search for: adpcm

...| 2 - libswfdec/swfdec_as_frame.c | 8 ++-- libswfdec/swfdec_audio.c | 1 libswfdec/swfdec_audio_event.c | 24 +++++-------- libswfdec/swfdec_audio_event.h | 3 + libswfdec/swfdec_shape_parser.c | 8 ++-- libswfdec/swfdec_sound.c | 1 test/sound/adpcm-2-2.swf.1.0.raw |binary test/sound/adpcm-2.swf.1.0.raw |binary test/sound/adpcm-3-2.swf.1.0.raw |binary test/sound/adpcm-3.swf.1.0.raw |binary test/sound/adpcm-4-2.swf.1.0.raw |binary test/sound/adpcm-4.swf.1.0.raw |binary test/sound/adpcm-5-2.swf.1.0.raw |binary test/sound/adpcm-5.swf...

...- player = NULL; - - return 0; -} - diff-tree 9490205e6f41fe442872a583d4170d3f25c5a0b4 (from 9f317479647298bbf1e720eb9c253a1d133d5744) Author: Benjamin Otte <otte at gnome.org> Date: Sun Aug 26 19:23:08 2007 +0200 rename since we have to do one iteration more diff --git a/test/sound/adpcm-2-2.swf.0.0.raw b/test/sound/adpcm-2-2.swf.0.0.raw deleted file mode 100644 index f959559..0000000 Binary files a/test/sound/adpcm-2-2.swf.0.0.raw and /dev/null differ diff --git a/test/sound/adpcm-2-2.swf.1.0.raw b/test/sound/adpcm-2-2.swf.1.0.raw new file mode 100644 index 0000000..f959559 Binary...

Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get this codec? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030916/b8be2453/attachment.htm

When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf gsm Call accepted by 69.73.19.178 (format ADPCM) -- Format for call is ADPCM My settings: [general] port=4569 register => xxxx:xxxx@iaxtel.com bandwidth=high jitterbuffer=no tos=lowdelay [voipjet] type=peer host= xxx.xxx.xxx.xx secret= xxx auth=md5 n...

I found some posts regarding this issue dating of September 2003, but no real answer. The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help migration. Is there an existing format/codec for this? If not, can I make myself a shared object in /usr/lib/asterisk/modules? Is this easy??? :-( Thanks, Yves...

...as wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - - - - - - - - - - - - - - gsm - - 3001 3002 6999 3001 3000 10999 - - 40994 8000 6999 - - 13998...

...1 12:01:33 2007 +0200 fix names of files in EXTRA_DIST diff --git a/test/sound/Makefile.am b/test/sound/Makefile.am index e576d5e..41e5e2b 100644 --- a/test/sound/Makefile.am +++ b/test/sound/Makefile.am @@ -19,20 +19,20 @@ sound_LDFLAGS = $(SWFDEC_LIBS) $(CAIRO_L EXTRA_DIST = \ README \ adpcm-2.swf \ - adpcm-2.swf.0.0.raw \ + adpcm-2.swf.1.0.raw \ adpcm-2-2.swf \ - adpcm-2-2.swf.0.0.raw \ + adpcm-2-2.swf.1.0.raw \ adpcm-3.swf \ - adpcm-3.swf.0.0.raw \ + adpcm-3.swf.1.0.raw \ adpcm-3-2.swf \ - adpcm-3-2.swf.0.0.raw \ + adpcm-3-2.swf.1.0.raw \ adpcm-4.swf \ - adpcm-4.swf.0.0.raw \...

I want to record audio in ADPCM format. According to the "show codecs" output of Asterisk, it looks like it supports adpcm. But I do not know what to tell the "RECORD FILE" directive in my AGI script. The RECORD FILE command usually has this form: RECORD FILE <filename> <format> <timeout>...

Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only in o...

I have been looking at the medium-rate codecs in Asterisk - ADPCM and G.726. Both of these are adaptive PCM codecs - the G.726 one is a little more expensive in processing power, however both are 32k bit-rate. I am experiencing problems using G.726 where the audio level is high. It produces loud clicks as if clipping. For quiet audio however, it seems fine. ADP...

Hi All; I need to buy one IAXy device, but I discovered that it supports only g711 and ADPCM codec, so I was wonder that it does not support g729 or GSM?! Anyway, what is that ADPCM and how much it consumes bandwitdh? Also, asterisk support such codec? What its name in the configuration? Any advise? Regards Bilal ___________________________________________________________________...

