webrtc_m130/webrtc/api/remoteaudiosource.h

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/*
* Copyright 2014 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.
*/
#ifndef WEBRTC_API_REMOTEAUDIOSOURCE_H_
#define WEBRTC_API_REMOTEAUDIOSOURCE_H_
#include <list>
#include <string>
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/notifier.h"
#include "webrtc/audio/audio_sink.h"
#include "webrtc/base/criticalsection.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/audiorenderer.h"
namespace rtc {
struct Message;
class Thread;
} // namespace rtc
namespace webrtc {
class AudioProviderInterface;
// This class implements the audio source used by the remote audio track.
class RemoteAudioSource : public Notifier<AudioSourceInterface> {
public:
// Creates an instance of RemoteAudioSource.
static rtc::scoped_refptr<RemoteAudioSource> Create(
uint32_t ssrc,
AudioProviderInterface* provider);
// MediaSourceInterface implementation.
MediaSourceInterface::SourceState state() const override;
bool remote() const override;
void AddSink(AudioTrackSinkInterface* sink) override;
void RemoveSink(AudioTrackSinkInterface* sink) override;
protected:
RemoteAudioSource();
~RemoteAudioSource() override;
// Post construction initialize where we can do things like save a reference
// to ourselves (need to be fully constructed).
void Initialize(uint32_t ssrc, AudioProviderInterface* provider);
private:
typedef std::list<AudioObserver*> AudioObserverList;
// AudioSourceInterface implementation.
void SetVolume(double volume) override;
void RegisterAudioObserver(AudioObserver* observer) override;
void UnregisterAudioObserver(AudioObserver* observer) override;
class Sink;
void OnData(const AudioSinkInterface::Data& audio);
void OnAudioProviderGone();
class MessageHandler;
void OnMessage(rtc::Message* msg);
AudioObserverList audio_observers_;
rtc::CriticalSection sink_lock_;
std::list<AudioTrackSinkInterface*> sinks_;
rtc::Thread* const main_thread_;
SourceState state_;
};
} // namespace webrtc
#endif // WEBRTC_API_REMOTEAUDIOSOURCE_H_