webrtc_m130/webrtc/pc/mediamonitor.h

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/*
* Copyright 2005 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// Class to collect statistics from a media channel
#ifndef WEBRTC_PC_MEDIAMONITOR_H_
#define WEBRTC_PC_MEDIAMONITOR_H_
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
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/sigslot.h"
#include "webrtc/rtc_base/thread.h"
#include "webrtc/rtc_base/thread_annotations.h"
namespace cricket {
// The base MediaMonitor class, independent of voice and video.
class MediaMonitor : public rtc::MessageHandler,
public sigslot::has_slots<> {
public:
MediaMonitor(rtc::Thread* worker_thread,
rtc::Thread* monitor_thread);
~MediaMonitor();
void Start(uint32_t milliseconds);
void Stop();
protected:
void OnMessage(rtc::Message *message);
void PollMediaChannel();
virtual void GetStats() = 0;
virtual void Update() = 0;
rtc::CriticalSection crit_;
rtc::Thread* worker_thread_;
rtc::Thread* monitor_thread_;
bool monitoring_;
uint32_t rate_;
};
// Templatized MediaMonitor that can deal with different kinds of media.
template<class MC, class MI>
class MediaMonitorT : public MediaMonitor {
public:
MediaMonitorT(MC* media_channel, rtc::Thread* worker_thread,
rtc::Thread* monitor_thread)
: MediaMonitor(worker_thread, monitor_thread),
media_channel_(media_channel) {}
sigslot::signal2<MC*, const MI&> SignalUpdate;
protected:
// These routines assume the crit_ lock is held by the calling thread.
virtual void GetStats() {
media_info_.Clear();
media_channel_->GetStats(&media_info_);
}
virtual void Update() EXCLUSIVE_LOCKS_REQUIRED(crit_) {
MI stats(media_info_);
crit_.Leave();
SignalUpdate(media_channel_, stats);
crit_.Enter();
}
private:
MC* media_channel_;
MI media_info_;
};
typedef MediaMonitorT<VoiceMediaChannel, VoiceMediaInfo> VoiceMediaMonitor;
typedef MediaMonitorT<VideoMediaChannel, VideoMediaInfo> VideoMediaMonitor;
typedef MediaMonitorT<DataMediaChannel, DataMediaInfo> DataMediaMonitor;
} // namespace cricket
#endif // WEBRTC_PC_MEDIAMONITOR_H_