ccb49e79fd
Remove Soundclip handling from libjingle.
...
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51009004
Cr-Commit-Position: refs/heads/master@{#9216}
2015-05-19 09:37:39 +00:00
2013aeced2
Propagating RTT from send-only channel to receive-only channel.
...
This is important for obtaining ntp time at receiver-only channel, which does not have RTT directly.
BUG=3978
TEST=chromium with hangout calls
R=henrika@webrtc.org , mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29989004
Cr-Commit-Position: refs/heads/master@{#9186}
2015-05-13 12:14:36 +00:00
300eeb68f5
Remove VideoEngine interfaces.
...
Removes ViE interfaces, _impl.cc files, managers (such as
ViEChannelManager and ViEInputManager) as well as ViESharedData.
Interfaces necessary to implement observers have been moved to a
corresponding header (such as vie_channel.h).
BUG=1695, 4491
R=mflodman@webrtc.org , solenberg@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55379004
Cr-Commit-Position: refs/heads/master@{#9179}
2015-05-12 14:51:08 +00:00
64dad838e6
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
...
The original change was reverted due to a breakage in the chrome build.
This change includes a fix for this.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49329004
Cr-Commit-Position: refs/heads/master@{#9169}
2015-05-11 10:44:20 +00:00
1f629232d5
Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
...
This reverts commit fd32f35aff8fc28ec084bddc274de284e0422a57.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55369004
Cr-Commit-Position: refs/heads/master@{#9165}
2015-05-10 09:06:20 +00:00
fd32f35aff
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
...
This reverts commit cdb47a4533b7b1e29e803ed6591a68bb1a4f1692.
Contains a tentative fix to the chrome build breakage caused by the
original change.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47139004
Cr-Commit-Position: refs/heads/master@{#9164}
2015-05-10 09:03:00 +00:00
cdb47a4533
Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
...
This reverts commit 208a2294cde839025318f1b3d57559cb0611a4e7.
Breaks the Chrome build.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53399004
Cr-Commit-Position: refs/heads/master@{#9161}
2015-05-08 12:03:46 +00:00
208a2294cd
Adding a new constraint to set NetEq buffer capacity from peerconnection
...
This change makes it possible to set a custom value for the maximum
capacity of the packet buffer in NetEq (the audio jitter buffer). The
default value is 50 packets, but any value can be set with the new
functionality.
R=jmarusic@webrtc.org , mflodman@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50869004
Cr-Commit-Position: refs/heads/master@{#9159}
2015-05-08 10:58:51 +00:00
d3ddc1b69e
Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/.
...
BUG=
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49929004
Cr-Commit-Position: refs/heads/master@{#9156}
2015-05-07 15:07:36 +00:00
aaf8ff2e45
WebRtcVoiceEngine: virtual to override + git cl format.
...
BUG=
R=kwiberg@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54369004
Cr-Commit-Position: refs/heads/master@{#9154}
2015-05-07 14:05:57 +00:00
6179b89e53
Remove unused API on WebRtcVoiceEngine.
...
BUG=1695
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46209004
Cr-Commit-Position: refs/heads/master@{#9153}
2015-05-07 14:01:30 +00:00
4b60c73e74
Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
...
BUG=4574,3109
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49269004
Cr-Commit-Position: refs/heads/master@{#9150}
2015-05-07 12:07:46 +00:00
9478437fde
rtc::Buffer improvements
...
1. Constructors, SetData(), and AppendData() now accept uint8_t*,
int8_t*, and char*. Previously, they accepted void*, meaning that
any kind of pointer was accepted. I think requiring an explicit
cast in cases where the input array isn't already of a byte-sized
type is a better compromise between convenience and safety.
2. data() can now return a uint8_t* instead of a char*, which seems
more appropriate for a byte array, and is harder to mix up with
zero-terminated C strings. data<int8_t>() is also available so
that callers that want that type instead won't have to cast, as
is data<char>() (which remains the default until all existing
callers have been fixed).
3. Constructors, SetData(), and AppendData() now accept arrays
natively, not just decayed to pointers. The advantage of this is
that callers don't have to pass the size separately.
4. There are new constructors that allow setting size and capacity
without initializing the array. Previously, this had to be done
separately after construction.
5. Instead of TransferTo(), Buffer now supports swap(), and move
construction and assignment, and has a Pass() method that works
just like std::move(). (The Pass method is modeled after
scoped_ptr::Pass().)
R=jmarusic@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42989004
Cr-Commit-Position: refs/heads/master@{#9033}
2015-04-20 12:03:00 +00:00
7100dcd317
Adding "usedtx" as Opus codec parameter.
...
This is according to https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
Specifically,
usedtx: specifies if the decoder prefers the use of DTX. values are 1 and 0. If no value is specified, usedtx is assumed to be 0.
BUG=1014
R=juberti@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48499004
Cr-Commit-Position: refs/heads/master@{#8872}
2015-03-27 04:06:35 +00:00
5225dd8180
If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size.
...
BUG=4289
TEST=Manual/Auto Test
R=juberti@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44629004
Cr-Commit-Position: refs/heads/master@{#8863}
2015-03-25 23:39:33 +00:00
bf395c1fc0
Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android
...
If built-in Echo Cancellation is available on a device it is automatically enabled. The reason is that it in most cases performs better than the WebRTC software echo control for mobile. The drawback is that we can not develop, test and rollout the delay agnostic AEC (DA-AEC) on Android as for desktops.
This CL includes
- adding a media constraint to enable/disable DA-AEC.
- automatically turning on echo cancellation if DA-AEC is enabled.
- a fix in the AEC that enables delay estimation when DA-AEC is enabled, but delay metrics is disabled.
- sets the Config struct ReportedDelay, which controls DA-AEC internally in the AEC.
The test code to verify that it works in AppRTCDemo can be found here:
https://webrtc-codereview.appspot.com/50479004/
BUG=4472
TESTED=locally on N7, N6, Android One
R=glaznev@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48699004
Cr-Commit-Position: refs/heads/master@{#8861}
2015-03-25 21:46:10 +00:00
c0bd7be0df
Adding two new stats to VoiceReceiverInfo
...
There have been requests of two new stats namely
speech_expand_rate and secondary_decoded_rate.
BUG=3867
R=henrik.lundin@webrtc.org , henrika@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40789004
Cr-Commit-Position: refs/heads/master@{#8415}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8415 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-18 15:24:39 +00:00
f9b5c1b3d0
Removing CELT.
...
CELT is not supported in WebRTC/Libjingle. There are a few left-over in our code base. They are cleaned up in this CL.
BUG=
R=pbos@webrtc.org , tina.legrand@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36099004
Cr-Commit-Position: refs/heads/master@{#8385}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8385 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 12:37:14 +00:00
8cf9bdb3fa
Remove USE_WEBRTC_DEV_BRANCH.
...
talk/ and webrtc/ are hosted in the same repository and it no longer
makes sense to support building talk/ without the corresponding webrtc/
catalog.
R=bjornv@webrtc.org , juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/39849004
Cr-Commit-Position: refs/heads/master@{#8291}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8291 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 10:17:12 +00:00
aef0779dab
Rewrite ThreadWindows.
...
* Remove "dead" and "alive" variables.
* Remove critical section
* Skip synchronizing with the worker thread to verify startup (no need).
* Remove implementation of SetNotAlive()
* Always set thread name
* Add thread checks for correct usage.
Also added some TODOs for myself for the ThreadWrapper interface.
I'm removing the HasNoMonitorThread test since it is no longer relevant and ends up checking the wrong thing (ProcessThread - a generic thread type) in the wrong way (parsing a debug log) :) I think it served a purpose some years ago, but things have changed since.
BUG=2902
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37069004
Cr-Commit-Position: refs/heads/master@{#8220}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8220 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 15:06:44 +00:00
4ef22d1d29
Setting Opus FEC as default
...
BUG=3986
R=mflodman@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7710 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 09:26:39 +00:00
f85dbce041
Reapply "Advertise G722 as 8 kHz rather than 16 kHz""
...
