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Side by Side Diff: talk/app/webrtc/mediastreamprovider.h

Issue 1351803002: Exposing RtpSenders and RtpReceivers from PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding some stubs so that Chromium build won't break. Created 5 years, 2 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2012 Google Inc. 3 * Copyright 2012 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 10 matching lines...) Expand all
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 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, 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 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 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */ 26 */
27 27
28 #ifndef TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ 28 #ifndef TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
29 #define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ 29 #define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
30 30
31 #include "webrtc/base/basictypes.h"
32
31 namespace cricket { 33 namespace cricket {
32 34
33 class AudioRenderer; 35 class AudioRenderer;
34 class VideoCapturer; 36 class VideoCapturer;
35 class VideoRenderer; 37 class VideoRenderer;
36 struct AudioOptions; 38 struct AudioOptions;
37 struct VideoOptions; 39 struct VideoOptions;
38 40
39 } // namespace cricket 41 } // namespace cricket
40 42
41 namespace webrtc { 43 namespace webrtc {
42 44
45 // TODO(deadbeef): Change the key from an ssrc to a "sender_id" or
46 // "receiver_id" string, which will be the MSID in the short term and MID in
47 // the long term.
48
49 // TODO(deadbeef): These interfaces are effectively just a way for the
50 // RtpSenders/Receivers to get to the BaseChannels. These interfaces should be
51 // refactored away eventually, as the classes converge.
52
43 // This interface is called by AudioTrackHandler classes in mediastreamhandler.h 53 // This interface is called by AudioTrackHandler classes in mediastreamhandler.h
44 // to change the settings of an audio track connected to certain PeerConnection. 54 // to change the settings of an audio track connected to certain PeerConnection.
45 class AudioProviderInterface { 55 class AudioProviderInterface {
46 public: 56 public:
47 // Enable/disable the audio playout of a remote audio track with |ssrc|. 57 // Enable/disable the audio playout of a remote audio track with |ssrc|.
48 virtual void SetAudioPlayout(uint32 ssrc, bool enable, 58 virtual void SetAudioPlayout(uint32 ssrc, bool enable,
49 cricket::AudioRenderer* renderer) = 0; 59 cricket::AudioRenderer* renderer) = 0;
50 // Enable/disable sending audio on the local audio track with |ssrc|. 60 // Enable/disable sending audio on the local audio track with |ssrc|.
51 // When |enable| is true |options| should be applied to the audio track. 61 // When |enable| is true |options| should be applied to the audio track.
52 virtual void SetAudioSend(uint32 ssrc, bool enable, 62 virtual void SetAudioSend(uint32 ssrc, bool enable,
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75 virtual void SetVideoSend(uint32 ssrc, bool enable, 85 virtual void SetVideoSend(uint32 ssrc, bool enable,
76 const cricket::VideoOptions* options) = 0; 86 const cricket::VideoOptions* options) = 0;
77 87
78 protected: 88 protected:
79 virtual ~VideoProviderInterface() {} 89 virtual ~VideoProviderInterface() {}
80 }; 90 };
81 91
82 } // namespace webrtc 92 } // namespace webrtc
83 93
84 #endif // TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ 94 #endif // TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
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