 ADPCM (codec adpcm_g726)
 A..... g726le               G.726 little endian ADPCM ("right-justified") (codec adpcm_g726le)
 A..... adpcm_ima_qt         ADPCM IMA QuickTime
 A..... adpcm_ima_wav        ADPCM IMA WAV
 A..... adpcm_ms             ADPCM Microsoft
 A..... adpcm_swf            ADPCM Shockwave Flash
 A..... adpcm_yamaha         ADPCM Yamaha
 A..... alac                 ALAC (Apple Lossless Audio Codec)
 A..... libopencore_amrnb    OpenCORE AMR-NB (Adaptive Multi-Rate Narrow-Band) (codec amr_nb)
 A..... aptx                 aptX (Audio Processing Technology for Bluetooth)
 A..... aptx_hd              aptX HD (Audio Processing Technology for Bluetooth)
 A..... comfortnoise         RFC 3389 comfort noise generator
 A..... flac                 FLAC (Free Lossless Audio Codec)
 A..... g723_1               G.723.1
 A..X.. mlp                  MLP (Meridian Lossless Packing)
 A..... mp2                  MP2 (MPEG audio layer 2)
 A..... mp2fixed             MP2 fixed point (MPEG audio layer 2) (codec mp2)
 A..... libmp3lame           libmp3lame MP3 (MPEG audio layer 3) (codec mp3)
 A..... nellymoser           Nellymoser Asao
 A..X.. opus                 Opus
 A..... pcm_alaw             PCM A-law / G.711 A-law
 A..... pcm_f32be            PCM 32-bit floating point big-endian
 A..... pcm_f32le            PCM 32-bit floating point little-endian
 A..... pcm_f64be            PCM 64-bit floating point big-endian
 A..... pcm_f64le            PCM 64-bit floating point little-endian
 A..... pcm_mulaw            PCM mu-law / G.711 mu-law
 A..... pcm_s16be            PCM signed 16-bit big-endian
 A..... pcm_s16be_planar     PCM signed 16-bit big-endian planar
 A..... pcm_s16le            PCM signed 16-bit little-endian
 A..... pcm_s16le_planar     PCM signed 16-bit little-endian planar
 A..... pcm_s24be            PCM signed 24-bit big-endian
 A..... pcm_s24daud          PCM D-Cinema audio signed 24-bit
 A..... pcm_s24le            PCM signed 24-bit little-endian
 A..... pcm_s24le_planar     PCM signed 24-bit little-endian planar
 A..... pcm_s32be            PCM signed 32-bit big-endian
 A..... pcm_s32le            PCM signed 32-bit little-endian
 A..... pcm_s32le_planar     PCM signed 32-bit little-endian planar
 A..... pcm_s64be            PCM signed 64-bit big-endian
 A..... pcm_s64le            PCM signed 64-bit little-endian
 A..... pcm_s8               PCM signed 8-bit
 A..... pcm_s8_planar        PCM signed 8-bit planar
 A..... pcm_u16be            PCM unsigned 16-bit big-endian
 A..... pcm_u16le            PCM unsigned 16-bit little-endian
 A..... pcm_u24be            PCM unsigned 24-bit big-endian
 A..... pcm_u24le            PCM unsigned 24-bit little-endian
 A..... pcm_u32be            PCM unsigned 32-bit big-endian
 A..... pcm_u32le            PCM unsigned 32-bit little-endian
 A..... pcm_u8               PCM unsigned 8-bit
 A..... pcm_vidc             PCM Archimedes VIDC
 A..... real_144             RealAudio 1.0 (14.4K) (codec ra_144)
 A..... roq_dpcm             id RoQ DPCM
 A..X.. s302m                SMPTE 302M
 A..... sbc                  SBC (low-complexity subband codec)
 A..X.. sonic                Sonic
 A..X.. sonicls              Sonic lossless
 A..... tta                  TTA (True Audio)
 A..X.. vorbis               Vorbis
 A..... wavpack              WavPack
 A..... wmav1                Windows Media Audio 1
 A..... wmav2                Windows Media Audio 2


    


  • Gstreamer AAC encoding no more supported ?

    22 juillet 2016, par Gianks

    i’d like to include AAC as one of the compatible formats in my app but i’m having troubles with its encoding.
    FAAC seems to be missing in GStreamer-1.0 Debian-derived packages (see Ubuntu) and the main reason for that (if i got it correctly) is the presence of avenc_aac (Lunchpad bugreport) as a replacement.

    I’ve tried the following :

    gst-launch-1.0 filesrc location="src.avi" ! tee name=t  t.! queue ! decodebin ! progressreport ! x264enc ! mux. t.! queue ! decodebin ! audioconvert ! audioresample ! avenc_aac compliance=-2 ! mux. avmux_mpegts name=mux ! filesink location=/tmp/test.avi

    It hangs prerolling with :

    ERROR libav :0:: AAC bitstream not in ADTS format and extradata missing

    Using mpegtsmux instead of avmux_mpegts seems to work since the file is created but it results with no working audio (with some players it’s completely unplayable).

    This is the trace of mplayer :

    Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
    
[aac @ 0x7f2860d6c3c0]channel element 3.15 is not allocated
    
[aac @ 0x7f2860d6c3c0]Sample rate index in program config element does not match the sample rate index configured by the container.
    
