similar to: opus-tools: Fix potential uninitialized access for set-ctl-int

This patch adds a new option "channel-mapping" to opusenc which sets the channel mapping family used by the multistream encoder. Please let me know whether adding this option is worthwhile and whether the help string is okay. I tried to keep it short but accurate. The error message for an unimplemented channel mapping is "Error cannot create encoder: request not implemented".

Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I noticed that there was severe corruption/attenuation on one of the channels(specifically Back/Rear Right). This would appear to be the last channel in the stream. I am attaching an image of the PCM dumps from the original and the one

Here is the modified patch. I added a flag to configure.ac which is set to 0 to disable ambisonics, and 1 to enable it. Right now the implementation simply creates a surround encoder with N uncoupled streams. Thanks, Michael Graczyk -------------- next part -------------- An HTML attachment was scrubbed... URL:

Tim, Would you mind giving me a more specific example of the sort of document that you think this should look like? I'd like to write up something that is somewhat final. On Mon, May 2, 2016 at 9:30 PM, Michael Graczyk <mgraczyk at google.com> wrote: > On Fri, Apr 29, 2016 at 4:32 PM, Timothy B. Terriberry > <tterribe at xiph.org> wrote: > > As a general point,

Change-Id: I16bd4cd990d8ad5888c9e30016218ac25242ecb5 --- src/opus_multistream_encoder.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/src/opus_multistream_encoder.c b/src/opus_multistream_encoder.c index 30cafe1..c07132f 100644 --- a/src/opus_multistream_encoder.c +++ b/src/opus_multistream_encoder.c @@ -747,7 +747,8 @@ static void ambisonics_rate_allocation( if

Hello Tim and others, Thanks for your help explaining this process on IRC. I wrote out a first draft in the RFC xml format. I have attached the xml (labeled as xml.txt so it will appear inline) and the rendered txt files. Please let me know where I can make improvements. I will upload this draft to the IETF datatracker and send it out to codec@ after addressing your comments. -------------- next

Hello, We (Google) have been experimenting with configuration and adjustments to CELT-only Opus that give good results for compressing ambisonic audio signals [1]. Based on our results so far, we would like to use Opus to encode spatial audio. We hope to make it easy/possible to use libopus with other common tools and software modules (ffmpeg/libav in particular). Based on my reading of the

Michael Graczyk wrote: > As for mixing matrices, we are not confident in any choices for setups > beyond stereo. Although there have been papers and studies on Okay, better to give no advice than bad advice. > provide only a stereo downmixing matrix. It looks like this would go > in 5.1.1.5? The matrix should be As a general point, it's too late to add anything to the soon-to-be

On Sat, 28 May 2016 16:21:33 -0700, Michael Graczyk <mgraczyk at google.com> wrote : > Hi Marc, Hi Micheal. > On Sat, May 28, 2016 at 10:44 AM, Marc Lavallée <marc at hacklava.net> > wrote: > > I subscribed because your discussion on the IETF draft ("Ambisonics > > in an Ogg Opus Container") was mentioned on the sursound list. > > Thanks for

Sounds good, I'll make those changes. Thanks for pointing out the make differences. Do you mind if I add an #ifndef, #define to the top of the file for the experiment flag? The code becomes significantly more nasty with #ifdefs everywhere and it would only get worse in subsequent patches. On May 23, 2016 21:28, "Jean-Marc Valin" <jmvalin at jmvalin.ca> wrote: > On

Alright, I've made the changes to the enum and flag usage. I'm don't see the difference with respect to "disabling blocks of code", since the compiler should completely elide the blocks that are conditioned on a constant 0. If there were compiler errors in the experimental code there could be a difference, but I'm hoping that is not the case on any platform! Either way I

Hi Opus list. I subscribed because your discussion on the IETF draft ("Ambisonics in an Ogg Opus Container") was mentioned on the sursound list. I tried Opus for ambisonics more than a year ago. It does works with uncoupled channels (I had to patch the encoder). I don't know what else could be done to optimize support for ambisonics, as I'm not a codec expert. So I think that

Hello Jean-Marc, Thanks for the quick reply and comments. On Thu, May 26, 2016 at 5:41 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > Hi Michael, > > Here's some more minor comments below. As long as you address the two > comments from my previous email (254 -> 2 and the draft name), the draft > is good for submitting as initial version on the IETF website (even

Hi Drew, I think the libopusenc patch is better, but there's still a few issues left: 1) The static MAX_PACKET_BUFFER_SIZE value is still problematic because if you link libopusenc with a new version of libopus that supports higher order projection or just more projection channels for order 3, then you will overflow the buffer. I think what you'd want is a _ope_opus_header_get_size() call

This patch adds top level functions to create an ambisonic multistream encoder. The implementation currently just calls the analogous surround sound functions with channel mapping 255 to create an encoder that bundles uncoupled streams. Forthcoming patches will actually set channel bitrate and other configuration. My main concern is that adding additional

This patch adds a static bitrate allocation for ambisonics encoded with the multistream API. We allocate 4 bits to the mono channel (channel index 0) for every 3 bits allocated to other channels. The allocation is static because it does not depend on the input audio. This patch also forces the underlying encoders to use CELT only for ambisonics. -- Thanks, Michael Graczyk -------------- next

Hi Michael, Any particular reason you assign explicit values to your MappingType enum? If not, I'd rather not depend on the int values. Other than that, I think your patch is good to go. Cheers, Jean-Marc On 05/23/2016 01:27 PM, Michael Graczyk wrote: > Hi Jean-Marc, > > On Sat, May 21, 2016 at 1:58 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: >> 1) I think

Jean-Marc, > Is there any reason you can't just use the generic multi-stream API, > i.e. opus_multistream_encoder_init() and give it the mapping you need? I would like the encoder to make decisions based on the fact that the audio is Ambisonics. For example, the bitrate allocation should not be the same for every channel. The original multistream API does not provide a way to pass in

If you encode with int iByteSizeEncoded = opus_encode(m_enc, m_ShortsInput, (1920/sizeof(short)), m_EncodedBytes, (960*6)); ... the byte size is 120. I thought that an odd value like 359 has to be a fault on my side. I thought it would be a multiple of something. So 359 is really correct??? Thank you for clearing this up. Am 06.10.2013 16:50, schrieb Jean-Marc Valin: > Why

On 19/04/16 12:17 PM, Timothy B. Terriberry wrote: > We should probably start keeping track of the space of invalid TOC > sequences somewhere global so that people don't define conflicting > extensions. I started such a page at https://wiki.xiph.org/OpusExtensions -r -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: