Fixes issues where statistics only was reported for the first stream if
configured with simulcast, and in case of RTX the reported statistics was
depending on the order of the report blocks.
Also fixes issues with multiple report blocks in the SendStatisticsProxy and the
RtcpStatisticsCallback. SendStatisticsProxy is now aware of RTX ssrcs, and the
RTCPReceiver is calling the RtcpStatisticsCallback with the correct SSRCs, and
not only the primary stream SSRC.
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6903 4adac7df-926f-26a2-2b94-8c16560cd09d
Breaks chrome compilation on Mac:
/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
error: no matching constructor for initialization of
'webrtc::StatsReport'
_Tp __x_copy = __x;
^ ~~~
/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::_M_insert_aux' requested here
_M_insert_aux(end(), __x);
^
../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::push_back' requested here
reports.push_back(report1);
^
../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
note: candidate constructor not viable: requires 0 arguments, but 1
was provided
StatsReport() : timestamp(0) {}
> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
>
> This is a reland of r6778 which was reverted due to fyi bots failing.
> I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
>
> R=xians@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/15119004TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6897 4adac7df-926f-26a2-2b94-8c16560cd09d
Some constraints, like kEchoCancellation, kMediaStreamAudioDucking are supported in Chrome but not in Libjingle, if the users set it in mandatory, LocalAudioSource::Initialize() will fail the getUserMedia call.
This patch fixes the problem by fully initializing the LocalAudioSource even though some constraints are not supported in libjingle.
BUT=crbug/398080
TEST=manual test:
var constraints = {audio: { mandatory: { googEchoCancellation: true } }};
getUserMedia(constraints, gotStream, gotStreamFailed);
verify you get a gotStream callback
R=henrika@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6885 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL turns the background noise generation in NetEq off by default. The noise generation used to kick in during long-duration packet losses, when there was no point in extrapolating the latest audio any longer. However, this sometimes produces annoying noise in situations where silence would have been preferable.
With this change, a long packet-loss concealment will be faded out to zeros instead of a low noise.
Reference files are updated where needed.
BUG=3519
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6882 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is only used at four places in iSAC fixed point and the macro have been replaced at those places.
In addition, it is used in a unit test, but throws a warning treated as error (issue3674).
The macro has both MIPS and armv7 optimizations. Removing them impacts only MIPS platforms without DSP ASE. This may cause a very small increase in complexity when using iSAC fix.
The armv7 optimizations are not used anywhere, since specific ones are used inline in iSAC fix.
BUG=3348,3353,3674
TESTED=locally and trybots
R=ljubomir.papuga@gmail.com, tina.legrand@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6871 4adac7df-926f-26a2-2b94-8c16560cd09d
In r6591 a shift macro was removed affecting AECM. In addition to that change a bug was fixed. The fix added a few voice_counts in ApmTest.Process.
This CL updates the reference file, even though it is not used in practice since the test is currently turned off for Android (where AECM is used).
BUG=3672
TESTED=locally
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6868 4adac7df-926f-26a2-2b94-8c16560cd09d
audio_processing did not compile when aec_untrusted_delay_for_testing=1 was set. The constant kDelayDiffOffsetSamples was declared only for Mac when WEBRTC_UNTRUSTED_DELAY was automatically turned on.
Moving the declaration outside the ifdef makes it build with the flag on for any platform.
BUG=3673
TESTED=locally and trybots
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6866 4adac7df-926f-26a2-2b94-8c16560cd09d
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
This is a reland of r6778 which was reverted due to fyi bots failing.
I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6863 4adac7df-926f-26a2-2b94-8c16560cd09d
This change replaces the old NETEQTEST_RTPpacket and
NETEQTEST_DummyRTPpacket with the new test::Packet class. Note that the
Packet class automatically handles "dummy" packets (i.e., packets for
which only the header and a length field was stored to file)
automatically. There is no need to explicitly signal this to the
application any longer. The RTP input file is now handled as a
test::PacketSource object.
Also adding a new ConvertHeader method to the Packet class. This is
needed to extract the header information as an alternative data type.
Finally, some dead code was deleted from rtp_analyze.cc (unrelated to
the reset of this change).
BUG=2692
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6862 4adac7df-926f-26a2-2b94-8c16560cd09d
