2013-07-10 00:45:36 +00:00
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/*
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2016-02-12 00:05:01 -08:00
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* Copyright 2005 The WebRTC project authors. All Rights Reserved.
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2013-07-10 00:45:36 +00:00
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*
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2016-02-12 00:05:01 -08:00
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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2013-07-10 00:45:36 +00:00
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*/
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// Class to collect statistics from a media channel
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2016-04-26 05:28:11 -07:00
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#ifndef WEBRTC_PC_MEDIAMONITOR_H_
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#define WEBRTC_PC_MEDIAMONITOR_H_
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2013-07-10 00:45:36 +00:00
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Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
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#include "webrtc/media/base/mediachannel.h"
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2017-07-06 19:44:34 +02:00
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#include "webrtc/rtc_base/criticalsection.h"
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#include "webrtc/rtc_base/sigslot.h"
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#include "webrtc/rtc_base/thread.h"
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#include "webrtc/rtc_base/thread_annotations.h"
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2013-07-10 00:45:36 +00:00
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namespace cricket {
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// The base MediaMonitor class, independent of voice and video.
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2014-07-29 17:36:52 +00:00
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class MediaMonitor : public rtc::MessageHandler,
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2013-07-10 00:45:36 +00:00
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public sigslot::has_slots<> {
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public:
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2014-07-29 17:36:52 +00:00
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MediaMonitor(rtc::Thread* worker_thread,
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rtc::Thread* monitor_thread);
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2013-07-10 00:45:36 +00:00
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~MediaMonitor();
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Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 12:23:21 +02:00
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void Start(uint32_t milliseconds);
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2013-07-10 00:45:36 +00:00
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void Stop();
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protected:
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2014-07-29 17:36:52 +00:00
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void OnMessage(rtc::Message *message);
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2013-07-10 00:45:36 +00:00
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void PollMediaChannel();
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virtual void GetStats() = 0;
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virtual void Update() = 0;
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2014-07-29 17:36:52 +00:00
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rtc::CriticalSection crit_;
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rtc::Thread* worker_thread_;
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rtc::Thread* monitor_thread_;
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2013-07-10 00:45:36 +00:00
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bool monitoring_;
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Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 12:23:21 +02:00
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uint32_t rate_;
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2013-07-10 00:45:36 +00:00
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};
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// Templatized MediaMonitor that can deal with different kinds of media.
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template<class MC, class MI>
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class MediaMonitorT : public MediaMonitor {
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public:
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2014-07-29 17:36:52 +00:00
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MediaMonitorT(MC* media_channel, rtc::Thread* worker_thread,
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rtc::Thread* monitor_thread)
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2013-07-10 00:45:36 +00:00
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: MediaMonitor(worker_thread, monitor_thread),
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media_channel_(media_channel) {}
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sigslot::signal2<MC*, const MI&> SignalUpdate;
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protected:
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// These routines assume the crit_ lock is held by the calling thread.
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virtual void GetStats() {
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media_info_.Clear();
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media_channel_->GetStats(&media_info_);
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}
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2014-09-24 07:10:57 +00:00
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virtual void Update() EXCLUSIVE_LOCKS_REQUIRED(crit_) {
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2013-07-10 00:45:36 +00:00
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MI stats(media_info_);
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crit_.Leave();
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SignalUpdate(media_channel_, stats);
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crit_.Enter();
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}
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private:
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MC* media_channel_;
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MI media_info_;
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};
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typedef MediaMonitorT<VoiceMediaChannel, VoiceMediaInfo> VoiceMediaMonitor;
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typedef MediaMonitorT<VideoMediaChannel, VideoMediaInfo> VideoMediaMonitor;
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typedef MediaMonitorT<DataMediaChannel, DataMediaInfo> DataMediaMonitor;
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} // namespace cricket
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2016-04-26 05:28:11 -07:00
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#endif // WEBRTC_PC_MEDIAMONITOR_H_
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