I'm trying to encode some audio on Ubuntu 12.04 Linux. I need to use libfaac (using version 1.28) to encode and I want to encode at a very low bitrate - the total including transport stream overhead must be below 64kbps.
The input audio is: PCM, 48kHz, 2 channel, s16, 1536 kb/s
Using the faac command-line tool, I get the output below, encoding 5 minutes of audio - so there's plenty of time for the bitrate to average out. I cannot get it to go below 64kbps, or above 165kbps for that matter. I have tried setting some of the other parameters, but to no avail. When I set the bitrate > 64, it works fine.
$ ./faac -b 40 -w -o /tmp/yellow-audio-40k.mp4 /tmp/yellow-audio.wavFreeware Advanced Audio Coder
FAAC 1.28
Average bitrate: 64 kbps
Quantization quality: 100
Bandwidth: 5442 Hz
Object type: Low Complexity(MPEG-4) + M/S
Container format: MPEG-4 File Format (MP4)
Encoding /tmp/yellow-audio.wav to /tmp/yellow-audio-40k.mp4
frame | bitrate | elapsed/estim | play/CPU | ETA
14063/14063 (100%)| 64.2 | 10.7/10.7 | 27.91x | 0.0
$ mediainfo /tmp/yellow-audio-40k.mp4
General
Complete name : /tmp/yellow-audio-40k.mp4
Format : MPEG-4
Format profile : Base Media / Version 2
Codec ID : mp42
File size : 2.35 MiB
Duration : 5mn 0s
Overall bit rate mode : Variable
Overall bit rate : 65.7 Kbps
Encoded date : UTC 2012-12-05 00:03:38
Tagged date : UTC 2012-12-05 00:03:49
Writing application : FAAC 1.28
Audio
ID : 1
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : 40
Duration : 5mn 0s
Bit rate mode : Variable
Bit rate : 64.2 Kbps
Maximum bit rate : 73.7 Kbps
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 KHz
Compression mode : Lossy
Stream size : 2.30 MiB (98%)
Language : English
Encoded date : UTC 2012-12-05 00:03:38
Tagged date : UTC 2012-12-05 00:03:49
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Hmmm, why do you "have to" use faac only?
Can't you use NeroAacEnc under WINE for example?
Keep in mind that lower bitrates sound better/acceptable only with the High-Efficiency mode
(read: Spectral Band Replication) -
I tried it with ffmpeg libvo-aacenc instead of libfaac (on windows) and it was able to reach the targets of 40kb/s, 48kb/s AAC . Some AAC encoders have unavailable bitrate options, and only specific numbers work (e.g. you might not be able to use 45kb/s)
Another option might be to pipe ffmpeg into another aac encoder like neroaac
see this thread:
https://forum.videohelp.com/threads/322328-libfaac-encoding-with-ffmpeg
bat99 is on linux , so either option should work for you on linux -
I agree with El Heggunte,
And even to my non-audiophile ears , there is a world of difference between something like neroaac HE-AAC and libfaac in terms of quality, especially at low bitrates -
Finally figured it out. In order to encode at bitrates < 64kbps, you must also set the bandwidth:
-c <freq> Set the bandwidth in Hz. (default=automatic)
the bitrate and bandwidth parameters work in tandem to generate an output bitrate. It's automatically set to
about 1/2 of the sample rate, but you must set it much lower (in my case <5000) to allow libfaac to
encode at 40kbps or similar. -
Yes, that's done the job.
I suppose this low bandwidth is OK for speech.
Code:@ubuntu:~$ faac -b 40 -c 4000 -o foo.m4a foo.wav Freeware Advanced Audio Coder FAAC 1.28 Average bitrate: 40 kbps Quantization quality: 100 Bandwidth: 4000 Hz Object type: Low Complexity(MPEG-4) + M/S Container format: MPEG-4 File Format (MP4) Encoding foo.wav to foo.m4a frame | bitrate | elapsed/estim | play/CPU | ETA 1297/1297 (100%)| 42.1 | 2.1/2.1 | 14.68x | 0.0
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