2014-02-13 23:18:49 +00:00
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/*
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2016-02-10 07:54:43 -08:00
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* Copyright 2014 The WebRTC project authors. All Rights Reserved.
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2014-02-13 23:18:49 +00:00
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*
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2016-02-10 07:54:43 -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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2014-02-13 23:18:49 +00:00
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*/
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2016-02-10 10:53:12 +01:00
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#ifndef WEBRTC_API_REMOTEAUDIOSOURCE_H_
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#define WEBRTC_API_REMOTEAUDIOSOURCE_H_
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2014-02-13 23:18:49 +00:00
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#include <list>
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2015-12-12 01:37:01 +01:00
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#include <string>
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/notifier.h"
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2015-12-12 01:37:01 +01:00
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#include "webrtc/audio/audio_sink.h"
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#include "webrtc/base/criticalsection.h"
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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/audiorenderer.h"
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2015-12-12 01:37:01 +01:00
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namespace rtc {
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struct Message;
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class Thread;
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} // namespace rtc
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namespace webrtc {
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class AudioProviderInterface;
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// This class implements the audio source used by the remote audio track.
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class RemoteAudioSource : public Notifier<AudioSourceInterface> {
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public:
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// Creates an instance of RemoteAudioSource.
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static rtc::scoped_refptr<RemoteAudioSource> Create(
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uint32_t ssrc,
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AudioProviderInterface* provider);
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// MediaSourceInterface implementation.
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MediaSourceInterface::SourceState state() const override;
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bool remote() const override;
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void AddSink(AudioTrackSinkInterface* sink) override;
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void RemoveSink(AudioTrackSinkInterface* sink) override;
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2014-02-13 23:18:49 +00:00
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protected:
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RemoteAudioSource();
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~RemoteAudioSource() override;
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// Post construction initialize where we can do things like save a reference
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// to ourselves (need to be fully constructed).
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void Initialize(uint32_t ssrc, AudioProviderInterface* provider);
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private:
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typedef std::list<AudioObserver*> AudioObserverList;
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// AudioSourceInterface implementation.
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2015-03-04 12:58:35 +00:00
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void SetVolume(double volume) override;
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void RegisterAudioObserver(AudioObserver* observer) override;
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void UnregisterAudioObserver(AudioObserver* observer) override;
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class Sink;
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void OnData(const AudioSinkInterface::Data& audio);
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void OnAudioProviderGone();
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class MessageHandler;
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void OnMessage(rtc::Message* msg);
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AudioObserverList audio_observers_;
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rtc::CriticalSection sink_lock_;
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std::list<AudioTrackSinkInterface*> sinks_;
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rtc::Thread* const main_thread_;
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SourceState state_;
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};
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} // namespace webrtc
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2016-02-10 10:53:12 +01:00
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#endif // WEBRTC_API_REMOTEAUDIOSOURCE_H_
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