how to decode g711 raw stream?

[niror]

I would like to decode that file on pc running window or linux to more familiar stream so i can mux it together with h264 video file.
so far couldn't find on the web any player that can play the file, nor decode it.

any help on that would be appreciated.

audio.zip- g711 with 1KHz tone signal.
audio parameters: 16000 sampling rate.

ran

[El Heggunte]

Hi there.

You can open it in Goldwave (trialware), select manually the file type, save it as an uncompressed .WAV file, and later convert this to a format suitable for the MP4 container.

[niror]

hi,

thank u for the reply

I'm currently working on a code for doing all that I mentioned above.
that means that i need a c library, or at least command line program for decoding the g711 file.

so far i managed to convert this 'audio.g711' file to 'audio.au' file by adding a header to it as described in:
"http://en.wikipedia.org/wiki/Au_file_format".
So now decoders know what type of stream thay have to deal with,
and winLame can do the job. but there is no command line for that

for the muxing part, i found a great piece of freeware called "tsMuxeR" that can accept mp3, aac, mpa, mp4, wav, AC3 , pcm, lpcm, and other well known formats, unfortunately i'm new to this codecs business and cannot figure out how to convert my audio.au file to one of those formats, and what do i need for that, decoder? encoder? filter?

thanks,
ran

[poisondeathray]

If you have converted it correctly to .au , then ffmpeg or sox should be able to handle it

ffmpeg should be able to mux your audio with video into various containers as well

audacity can handle it if you import as raw. It has limited CLI options , but open source, so you can have a look at the code.

[niror]

thanks guys
sox did the job.
(i had to also upsample it to fit the "tsMuxeR")

now i'm facing some problems with the video converting, but i guess i should post it in the video section of this forum.

again, thanks a lot

ran

[Niko68]

The G711 A-law compression algorithm is used in Europe, and almost all over the world. The A-law is logarithmic, and lighter for the computer to process. The G711 A-law encodes a 13 bit signed linear PCM sample into logarithmic 8-bit sample. As a result, the G711 encoder will be able to produce a 64 kbit/s bitstream for a signal that is sampled at 8 KHz. The A-law compression enables more quantization levels at lower signal levels. A 13-bit signed linear audio sample as input is converted to an 8 bit value as follows:

To optimize the telephony the G711 aLaw codes is perfect. You can modify the dynamic range of an analog signal for digitizing as well. This is a logarithmic algorithm and it has been designed to be simpler for computer processing than uLaw algorithm. It also provides a more dynamic range resulted in a better sound quality because sampling artifacts are better suppressed. It has a very low processor requirements and needs at least 128 kbps for two-way.

According to my opinion, in communication it is vital to use the best possible solutions that provide the quality and excellence. For this reason, the solution of G711 codec, also known as Pulse Code Modulation (PCM) is a frequently used waveform codec. The G711 uses a sampling rate of 8000 samples/sec, with a tolerance of 50 parts per million (ppm). The G711 codec comes in two different compression algorithms: µ-law and A-law.

I wish to share a useful article about the topic. If you are interested you find it here:
http://voip-sip-sdk.com/p_214-g711-alaw-codec-voip.html

Best Wishes,
Niko