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parse AAC file with adts headers to extract out AAC
EventHorizon
post Jun 28 2012, 01:46
Post #1





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Member No.: 100271



I am writing some c code to read AAC files with ADTS headers
with goal of extracting just the AAC data for downstream
hardware decompression into linear PCM. From command line
this works :

ffmpeg -i input.aac output.wav

CODE

ffmpeg version git-2012-06-13-4a6d790 Copyright © 2000-2012 the FFmpeg developers
built on Jun 13 2012 14:43:00 with llvm_gcc 4.2.1 (Based on Apple Inc. build 5658) (LLVM build 2336.9.00)
configuration:
libavutil 51. 58.100 / 51. 58.100
libavcodec 54. 25.100 / 54. 25.100
libavformat 54. 6.101 / 54. 6.101
libavdevice 54. 0.100 / 54. 0.100
libavfilter 2. 78.101 / 2. 78.101
libswscale 2. 1.100 / 2. 1.100
libswresample 0. 15.100 / 0. 15.100
[aac @ 0x7fc39a03d800] Format aac detected only with low score of 25, misdetection possible!
[aac @ 0x7fc39a03fc00] channel element 0.5 is not allocated
[aac @ 0x7fc39a03d800] max_analyze_duration 5000000 reached at 5013333
[aac @ 0x7fc39a03d800] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'input.aac':
Duration: 00:00:06.03, bitrate: 46 kb/s
Stream #0:0: Audio: aac, 48000 Hz, mono, s16, 46 kb/s
Output #0, wav, to 'output.wav':
Metadata:
encoder : Lavf54.6.101
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (aac -> pcm_s16le)
Press [q] to stop, [?] for help
[aac @ 0x7fc39a03fc00] Number of scalefactor bands in group (56) exceeds limit (49).
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0x7fc39a03fc00] channel element 2.10 is not allocated
Error while decoding stream #0:0: Operation not permitted
size= 540kB time=00:00:09.71 bitrate= 455.6kbits/s
video:0kB audio:540kB global headers:0kB muxing overhead 0.008319%



the output.wav plays OK using say ffplay or afplay
interestingly my input.aac plays OK with ffplay
yet fails to play using afplay : Error: AudioFileOpen failed ('sync')


here is the input file :

CODE

mediainfo input.aac

General
Complete name : input.aac
Format : ADTS
Format/Info : Audio Data Transport Stream
File size : 34.0 KiB
Overall bit rate mode : Variable

Audio
Format : AAC
Format/Info : Advanced Audio Codec
Format version : Version 4
Format profile : LC
Bit rate mode : Variable
Bit rate : 47.1 Kbps
Minimum bit rate : 39.8 Kbps
Maximum bit rate : 55.5 Kbps
Channel(s) : 1 channel
Channel positions : Front: C
Sampling rate : 48.0 KHz
Compression mode : Lossy
Stream size : 33.8 KiB (100%)


here is output file after above ffmpeg conversion :

CODE

mediainfo output.wav

General
Complete name : output.wav
Format : Wave
File size : 540 KiB
Duration : 5s 760ms
Overall bit rate mode : Constant
Overall bit rate : 768 Kbps

Audio
ID : 0
Format : PCM
Format settings, Endianness : Little
Codec ID : 1
Duration : 5s 760ms
Bit rate mode : Constant
Bit rate : 768 Kbps
Channel(s) : 1 channel
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Stream size : 540 KiB (100%)



to write my c code I am reading the AAC ISO spec 13818-7
however I am not sure at what point I have parsed out the AAC data format.

Here is output from my code when parsing an input AAC file :

CODE


about to show values for fixed header

0 syncword fff 4095
1 ID 1 1
2 layer 3 3
3 protection_absent 1 1
4 profile 1 1
5 sampling_frequency_index 4 4
6 private_bit 0 0
7 channel_configuration 2 2
8 original/copy 0 0
9 home 0 0
about to show values for variable header

0 copyright_identification_bit 0 0
1 copyright_identification_start 0 0
2 frame_length 802 2050
3 adts_buffer_fullness 600 1536
4 number_of_raw_data_blocks_in_frame 0 0



So a few things might help : (1) some tool to breakdown my input.aac
to indicate each frame or (2) other ISO spec relevant guideposts
so I can confirm my code is correctly identifying just the AAC data.
When I execute my code it reaches what the ISO spec calls the : raw_data_block.
Do I just output the bytes of each frame and consider that the AAC data ?

Once I can extract out the AAC data from my input.aac file
I will then feed it into some Core Audio API call to decompress into linear PCM.

thanks
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