...well. If you visit the AeroPlayer FAQ, they claim that the Sony Clie DSP API is unreleased, and thus not supported. But it looks like Sony *did* recently release a library and API. I downloaded it this afternoon, and was pretty disappointed. It only allows your application to play a proprietary ADPCM block of data: <p> The Pa1Lib library can play MIDI and ADPCM data. * MIDI: SMF 0 * ADPCM: Yamaha ADPCM o should not contain a RIFF header o 4 KHz or 8 KHz sampling rate o single channel (mono) The Yamaha ADPCM format is not compatible with IMA...

Hi everyone, We are using the IAXy boxes and Asterisk over the internet and I was wondering if Asterisk can do a codec translation during a call in order to lower the bandwidth that the comunications consumes? I mean, the IAXy boxes only support the ADPCM and uLAW codecs, but for a certain number of calls our bandwidth runs out, then I think if Asterisk can convert the signal that comes in ADPCM format (the lighter codec out of these two) to another codec that use less bandwidth to interconect to another Asterisk Server via IAX2 protocol. We are tr...

Greetings: It would be nice if Icecast supported RTSP; however I would appreciate any suggestions for a small RTSP/RTP solution to encode 8kHz mono audio in GSM or ADPCM and service multiple unicast client connections. The ideal would be a black-box hardware solution with an audio input and ethernet interface similar to broadcast studio IP audio links or the network audio capabilities of certain X-terminals. An acceptable solution would be *nix or Win32 software...

Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only in o...

...,1 exten => 99,2,Record,/tmp/pcmfile:pcm exten => 99,3,Wait,1 exten => 99,4,Playback,/tmp/pcmfile exten => 99,5,Wait,1 exten => 99,6,Record,/tmp/voxfile:vox exten => 99,7,Wait,1 exten => 99,8,Playback,/tmp/voxfile (put your own extension). Pcm recording is OK, playback is OK. Adpcm recording is noticeably worse. Adpcm playback is very bad/unusable. Is it just us (all our servers with T400P and TE410P), or it's a common adpcm codec problem? Thank you. Alex Zarubin Webley Systems, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http...

Dear All, I'm using LLVM2.9 for profiling basic block frequency. I'm using following commands. rdpatel55 at ubuntu:~$ llvm-gcc -emit-llvm -O0 -c -o adpcm.bc adpcm.c rdpatel55 at ubuntu:~$ llvm-gcc -emit-llvm -O0 -c -o rawcaudio.bc rawcaudio.c rdpatel55 at ubuntu:~$ llvm-link -o main.bc rawcaudio.bc adpcm.bc rdpatel55 at ubuntu:~$ opt -q -f -insert-edge-profiling -o main.inst main.bc rdpatel55 at ubuntu:~$ lli -fake-argv0 'main.bc' -load /...

...e not got the first digest yet. >First, the results they show are for compression of data >that has already been lossily quantized to fewer bits per >sample, e.g. u-Law and A-Law are logarithmic quantizations >of 16-bit data to 8-bit. I thought the author has two models, A-law and AMI ADPCM, which both he extends. AMI ADPCM starts with 16-bit samples, but I'm not sure if A-law is involved in that process. >Second, the average ratio (assuming the table describes >ratios, since the omitted the units) for 44.1kHz audio >is 3:1. I though they are bits/sample. The text says...

...cker wrote: > Michael Grigoni wrote: > >>Greetings: >> >>It would be nice if Icecast supported RTSP; > > It probably never will > >>however I would >>appreciate any suggestions for a small RTSP/RTP solution to >>encode 8kHz mono audio in GSM or ADPCM and service multiple >>unicast client connections. > > why not use icecast (with adjusted buffers) + speex? are you really that > reliant on very low latency? > Thanks for your reply. Yes, the application is an online remote- controlled HF receiver and a DX'er can't...

In case a system is incapable of fax relay or if it is disabled, one of the easiest and safest options is to go for 40 kbps ADPCM compression (for fax upto 14.4 kbps)..even am new to this problem and the fair bit of seraching which i've done seems to suggest that the standard sloutions are to simply 'bypass' it else compress using ADPCM (40 k for fax upto 14.4 k, 32 k for fax upto 9.6 k and so on), both of which e...