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Removed dummy "mediastreamsignaling.h"

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Review URL: https://codereview.webrtc.org/1460483005

Cr-Commit-Position: refs/heads/master@{#10717}
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perkj 2015-11-19 12:00:55 -08:00 committed by Commit bot
parent 191c1f9d5b
commit 726b1f7a14
9 changed files with 0 additions and 205 deletions
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30
talk/app/webrtc/mediastreamsignaling.cc
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@ -1,30 +0,0 @@
/*
* libjingle
* Copyright 2012 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/mediastreamsignaling.h"
// TODO(deadbeef): Remove this file once Chrome build files don't reference it.

28
talk/app/webrtc/mediastreamsignaling.h
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@ -1,28 +0,0 @@
/*
* libjingle
* Copyright 2012 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
// TODO(deadbeef): Remove this file once Chrome build files don't reference it.

1
talk/app/webrtc/statscollector.h
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@ -36,7 +36,6 @@
#include <vector>
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/statstypes.h"
#include "talk/app/webrtc/webrtcsession.h"

1
talk/app/webrtc/statscollector_unittest.cc
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@ -35,7 +35,6 @@
#include "talk/app/webrtc/peerconnectionfactory.h"
#include "talk/app/webrtc/mediastream.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/mediastreamtrack.h"
#include "talk/app/webrtc/test/fakedatachannelprovider.h"
#include "talk/app/webrtc/videotrack.h"

140
talk/app/webrtc/test/fakemediastreamsignaling.h
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@ -1,140 +0,0 @@
/*
* libjingle
* Copyright 2013 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
#define TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/videotrack.h"
static const char kStream1[] = "stream1";
static const char kVideoTrack1[] = "video1";
static const char kAudioTrack1[] = "audio1";
static const char kStream2[] = "stream2";
static const char kVideoTrack2[] = "video2";
static const char kAudioTrack2[] = "audio2";
class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling,
public webrtc::MediaStreamSignalingObserver {
public:
explicit FakeMediaStreamSignaling(cricket::ChannelManager* channel_manager) :
webrtc::MediaStreamSignaling(rtc::Thread::Current(), this,
channel_manager) {
}
void SendAudioVideoStream1() {
ClearLocalStreams();
AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
}
void SendAudioVideoStream2() {
ClearLocalStreams();
AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
}
void SendAudioVideoStream1And2() {
ClearLocalStreams();
AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
}
void SendNothing() {
ClearLocalStreams();
}
void UseOptionsAudioOnly() {
ClearLocalStreams();
AddLocalStream(CreateStream(kStream2, kAudioTrack2, ""));
}
void UseOptionsVideoOnly() {
ClearLocalStreams();
AddLocalStream(CreateStream(kStream2, "", kVideoTrack2));
}
void ClearLocalStreams() {
while (local_streams()->count() != 0) {
RemoveLocalStream(local_streams()->at(0));
}
}
// Implements MediaStreamSignalingObserver.
virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) {}
virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) {}
virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) {}
virtual void OnAddLocalAudioTrack(webrtc::MediaStreamInterface* stream,
webrtc::AudioTrackInterface* audio_track,
uint32_t ssrc) {}
virtual void OnAddLocalVideoTrack(webrtc::MediaStreamInterface* stream,
webrtc::VideoTrackInterface* video_track,
uint32_t ssrc) {}
virtual void OnAddRemoteAudioTrack(webrtc::MediaStreamInterface* stream,
webrtc::AudioTrackInterface* audio_track,
uint32_t ssrc) {}
virtual void OnAddRemoteVideoTrack(webrtc::MediaStreamInterface* stream,
webrtc::VideoTrackInterface* video_track,
uint32_t ssrc) {}
virtual void OnRemoveRemoteAudioTrack(
webrtc::MediaStreamInterface* stream,
webrtc::AudioTrackInterface* audio_track) {}
virtual void OnRemoveRemoteVideoTrack(
webrtc::MediaStreamInterface* stream,
webrtc::VideoTrackInterface* video_track) {}
virtual void OnRemoveLocalAudioTrack(webrtc::MediaStreamInterface* stream,
webrtc::AudioTrackInterface* audio_track,
uint32_t ssrc) {}
virtual void OnRemoveLocalVideoTrack(
webrtc::MediaStreamInterface* stream,
webrtc::VideoTrackInterface* video_track) {}
virtual void OnRemoveLocalStream(webrtc::MediaStreamInterface* stream) {}
private:
rtc::scoped_refptr<webrtc::MediaStreamInterface> CreateStream(
const std::string& stream_label,
const std::string& audio_track_id,
const std::string& video_track_id) {
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
webrtc::MediaStream::Create(stream_label));
if (!audio_track_id.empty()) {
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
webrtc::AudioTrack::Create(audio_track_id, NULL));
stream->AddTrack(audio_track);
}
if (!video_track_id.empty()) {
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
webrtc::VideoTrack::Create(video_track_id, NULL));
stream->AddTrack(video_track);
}
return stream;
}
};
#endif // TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_

1
talk/app/webrtc/webrtcsession.cc
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@ -36,7 +36,6 @@
#include "talk/app/webrtc/jsepicecandidate.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/mediaconstraintsinterface.h"
#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/sctputils.h"
#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"

1
talk/app/webrtc/webrtcsession_unittest.cc
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@ -33,7 +33,6 @@
#include "talk/app/webrtc/jsepicecandidate.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/peerconnection.h"
#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/sctputils.h"
#include "talk/app/webrtc/streamcollection.h"
#include "talk/app/webrtc/streamcollection.h"

2
talk/libjingle.gyp
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@ -743,8 +743,6 @@
'app/webrtc/mediastreaminterface.h',
'app/webrtc/mediastreamprovider.h',
'app/webrtc/mediastreamproxy.h',
'app/webrtc/mediastreamsignaling.cc',
'app/webrtc/mediastreamsignaling.h',
'app/webrtc/mediastreamtrack.h',
'app/webrtc/mediastreamtrackproxy.h',
'app/webrtc/notifier.h',

1
talk/libjingle_tests.gyp
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@ -215,7 +215,6 @@
'app/webrtc/test/fakeconstraints.h',
'app/webrtc/test/fakedatachannelprovider.h',
'app/webrtc/test/fakedtlsidentitystore.h',
'app/webrtc/test/fakemediastreamsignaling.h',
'app/webrtc/test/fakeperiodicvideocapturer.h',
'app/webrtc/test/fakevideotrackrenderer.h',
'app/webrtc/test/mockpeerconnectionobservers.h',