#1614 (AAC stream with SBR fails with "channel element 0.1 is not allocated") – FFmpeg (2024)

Summary of the bug:
Getting the following message repeatedly when trying to play the AAC stream with ffplay from master:
[aac @ 0x7ff1c2023800] channel element 0.1 is not allocated

Stream is ​http://3043.live.streamtheworld.com/SAM04AAC41_SC

This stream works fine in VideoLan VLC player.

How to reproduce:
./ffplay ​http://3043.live.streamtheworld.com/SAM04AAC41_SC

For future bug-reports: Please always add complete, uncut console output.

$ ffmpeg -i SAM04AAC41_SCffmpeg version N-43296-g13c5069 Copyright (c) 2000-2012 the FFmpeg developers built on Aug 7 2012 20:51:48 with gcc 4.3.2 (GCC) configuration: --cc=/usr/local/gcc-4.3.2/bin/gcc --enable-gpl --enable-libopenjpeg --enable-libvorbis --enable-libspeex --enable-libmp3lame --enable-libtheora --extra-ldflags=-lm --enable-libvpx --enable-libxavs --enable-x11grab --enable-libass --enable-nonfree --enable-libfaac --enable-libopus --enable-libfdk-aac libavutil 51. 66.101 / 51. 66.101 libavcodec 54. 50.100 / 54. 50.100 libavformat 54. 22.101 / 54. 22.101 libavdevice 54. 2.100 / 54. 2.100 libavfilter 3. 6.100 / 3. 6.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100[aac @ 0x911b780] channel element 0.1 is not allocated Last message repeated 2703 times[aac @ 0x9115500] decoding for stream 0 failed[aac @ 0x9115500] Could not find codec parameters for stream 0 (Audio: aac, stereo, s16, 60 kb/s): unspecified sample rateConsider increasing the value for the 'analyzeduration' and 'probesize' options[aac @ 0x9115500] Estimating duration from bitrate, this may be inaccurateSAM04AAC41_SC: could not find codec parameters

Hi
The "3043.live.streamtheworld.com/SAM04AAC41_SC" stream is no longer active.
Is there an alternative stream to test instead?

Maybe the attached file "SAM04AAC41_SC" corrupt.
Latest FFplay seems to play similar files OK.

Format : AAC
Format/Info : Advanced Audio Codec
Format version : Version 4
Format profile : HE-AACv2 / HE-AAC / LC
Bit rate mode : Variable
Channel(s) : 2 channels / 1 channel / 1 channel
Channel positions : Front: L R / Front: C / Front: C
Sampling rate : 44.1 KHz / 44.1 KHz / 22.05 KHz

@Xubuntu:~/ffmpeg-HEAD-6286bb5$ ./ffplay similar_file.aac
ffplay version 2.2.git-6286bb5 Copyright (c) 2003-2014 the FFmpeg developers

built on Jun 12 2014 21:03:29 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
configuration:
libavutil 52. 89.100 / 52. 89.100
libavcodec 55. 66.100 / 55. 66.100
libavformat 55. 43.100 / 55. 43.100
libavdevice 55. 13.101 / 55. 13.101
libavfilter 4. 8.100 / 4. 8.100
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 19.100 / 0. 19.100

[aac @ 0xb0e026a0] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'similar_file.aac':

Duration: 00:00:32.36, bitrate: 45 kb/s

Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 45 kb/s

41.02 M-A: -0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0

​http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/184214
https://ffmpeg.org/pipermail/ffmpeg-devel/2014-October/164080.html

Last edited 7 years ago by Carl Eugen Hoyos (previous) (diff)

Recently encountered the same issue, on latest ffmpeg 5.0. I took a look at the attached file to this ticket and it exhibits the same problem as the streams I'm seeing errors on. This "stereo" audio is actually coded as single channel elements instead of 1 channel pair element. From the header we see that it's a typical AAC-LC stereo

5E291 AudioSpecificConfig (2 bytes)5E291 audioObjectType - 2 (0x2) - AAC - LC (0 bytes)5E291 audioObjectType: 2 (0x02) - (5 bits)5E291 samplingFrequencyIndex: 3 (0x3) - (4 bits) - 48000 (0xBB80)5E292 channelConfiguration: 2 (0x2) - (4 bits) - Front: L R

but the AAC data contains two individual SCEs

0002C 1 (379 bytes)0002C raw_data_block (379 bytes)0002C SCE - single_channel_element (184 bytes)0002C id_syn_ele: 0 (0x0) - (3 bits) - SCE - single_channel_element0002E SCE - single_channel_element (376 bytes)000E4 id_syn_ele: 0 (0x0) - (3 bits) - SCE - single_channel_element00032 END - End (372 bytes)001A6 id_syn_ele: 7 (0x7) - (3 bits) - END - End001A6 byte_alignment: 0 (0x0) - (3 bits)

whereas a typica AAC stereo channel pair element would look like this

00913 1 (441 bytes)00913 raw_data_block (441 bytes)00913 CPE - channel_pair_element (439 bytes)00913 id_syn_ele: 1 (0x1) - (3 bits) - CPE - channel_pair_element00915 FIL - fill_element (438 bytes)00ACA id_syn_ele: 6 (0x6) - (3 bits) - FIL - fill_element00916 END - End (438 bytes)00ACB id_syn_ele: 7 (0x7) - (3 bits) - END - End

The parsing code in aacdec_template.c seems to intend to be able to parse stereo from individual single channel elements (SCE), but it seems to trip on a bug when getting to the second SCE.

As a workaround using the fdk_aac decoder will work, but it would be great to fix the AAC decoder in ffmpeg itself to handle such cases.

Should be fixed in f9785a78da117be0c66f6d5d58ce7277c7029ae7