Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(499)

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

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « talk/app/webrtc/mediastreamobserver.cc ('k') | talk/app/webrtc/mediastreamproxy.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
(Empty)
1 /*
2 * libjingle
3 * Copyright 2012 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_MEDIASTREAMPROVIDER_H_
29 #define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
30
31 #include "webrtc/base/basictypes.h"
32 #include "webrtc/base/scoped_ptr.h"
33 #include "webrtc/media/base/videosinkinterface.h"
34
35 namespace cricket {
36
37 class AudioRenderer;
38 class VideoCapturer;
39 class VideoFrame;
40 class VideoRenderer;
41 struct AudioOptions;
42 struct VideoOptions;
43
44 } // namespace cricket
45
46 namespace webrtc {
47
48 class AudioSinkInterface;
49
50 // TODO(deadbeef): Change the key from an ssrc to a "sender_id" or
51 // "receiver_id" string, which will be the MSID in the short term and MID in
52 // the long term.
53
54 // TODO(deadbeef): These interfaces are effectively just a way for the
55 // RtpSenders/Receivers to get to the BaseChannels. These interfaces should be
56 // refactored away eventually, as the classes converge.
57
58 // This interface is called by AudioRtpSender/Receivers to change the settings
59 // of an audio track connected to certain PeerConnection.
60 class AudioProviderInterface {
61 public:
62 // Enable/disable the audio playout of a remote audio track with |ssrc|.
63 virtual void SetAudioPlayout(uint32_t ssrc, bool enable) = 0;
64 // Enable/disable sending audio on the local audio track with |ssrc|.
65 // When |enable| is true |options| should be applied to the audio track.
66 virtual void SetAudioSend(uint32_t ssrc,
67 bool enable,
68 const cricket::AudioOptions& options,
69 cricket::AudioRenderer* renderer) = 0;
70
71 // Sets the audio playout volume of a remote audio track with |ssrc|.
72 // |volume| is in the range of [0, 10].
73 virtual void SetAudioPlayoutVolume(uint32_t ssrc, double volume) = 0;
74
75 // Allows for setting a direct audio sink for an incoming audio source.
76 // Only one audio sink is supported per ssrc and ownership of the sink is
77 // passed to the provider.
78 virtual void SetRawAudioSink(
79 uint32_t ssrc,
80 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0;
81
82 protected:
83 virtual ~AudioProviderInterface() {}
84 };
85
86 // This interface is called by VideoRtpSender/Receivers to change the settings
87 // of a video track connected to a certain PeerConnection.
88 class VideoProviderInterface {
89 public:
90 virtual bool SetCaptureDevice(uint32_t ssrc,
91 cricket::VideoCapturer* camera) = 0;
92 // Enable/disable the video playout of a remote video track with |ssrc|.
93 virtual void SetVideoPlayout(
94 uint32_t ssrc,
95 bool enable,
96 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0;
97 // Enable sending video on the local video track with |ssrc|.
98 virtual void SetVideoSend(uint32_t ssrc,
99 bool enable,
100 const cricket::VideoOptions* options) = 0;
101
102 protected:
103 virtual ~VideoProviderInterface() {}
104 };
105
106 } // namespace webrtc
107
108 #endif // TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
OLDNEW
« no previous file with comments | « talk/app/webrtc/mediastreamobserver.cc ('k') | talk/app/webrtc/mediastreamproxy.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698