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webrtc_m130/webrtc/pc
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zhihuang f816493c4f Add media related stats (audio level etc.) to unsignaled streams. 
The media related stats wasn't working for unsignaled stream because there
is no mapping between the receiver_info and unsignaled tracks.

This CL fixes the issue by adding some special logic to the TrackMediaInfoMap
which would create the mapping.

BUG=b/37836881
BUG=webrtc:7685

TBR=deadbeef@webrtc.org

Review-Url: https://codereview.webrtc.org/2883943003
Cr-Commit-Position: refs/heads/master@{#18217}
2017-05-19 20:09:47 +00:00
..
test
Revert of Removes usage of native base::android::GetApplicationContext() (patchset #6 id:120001 of https://codereview.webrtc.org/2888093004/ )
2017-05-18 13:44:20 +00:00
audiomonitor.cc
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
audiomonitor.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
audiotrack.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
audiotrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
BUILD.gn
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
bundlefilter_unittest.cc
…
bundlefilter.cc
…
bundlefilter.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
channel_unittest.cc
Move ready to send logic from BaseChannel to RtpTransport.
2017-04-24 23:54:35 +00:00
channel.cc
Move ready to send logic from BaseChannel to RtpTransport.
2017-04-24 23:54:35 +00:00
channel.h
Move ready to send logic from BaseChannel to RtpTransport.
2017-04-24 23:54:35 +00:00
channelmanager_unittest.cc
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
channelmanager.cc
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
channelmanager.h
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
currentspeakermonitor_unittest.cc
Disable flaky tests on iOS
2017-02-27 06:10:14 +00:00
currentspeakermonitor.cc
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
currentspeakermonitor.h
Reland of place basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2603203003/ )
2017-01-02 16:42:32 +00:00
datachannel_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
datachannel.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
datachannel.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
DEPS
Injectable audio encoders: BuiltinAudioEncoderFactory
2017-04-06 17:03:21 +00:00
dtmfsender_unittest.cc
Change minimum DTMF event duration to be 40 milliseconds
2017-03-28 18:18:32 +00:00
dtmfsender.cc
Change minimum DTMF event duration to be 40 milliseconds
2017-03-28 18:18:32 +00:00
dtmfsender.h
Move DTMF sender to RtpSender (as opposed to WebRtcSession).
2017-02-02 04:27:00 +00:00
externalhmac.cc
Remove HAVE_SRTP define and unmaintained code.
2017-03-06 19:32:22 +00:00
externalhmac.h
Remove HAVE_SRTP define and unmaintained code.
2017-03-06 19:32:22 +00:00
iceserverparsing_unittest.cc
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
iceserverparsing.cc
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
iceserverparsing.h
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
jsepicecandidate.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
jsepsessiondescription_unittest.cc
Parse the connection data in SDP (c= line).
2017-03-21 18:04:53 +00:00
jsepsessiondescription.cc
Fix the fuzz test.
2017-03-23 17:32:12 +00:00
localaudiosource_unittest.cc
Remove PC factory options param from LocalAudioSource::Create.
2017-02-11 05:26:48 +00:00
localaudiosource.cc
Remove PC factory options param from LocalAudioSource::Create.
2017-02-11 05:26:48 +00:00
localaudiosource.h
Remove PC factory options param from LocalAudioSource::Create.
2017-02-11 05:26:48 +00:00
mediaconstraintsinterface_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediamonitor.cc
Delete unneeded includes of base/common.h.
2017-01-25 09:47:24 +00:00
mediamonitor.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
mediasession_unittest.cc
Add "ice-option:trickle" to generated offers/answers.
2017-04-21 09:41:29 +00:00
mediasession.cc
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
mediasession.h
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
mediastream_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastream.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastream.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamobserver.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamobserver.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamtrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
OWNERS
Make hbos and hta rtcstats* OWNERS of webrtc/pc, not webrtc/api.
2017-02-28 14:46:44 +00:00
peerconnection_integrationtest.cc
Add media related stats (audio level etc.) to unsignaled streams.
2017-05-19 20:09:47 +00:00
peerconnection.cc
Initialize PeerConnection members in declaration order and destroy them in reverse order.
2017-05-13 06:37:18 +00:00
peerconnection.h
Initialize PeerConnection members in declaration order and destroy them in reverse order.
2017-05-13 06:37:18 +00:00
peerconnectionendtoend_unittest.cc
Remove temporary include of builtin_audio_encoder_factory.h.
2017-05-17 11:39:36 +00:00
peerconnectionfactory_unittest.cc
Delete deprecated PeerConnection methods, and corresponding using declarations.
2017-03-08 14:59:45 +00:00
peerconnectionfactory.cc
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
peerconnectionfactory.h
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
peerconnectioninterface_unittest.cc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
proxy_unittest.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
quicdatachannel_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatachannel.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatachannel.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatatransport_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatatransport.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatatransport.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
remoteaudiosource.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
remoteaudiosource.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
rtcpmuxfilter_unittest.cc
Some cleanup in BaseChannel RTCP mux code.
2016-08-22 23:00:37 +00:00
rtcpmuxfilter.cc
Some cleanup in BaseChannel RTCP mux code.
2016-08-22 23:00:37 +00:00
rtcpmuxfilter.h
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
rtcstats_integrationtest.cc
Remove temporary include of builtin_audio_encoder_factory.h.
2017-05-17 11:39:36 +00:00
rtcstatscollector_unittest.cc
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
rtcstatscollector.cc
RTCStatsCollector: Remove closed channels from opened set.
2017-03-20 10:14:14 +00:00
rtcstatscollector.h
Move trackmediainfomap files from api/ to pc/.
2017-02-12 03:07:31 +00:00
rtpreceiver.cc
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
2017-04-10 14:39:05 +00:00
rtpreceiver.h
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
2017-04-10 14:39:05 +00:00
rtpsender.cc
Protect APM in webkit builds.
2017-02-03 14:37:05 +00:00
rtpsender.h
Move DTMF sender to RtpSender (as opposed to WebRtcSession).
2017-02-02 04:27:00 +00:00
rtpsenderreceiver_unittest.cc
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
rtptransport_unittest.cc
Move ready to send logic from BaseChannel to RtpTransport.
2017-04-24 23:54:35 +00:00
rtptransport.cc
Move ready to send logic from BaseChannel to RtpTransport.
2017-04-24 23:54:35 +00:00
rtptransport.h
Move ready to send logic from BaseChannel to RtpTransport.
2017-04-24 23:54:35 +00:00
sctputils_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
sctputils.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
sctputils.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
srtpfilter_unittest.cc
Reland: Improve testing of SRTP external auth code paths.
2017-03-06 08:14:10 +00:00
srtpfilter.cc
Remove HAVE_SRTP define and unmaintained code.
2017-03-06 19:32:22 +00:00
srtpfilter.h
Reland: Improve testing of SRTP external auth code paths.
2017-03-06 08:14:10 +00:00
statscollector_unittest.cc
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
statscollector.cc
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
statscollector.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
streamcollection.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
trackmediainfomap_unittest.cc
TrackMediaInfoMap: Allow same SSRC for send and receive side.
2017-02-20 10:05:13 +00:00
trackmediainfomap.cc
Add media related stats (audio level etc.) to unsignaled streams.
2017-05-19 20:09:47 +00:00
trackmediainfomap.h
Move trackmediainfomap files from api/ to pc/.
2017-02-12 03:07:31 +00:00
videocapturertracksource_unittest.cc
Switching some interfaces to use std::unique_ptr<>.
2017-02-11 04:13:37 +00:00
videocapturertracksource.cc
Fix compilation issues of std::unique_ptr
2017-04-09 16:09:06 +00:00
videocapturertracksource.h
Fix occasional race in VideoCapturerTrackSource seen by memcheck bot.
2017-02-25 21:37:59 +00:00
videotrack_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotrack.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotracksource.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotracksource.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
voicechannel.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
webrtcsdp_unittest.cc
Fix webrtcsdp_unittest.
2017-05-05 19:04:36 +00:00
webrtcsdp.cc
Parse the connection data in SDP (c= line).
2017-03-21 18:04:53 +00:00
webrtcsdp.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
webrtcsession_unittest.cc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
webrtcsession.cc
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
webrtcsession.h
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
webrtcsessiondescriptionfactory.cc
Fix comment about remote restart being requested in createOffer
2017-04-25 19:57:14 +00:00
webrtcsessiondescriptionfactory.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00