[aac @ 0x7f2860d6c3c0]Inconsistent channel configuration.
    
[aac @ 0x7f2860d6c3c0]get_buffer() failed
    
[aac @ 0x7f2860d6c3c0]Assuming an incorrectly encoded 7.1 channel layout instead of a spec-compliant 7.1(wide) layout, use -strict 1 to decode according to the specification instead.
    
[aac @ 0x7f2860d6c3c0]Reserved bit set.
    
[aac @ 0x7f2860d6c3c0]Number of bands (20) exceeds limit (14).
    
[aac @ 0x7f2860d6c3c0]invalid band type
    
[aac @ 0x7f2860d6c3c0]More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
    
[aac @ 0x7f2860d6c3c0]Reserved bit set.
    
[aac @ 0x7f2860d6c3c0]Number of bands (45) exceeds limit (28).
    
Unknown/missing audio format -> no sound
    
ADecoder init failed :(
    
Opening audio decoder: [faad] AAC (MPEG2/4 Advanced Audio Coding)
    
FAAD: compressed input bitrate missing, assuming 128kbit/s!
    
AUDIO: 44100 Hz, 2 ch, floatle, 128.0 kbit/9.07% (ratio: 16000->176400)
    
Selected audio codec: [faad] afm: faad (FAAD AAC (MPEG-2/MPEG-4 Audio))
    
==========================================================================
    
AO: [pulse] 44100Hz 2ch floatle (4 bytes per sample)
    
Starting playback...
    
FAAD: error: Bitstream value not allowed by specification, trying to resync!
    
FAAD: error: Invalid number of channels, trying to resync!
    
FAAD: error: Invalid number of channels, trying to resync!
    
FAAD: error: Bitstream value not allowed by specification, trying to resync!
    
FAAD: error: Invalid number of channels, trying to resync!
    
FAAD: error: Bitstream value not allowed by specification, trying to resync!
    
FAAD: error: Channel coupling not yet implemented, trying to resync!
    
FAAD: error: Invalid number of channels, trying to resync!
    
FAAD: error: Invalid number of channels, trying to resync!
    
FAAD: error: Bitstream value not allowed by specification, trying to resync!
    
FAAD: Failed to decode frame: Bitstream value not allowed by specification 
    
Movie-Aspect is 1.33:1 - prescaling to correct movie aspect.
    
VO: [vdpau] 640x480 => 640x480 Planar YV12 
    
A:3602.2 V:3600.0 A-V:  2.143 ct:  0.000   3/  3 ??% ??% ??,?% 0 0 
    
FAAD: error: Array index out of range, trying to resync!
    
FAAD: error: Bitstream value not allowed by specification, trying to resync!
    
FAAD: error: Bitstream value not allowed by specification, trying to resync!
    
FAAD: error: Unexpected fill element with SBR data, trying to resync!
    
FAAD: error: Bitstream value not allowed by specification, trying to resync!
    
FAAD: error: Bitstream value not allowed by specification, trying to resync!
    
FAAD: error: Channel coupling not yet implemented, trying to resync!
    
FAAD: error: Invalid number of channels, trying to resync!
    
FAAD: error: PCE shall be the first element in a frame, trying to resync!
    
FAAD: error: Invalid number of channels, trying to resync!
    
FAAD: Failed to decode frame: Invalid number of channels 
    
A:3602.2 V:3600.1 A-V:  2.063 ct:  0.000   4/  4 ??% ??% ??,?% 0 0

    These the messages produced by VLC (10 seconds of playback) :

    ts info: MPEG-4 descriptor not found for pid 0x42 type 0xf
    
core error: option sub-original-fps does not exist
    
subtitle warning: failed to recognize subtitle type
    
core error: no suitable demux module for `file/subtitle:///tmp//test.avi.idx'
    
avcodec info: Using NVIDIA VDPAU Driver Shared Library 361.42 Tue Mar 22 17:29:16 PDT 2016 for hardware decoding.
    
core warning: VoutDisplayEvent 'pictures invalid'
    
core warning: VoutDisplayEvent 'pictures invalid'
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio warning: ADTS CRC not supported
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio warning: ADTS CRC not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio warning: Invalid ADTS header
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported
    
packetizer_mpeg4audio error: Multiple blocks per frame in ADTS not supported

    Using the error of the hanging pipeline I’ve finally discovered that avenc_aac should be told in such way to output the data NOT in RAW AAC but in ADTS AAC, the point is that i’ve no idea how to do that with Gstreamer. See here, bottom of the page : FFMPEG Ticket

    At this point since i’ve found no documentation seems right to say we have no support for AAC encoding in GStreamer... which isn’t true, i guess ! (IMHO anyway seems strange the missing of FAAC if AVENC_AAC requires all the time to be set in experimental mode)

    Can someone propose a working pipeline for this ?

    UPDATE

    After some more research i’ve found (via gst-inspect on avenc_aac) what i’m probably looking for but i don’t know how to setup it as needed.
    Have a look at stream-format :

    Pad Templates:
    
  SRC template: 'src'
    
    Availability: Always
    
    Capabilities:
    
      audio/mpeg
    
               channels: [ 1, 6 ]
    
                   rate: [ 4000, 96000 ]
    
            mpegversion: 4
    
          stream-format: raw
    
        base-profile: lc

    Thanks

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