talk/app/webrtc/test/fakemediastreamsignaling.h - Issue 1413713003: Adding the ability to create an RtpSender without a track.

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Issue 1413713003: Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Adding some unit tests for new methods on the sender. Created 5 years, 2 months ago

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1 /*
Taylor Brandstetter 2015/10/19 23:51:34 I could have deleted this in the MediaStreamSignal I could have deleted this in the MediaStreamSignaling removal CL, I just forgot.
2 * libjingle

3 * Copyright 2013 Google Inc.

4 *

5 * Redistribution and use in source and binary forms, with or without

6 * modification, are permitted provided that the following conditions are met:

7 *

8 * 1. Redistributions of source code must retain the above copyright notice,

9 * this list of conditions and the following disclaimer.

10 * 2. Redistributions in binary form must reproduce the above copyright notice,

11 * this list of conditions and the following disclaimer in the documentation

12 * and/or other materials provided with the distribution.

13 * 3. The name of the author may not be used to endorse or promote products

14 * derived from this software without specific prior written permission.

15 *

16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED

17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF

18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO

19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,

20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,

21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;

22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,

23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR

24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF

25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

26 */

27

28 #ifndef TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_

29 #define TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_

30

31 #include "talk/app/webrtc/audiotrack.h"

32 #include "talk/app/webrtc/mediastreamsignaling.h"

33 #include "talk/app/webrtc/videotrack.h"

34

35 static const char kStream1[] = "stream1";

36 static const char kVideoTrack1[] = "video1";

37 static const char kAudioTrack1[] = "audio1";

38

39 static const char kStream2[] = "stream2";

40 static const char kVideoTrack2[] = "video2";

41 static const char kAudioTrack2[] = "audio2";

42

43 class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling,

44 public webrtc::MediaStreamSignalingObserver {

45 public:

46 explicit FakeMediaStreamSignaling(cricket::ChannelManager* channel_manager) :

47 webrtc::MediaStreamSignaling(rtc::Thread::Current(), this,

48 channel_manager) {

49 }

50

51 void SendAudioVideoStream1() {

52 ClearLocalStreams();

53 AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));

54 }

55

56 void SendAudioVideoStream2() {

57 ClearLocalStreams();

58 AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));

59 }

60

61 void SendAudioVideoStream1And2() {

62 ClearLocalStreams();

63 AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));

64 AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));

65 }

66

67 void SendNothing() {

68 ClearLocalStreams();

69 }

70

71 void UseOptionsAudioOnly() {

72 ClearLocalStreams();

73 AddLocalStream(CreateStream(kStream2, kAudioTrack2, ""));

74 }

75

76 void UseOptionsVideoOnly() {

77 ClearLocalStreams();

78 AddLocalStream(CreateStream(kStream2, "", kVideoTrack2));

79 }

80

81 void ClearLocalStreams() {

82 while (local_streams()->count() != 0) {

83 RemoveLocalStream(local_streams()->at(0));

84 }

85 }

86

87 // Implements MediaStreamSignalingObserver.

88 virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) {}

89 virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) {}

90 virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) {}

91 virtual void OnAddLocalAudioTrack(webrtc::MediaStreamInterface* stream,

92 webrtc::AudioTrackInterface* audio_track,

93 uint32_t ssrc) {}

94 virtual void OnAddLocalVideoTrack(webrtc::MediaStreamInterface* stream,

95 webrtc::VideoTrackInterface* video_track,

96 uint32_t ssrc) {}

97 virtual void OnAddRemoteAudioTrack(webrtc::MediaStreamInterface* stream,

98 webrtc::AudioTrackInterface* audio_track,

99 uint32_t ssrc) {}

100 virtual void OnAddRemoteVideoTrack(webrtc::MediaStreamInterface* stream,

101 webrtc::VideoTrackInterface* video_track,

102 uint32_t ssrc) {}

103 virtual void OnRemoveRemoteAudioTrack(

104 webrtc::MediaStreamInterface* stream,

105 webrtc::AudioTrackInterface* audio_track) {}

106 virtual void OnRemoveRemoteVideoTrack(

107 webrtc::MediaStreamInterface* stream,

108 webrtc::VideoTrackInterface* video_track) {}

109 virtual void OnRemoveLocalAudioTrack(webrtc::MediaStreamInterface* stream,

110 webrtc::AudioTrackInterface* audio_track,

111 uint32_t ssrc) {}

112 virtual void OnRemoveLocalVideoTrack(

113 webrtc::MediaStreamInterface* stream,

114 webrtc::VideoTrackInterface* video_track) {}

115 virtual void OnRemoveLocalStream(webrtc::MediaStreamInterface* stream) {}

116

117 private:

118 rtc::scoped_refptr<webrtc::MediaStreamInterface> CreateStream(

119 const std::string& stream_label,

120 const std::string& audio_track_id,

121 const std::string& video_track_id) {

122 rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(

123 webrtc::MediaStream::Create(stream_label));

124

125 if (!audio_track_id.empty()) {

126 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(

127 webrtc::AudioTrack::Create(audio_track_id, NULL));

128 stream->AddTrack(audio_track);

129 }

130

131 if (!video_track_id.empty()) {

132 rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(

133 webrtc::VideoTrack::Create(video_track_id, NULL));

134 stream->AddTrack(video_track);

135 }

136 return stream;

137 }

138 };

139

140 #endif // TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_

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« talk/app/webrtc/rtpsenderreceiver_unittest.cc ('K') | « talk/app/webrtc/rtpsenderreceiver_unittest.cc ('k') | talk/app/webrtc/webrtcsession.cc » ('j') | webrtc/base/helpers.h » ('J')