similar to: Compress interleaved multi-channels pcm/wav with opus

Hello everyone, I tried to compress audio with opus recently, which works fine on mono and stereo data . But I want to compress 6 channels pcm with opus, I have not found an interface that compresses multi-channels. 1、Is there a multi-channel compression interface? If so, where is it, and how can I call it? If it doesn't exist, how can I modify it to make it support 6 channels? 3、In

I am working on a project involving the conversion of a Mu-law sound format (Dialogic Mu-law wav format) file into a standard PCM encoded wav file format. Could somebody tell me if this feature is supported in the Vorbis software and if there is any source code available that performs this task. If not, does anybody know of any resources that might provide me with this tools or information.

Speex 1.0.3 uses floats, but the unstable version 1.1.x uses short. Jean-Marc Le ven 16/04/2004 16:04, Kenji Chan a crit : > Im confused about the format of the PCM/WAV data for speex_encode & > speex_decode > > speex_encode(enc_state, input_frame, &bits); > > In the manual, it says input_frame is a (float *) pointing to the > beginning of a speech

Le ven 16/04/2004 à 16:39, Kenji Chan a écrit : > Ok, I'm using 1.1.4, so I should pass signed short(wav data) directly to > speex_encode()? First, I'd like to ak why you want to use 1.1.4. If you don't need the new features (fixed-point or preprocessor), you should probably stick to 1.0.3, which has received more testing. If you choose to use 1.1.4, you need to send shorts to

> You said, it's unstable, yes, I got compiling errors with MsVC6 and MsVC.net > And I modified it a bit to get it compiled. Should I post the errors > here? YES, kindly post the errors that you encountered. - Abhishek <p><p><p>---------- Original Message ----------- From: "Kenji Chan" <adslbqmr@tpg.com.au> To: <speex-dev@xiph.org> Sent:

I am having trouble converting a .flv file uploaded from Adobe Flex / Flash Player 10 to a Red5 server using the speex coder: http://livedocs.adobe.com/flex/3/langref/flash/media/Microphone.html http://jira.red5.org/confluence/display/codecs/Speex+Codec Questions: 1. How do I extract the audio track out of such a .flv file? 2. How do I convert it from Speex to .wav PCM? Thanks.

Thanks. sox was exactly what I was looking for. -----Original Message----- From: volsung@asu.edu [mailto:volsung@asu.edu] Sent: Thursday, February 08, 2001 2:06 PM To: vorbis-dev@xiph.org Subject: Re: [vorbis-dev] Conversion API for computer telephony systems (Dialogic Mu-law wa v format to PCM encoded wav format) On Thu, 8 Feb 2001, Giovanni Sanfelici wrote: > I am working on a project

Ok, I'm using 1.1.4, so I should pass signed short(wav data) directly to speex_encode()? But the samplecode that comes with 1.1.4 shows me to put short in float array, and pass the float array to speex_encode() <p>-----Original Message----- From: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org] On Behalf Of Jean-Marc Valin Sent: Saturday, April 17, 2004 6:25 AM To: speex

I'm confused about the format of the PCM/WAV data for speex_encode & speex_decode speex_encode(enc_state, input_frame, &bits); In the manual, it says "input_frame is a (float *) pointing to the beginning of a speech frame" (for encode) But I've found that in speexenc.c and testenc.c, short* is used instead of float* So, isn't it signed 16 bit samples(if

I thought it would be more update/better/faster, any thing wrong? In the future, would you use short, and forget about float? If so, I use 1.1.4, I wouldn't need to change my code in the future, right? You said, it's unstable, yes, I got compiling errors with MsVC6 and MsVC.net And I modified it a bit to get it compiled. Should I post the errors here? <p>-----Original Message-----

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

This is resolved: apt-get remove libspeex-dev cd ~/src/speex-1.2rc1/ ./configure --prefix=/usr make; make install cd ../ffmpeg ./configure --enable-libspeex make; make install worked; then I was able to decode a Speex .flv file: ~/flvs$ ffmpeg -i SpeexQ6R16Efalse.flv foo.wav FFmpeg version SVN-r17174, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-libspeex

September 18, 2017 8:17 PM, "Adrian Prantl" <aprantl at apple.com> wrote: > Have you tried looking at the --dump-after-all output to see which pass (if any) is adding the > extra line table entries? > -- adrian You mean --print-after-all ? I don't see any way to pass that to lld; The debug info is fine when using llc. Carlo Kok RemObjects Software

> On Sep 18, 2017, at 10:13 PM, Carlo Kok <ck at remobjects.com> wrote: > > September 18, 2017 8:17 PM, "Adrian Prantl" <aprantl at apple.com> wrote: > >> Have you tried looking at the --dump-after-all output to see which pass (if any) is adding the >> extra line table entries? >> -- adrian > > > You mean --print-after-all ? I

Hi all, I'm mixing logical streams at the page level. One stream has vorbis data, the other has my own data. Do I need two ogg_sync_states or just one? Thanks, Martin --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org' containing only the word

Since we construct our error/debug messages via multiple calls to stdio primitives, we are at risk of multiple threads interleaving their output if they try to output at once. Add a mutex to group related outputs into an atomic chunk. Signed-off-by: Eric Blake <eblake@redhat.com> --- src/errors.c | 30 +++++++++++++++++++++++++++++- 1 file changed, 29 insertions(+), 1 deletion(-) diff

Hi Jerry I'm still able to recreate failures in with 2.2.4 when interleaving file creation/writing from W2k and NT4 machines to a Samba server. I orginally reported this in 2.2.2a, also 2.2.3: http://lists.samba.org/pipermail/samba/2001-December/063396.html http://lists.samba.org/pipermail/samba/2002-January/063483.html http://lists.samba.org/pipermail/samba/2002-February/067221.html I sent

On 11/15/05, Erik de Castro Lopo <mle+xiph@mega-nerd.com> wrote: > Hi all, > > The remaining issue to be decided for the OggPCM2 spec is the support > of chunked vs interleaved data. I think interleaved is the obvious choice - that's what most audio applications are used to dealing with, it's what we need to feed to audio hardware in the end usually, etc. Whilst I

Rene Herman wrote: > Why store N-bit in the most significant bits and not least? Doesn't > that mean an application would likely need to shift everything down > again? One advantage of storing in the MSB's is that the relative value remains correct when processed as the larger word size. For instance, a signed 12 bit integer would use 0x400 to represent +50% amplitude. By

NOTE: I have read the FAQ, Verzani's book, Rtips, and googled. For various reasons I don't want to use a density plot when comparing two distributions, I would prefer to have interleaved histograms over the same ranges. In addition to this, I would also like to normalize the two histograms so that both of their max Y values are the same (so I can compare relative distributions within