This reverts r7653 and relands r7645. The reason for the original revert was that G722 disappeared from the SDP offer. This is now fixed. Also, a unit test was updated compared with the original change.
BUG=3951
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7662 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-07 12:25:00 +00:00
dced5d7835
Revert "Advertise G722 as 8 kHz rather than 16 kHz"
...
This reverts r7645.
TBR=pthatcher@webrtc.org
BUG=3951
Review URL: https://webrtc-codereview.appspot.com/24199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7653 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 15:27:43 +00:00
1dcca4028f
Advertise G722 as 8 kHz rather than 16 kHz
...
G722 is a 16 kHz (wideband) speech codec, but a "bug" in the RFC
has it listed as 8 kHz. This means that the codec should be
advertised as 8 kHz in SDP messages. This change fixes that.
R=juberti@google.com
TBR=pthatcher@webrtc.org
BUG=3951
TEST=Verify that the G722 is advertised as a=rtpmap:9 G722/8000, not /16000.
Review URL: https://webrtc-codereview.appspot.com/27879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7645 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 08:55:01 +00:00
2dc6f3154d
Adapting bitrate according to maxplaybackrate for Opus.
...
BUG=
R=mflodman@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 05:33:10 +00:00
c11cc8d947
Revert 7564 "to submit" due to wrong submission
...
> to submit
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7565 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 07:46:47 +00:00
de386bf67b
to submit
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7564 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 07:20:09 +00:00
269fb4bc90
move xmpp and p2p to webrtc
...
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.
BUG=3379
Review URL: https://webrtc-codereview.appspot.com/26999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00
9d446f2e16
(Auto)update libjingle 78296920-> 78342456
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7507 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:22:06 +00:00
28100cb388
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
...
BUG=N/A
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 22:03:39 +00:00
d1ba6d9cbf
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
...
BUG=3379
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27709005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 17:30:28 +00:00
5d639b3ef3
(Auto)update libjingle 75141932-> 75179475
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7129 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 07:57:12 +00:00
b4c7b09c13
(Auto)update libjingle 73927775-> 74032598
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 12:11:58 +00:00
0481f15f02
(Auto)update libjingle 73399579-> 73626167
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6928 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19 14:56:59 +00:00
6b21b71068
(Auto)update libjingle 72205295-> 72320533
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6806 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 15:08:53 +00:00
d4e598d57a
(Auto)update libjingle 72097588-> 72159069
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00
a8d8ad2be6
(Auto)update libjingle 71240799-> 71250251
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6705 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 14:23:08 +00:00
d8524348bb
(Auto)update libjingle 71107853-> 71115715
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6675 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-14 20:05:09 +00:00
3ffa1f917e
(Auto)update libjingle 70422491-> 70424781
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6586 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-02 19:51:26 +00:00
0bb9fac98c
(Auto)update libjingle 70343444-> 70394475
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6581 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-02 11:54:09 +00:00
d8a9069080
(Auto)update libjingle 70340027-> 70343444
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6579 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 19:26:43 +00:00
0d15159b04
(Auto)update libjingle 69634309-> 69640360
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6512 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 19:02:09 +00:00
8563ef448a
(Auto)update libjingle 69587333-> 69588608
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6501 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 00:13:01 +00:00
d27d9ae644
(Auto)update libjingle 69506154-> 69515138
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6488 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 01:56:46 +00:00
ae740dd94c
(Auto)update libjingle 69359922-> 69365993
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6463 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 10:56:41 +00:00
d054bff3b9
(Auto)update libjingle 69292418-> 69293749
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6452 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 14:37:41 +00:00
d159140965
(Auto)update libjingle 69260070-> 69276003
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6439 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 07:49:00 +00:00
7e71b77f8a
(Auto)update libjingle 69102234-> 69116997
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6424 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 01:14:01 +00:00
18dfa8d574
(Auto)update libjingle 69069003-> 69082899
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6417 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 18:11:02 +00:00
b90619c07f
(Auto)update libjingle 69049090-> 69054765
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6412 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 09:19:08 +00:00
