About Gstreamer Recorder



  • Dear All,

    I have question about the gstreamer.
    There is something wrong sample rate recorded..

    I try to record audio (44Khz, S16_LE, 1channel) using arecord and gstreamer.
    And probe the file using ffprobe.

    1. Record using Gstreamer
    : There is wrong sample rate and warning ...

    # gst-launch-1.0 -v pulsesrc ! audio/x-raw,format=S16LE,rate=44100,channels=1 ! filesink location=gstrec.wav
    ...
    # ffprobe gstrec.wav
    ...
    [aac @ 0x25c15e0] Format aac detected only with low score of 1, misdetection possible!
    [aac @ 0x25c2340] Error decoding AAC frame header.
    [aac @ 0x25c2340] 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 @ 0x25c2340] Reserved bit set.
    [aac @ 0x25c2340] Number of bands (8) exceeds limit (2).
    [aac @ 0x25c2340] Prediction is not allowed in AAC-LC.
    [aac @ 0x25c2340] channel element 1.10 is not allocated
    [aac @ 0x25c2340] invalid band type
    [aac @ 0x25c2340] Number of scalefactor bands in group (15) exceeds limit (14).
    [aac @ 0x25c2340] channel element 3.15 is not allocated
    [aac @ 0x25c2340] Reserved bit set.
    [aac @ 0x25c2340] Invalid Predictor Reset Group.
    [aac @ 0x25c2340] SSR 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 @ 0x25c2340] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
    [aac @ 0x25c2340] 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 @ 0x25c2340] channel element 1.13 is not allocated
    [aac @ 0x25c2340] Number of bands (8) exceeds limit (1).
    [aac @ 0x25c2340] channel element 1.4 is not allocated
    [aac @ 0x25c2340] Reserved bit set.
    [aac @ 0x25c2340] Prediction is not allowed in AAC-LC.
    [aac @ 0x25c2340] Sample rate index in program config element does not match the sample rate index configured by the container.
    [aac @ 0x25c2340] Remapped id too large
    [aac @ 0x25c2340]  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 @ 0x25c2340] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
    [aac @ 0x25c2340] Reserved bit set.
    [aac @ 0x25c2340] channel element 0.0 is not allocated
    [aac @ 0x25c2340] 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 @ 0x25c2340] channel element 2.14 is not allocated
    [aac @ 0x25c2340] channel element 3.4 is not allocated
    [aac @ 0x25c2340] channel element 3.3 is not allocated
    [aac @ 0x25c2340] channel element 1.0 is not allocated
    [aac @ 0x25c2340] 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 @ 0x25c2340] Number of bands (29) exceeds limit (26).
    [aac @ 0x25c2340] Reserved bit set.
    [aac @ 0x25c2340] channel element 0.0 is not allocated
    [aac @ 0x25c2340] 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 @ 0x25c2340] Number of bands (15) exceeds limit (7).
    [aac @ 0x25c2340] channel element 3.10 is not allocated
    [aac @ 0x25c2340] Reserved bit set.
    [aac @ 0x25c2340] Number of scalefactor bands in group (46) exceeds limit (41).
    [aac @ 0x25c2340] channel element 3.11 is not allocated
    [aac @ 0x25c2340] channel element 3.3 is not allocated
    [aac @ 0x25c2340] Prediction is not allowed in AAC-LC.
    [aac @ 0x25c2340] channel element 1.15 is not allocated
    [aac @ 0x25c2340] Prediction is not allowed in AAC-LC.
    [aac @ 0x25c2340] 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 @ 0x25c2340] Number of scalefactor bands in group (63) exceeds limit (47).
    [aac @ 0x25c2340] channel element 2.9 is not allocated
    [aac @ 0x25c2340] Reserved bit set.
    [aac @ 0x25c2340] Number of bands (13) exceeds limit (2).
    [aac @ 0x25c2340] channel element 1.1 is not allocated
    [aac @ 0x25c2340] TNS filter order 31 is greater than maximum 20.
    [aac @ 0x25c2340] Reserved bit set.
    [aac @ 0x25c2340] Prediction is not allowed in AAC-LC.
    [aac @ 0x25c2340] Reserved bit set.
    [aac @ 0x25c2340] Number of bands (20) exceeds limit (15).
    [aac @ 0x25c2340] Reserved bit set.
    [aac @ 0x25c2340] Prediction is not allowed in AAC-LC.
    [aac @ 0x25c2340] Sample rate index in program config element does not match the sample rate index configured by the container.
    [aac @ 0x25c2340] Inconsistent channel configuration.
    [aac @ 0x25c2340] get_buffer() failed
    [aac @ 0x25c2340] channel element 3.7 is not allocated
    [aac @ 0x25c2340] Pulse tool not allowed in eight short sequence.
    [aac @ 0x25c2340] channel element 1.3 is not allocated
    [aac @ 0x25c2340] Inconsistent channel configuration.
    [aac @ 0x25c2340] get_buffer() failed
    [aac @ 0x25c2340] channel element 2.5 is not allocated
    [aac @ 0x25c15e0] decoding for stream 0 failed
    [aac @ 0x25c15e0] Estimating duration from bitrate, this may be inaccurate
    [aac @ 0x25c15e0] Could not find codec parameters for stream 0 (Audio: aac (LC), 3.0, fltp, 104 kb/s): unspecified sample rate
    Consider increasing the value for the 'analyzeduration' and 'probesize' options
    gstrec.wav: End of file
    

    2. Record using arecord
    : There is correct audio information and no warning.

    # arecord -r44100 -fS16_LE -c1 alsa.wav
    ...
    # ffprobe alsa.wav
    ...
    Duration: 00:00:04.12, bitrate: 705 kb/s
        Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 1 channels, s16, 705 kb/s
    

    Please let me know why the recorded file is some problem using gstreamer.
    I already know this question is not property this forum but QT also support gstreamer...
    I hope there is gstreamer expert in here.

    Thanks.


  • Lifetime Qt Champion

    Hi,

    The first error is pretty clear: you don't have a wav file, you have an AAC file named gstrec.wav.

    From a quick search, try adding a wavenc element just before filesink.

    And indeed, you are right, that's a question better asked on the GStreamer forum. All the more, because this is a Qt user forum.



  • Dear @SGaist,

    Thanks to reply about unrelated QT.
    I'm sorry but I couldn't find the gstreamer forum :)

    I love the QT and active community~ :)

    Best Regard.


  • Lifetime Qt Champion

    It looks like the project uses mailing lists for their communication.


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