[asterisk-dev] Opus and VP8
Lorenzo Miniero
lminiero at gmail.com
Mon Jul 8 08:50:58 CDT 2013
Dear all,
I've just uploaded the patch. I've also added a few comments to the bug for
further discussion. Let me know if anything else is required of me at this
point.
Cheers,
Lorenzo
2013/7/3 Lorenzo Miniero <lminiero at gmail.com>
> 2013/7/1 Matthew Jordan <mjordan at digium.com>
>
>>
>> On Mon, Jul 1, 2013 at 8:20 AM, Lorenzo Miniero <lminiero at gmail.com>wrote:
>>
>>> 2013/7/1 Tzafrir Cohen <tzafrir.cohen at xorcom.com>
>>>
>>>> On Mon, May 27, 2013 at 09:35:21PM +0200, Lorenzo Miniero wrote:
>>>> > 2013/5/27 Tzafrir Cohen <tzafrir.cohen at xorcom.com>
>>>> >
>>>> > > On Mon, May 27, 2013 at 04:09:08PM +0200, Lorenzo Miniero wrote:
>>>> > > > Dear all,
>>>> > > >
>>>> > > > I've just published the patch on github:
>>>> > > >
>>>> > > > https://github.com/meetecho/asterisk-opus
>>>> > > >
>>>> > > > The README should be quite self explainatory, but if you need any
>>>> > > > additional info feel free to ask me.
>>>> > > > Any feedback will be more than welcome!
>>>> > >
>>>> > > $ diffstat asterisk-opus/asterisk_opus+vp8.diff
>>>> > > build_tools/menuselect-deps.in | 1
>>>> > > channels/chan_sip.c | 91 ++++++-
>>>> > > codecs/codec_opus.c | 529
>>>> > > +++++++++++++++++++++++++++++++++++++++++
>>>> > > codecs/ex_opus.h | 35 ++
>>>> > > configure.ac | 3
>>>> > > formats/format_vp8.c | 195 +++++++++++++++
>>>> > > include/asterisk/format.h | 4
>>>> > > main/channel.c | 2
>>>> > > main/format.c | 16 +
>>>> > > main/frame.c | 38 ++
>>>> > > main/rtp_engine.c | 6
>>>> > > makeopts.in | 3
>>>> > > res/res_rtp_asterisk.c | 42 +++
>>>> > > 13 files changed, 960 insertions(+), 5 deletions(-)
>>>> > >
>>>> > > Any problem with including the format parts of this patch into
>>>> Asterisk?
>>>> > > Asterisk has limited support for H.264 video. I don't suppose
>>>> Asterisk
>>>> > > comes with a license for a H.264 video playback (let alone
>>>> encoding).
>>>> > >
>>>> > > At first glance, the following parts of the patch seem to be
>>>> related to
>>>> > > formats, rather than codecs:
>>>> > >
>>>> > > channels/chan_sip.c | 91 ++++++-
>>>> > > formats/format_vp8.c | 195 +++++++++++++++
>>>> > > include/asterisk/format.h | 4
>>>> > > main/channel.c | 2
>>>> > > main/format.c | 16 +
>>>> > > main/rtp_engine.c | 6
>>>> > > res/res_rtp_asterisk.c | 42 +++
>>>> > >
>>>> > > Would a patch / review of those parts by Lorenzo be welcomed?
>>>> > >
>>>> > >
>>>> >
>>>> > Actually all the files have stuff related to both codecs: so the code
>>>> > related to Opus should be stripped from chan_sip, format, rtp_engine,
>>>> etc.
>>>> > first in order to have them refer to VP8 only. The format_vp8.c file
>>>> itself
>>>> > is pretty much a clone of the H.264 one, so if this one's ok I guess
>>>> that
>>>> > one will be too.
>>>>
>>>> In case I was not clear enough: I believe (and I suppose Matt's answer
>>>> clarified it) that those parts of the patch can be merged into Asterisk
>>>> as there's no reasonable fear of abusing any patents with them.
>>>>
>>>> If you agree with me, please follow up with the bug I opened:
>>>> https://issues.asterisk.org/jira/browse/ASTERISK-21981
>>>> and attach the partial patch.
>>>>
>>>> The nice thing about it is that it includes almost all of the patching
>>>> needed to actual Asterisk code (besides the changes to frame.c . I
>>>> wonder if those are actually needed). Apart from that you only have
>>>> codec modules and changes to the build system required to build them.
>>>>
>>>> (if this bug is a duplicate: sorry for the noise. I didn't see any
>>>> progress here)
>>>>
>>>> If this is to be merged before the Asterisk 12 freeze, we don't have
>>>> much time.
>>>>
>>>>
>>>
>>> The code in frame.c is only needed if Asterisk needs to be aware of how
>>> many samples an Opus RTP packet actually includes. Considering the patch
>>> would be targeted to only add passthrough support for the codec, I guess
>>> the changes in frame.c can be safely ignored.
>>>
>>> I think format_vp8.c can be removed as well: in fact, VP8 passthrough
>>> support does not strictly require that file, which is only needed in case
>>> you also want to provide ability to read/write VP8 files (e.g., for
>>> announcements), something that code doesn't currently provide anyway (it's
>>> just a copy of format_h264.c acting as a placeholder).
>>>
>>> I'm out of the lab today so I won't be able to upload the partial patch
>>> until tomorrow, though, is it OK anyway considering the Asterisk 12 freeze
>>> deadline you mentioned?
>>>
>>>
>> I'd put up for review all parts of the code that do not perform
>> transcoding. If additional work in frame.c is needed to know how many
>> samples *any* RTP packet contains, then I'd include that as well (the fact
>> that some codecs support varying sample rates is not strictly unique to any
>> one codec). The same could be true for some portion of Asterisk that reacts
>> to changes in sample rates. Infrastructure support is entirely appropriate
>> for Asterisk 12; the easier it makes it for someone to potentially add
>> future enhancements to Asterisk the better.
>>
>> I'd consider limited support of pass-through for VP8 to be a candidate as
>> well.
>>
>> When you have those portions ready, attach them to the issue Tzafrir has
>> made. Feature freeze for Asterisk 12 is July 31st, but all that means is
>> that the patch has to be on the issue and posted to Review Board. We can
>> help get that second step going once it is on the issue.
>>
>> Thanks,
>>
>> Matt
>>
>>
>
> I prepared the patch but it seems I can't upload it yet. I had to sign the
> agreement again, even though I thought I already had it done a long time
> ago (but this was before Jira, so something may have changed in the
> meanwhile). As soon as I'm accepted I'll post the patch on the issue page.
>
> Lorenzo
>
>
>> --
>> Matthew Jordan
>> Digium, Inc. | Engineering Manager
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> Check us out at: http://digium.com & http://asterisk.org
>>
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