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Side by Side Diff: talk/media/base/mediaengine.h

Issue 1646253004: Split out dscp option from VideoOptions to new struct MediaChannelOptions. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix accidentally broken combined_audio_video_bwe option. Created 4 years, 10 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2004 Google Inc. 3 * Copyright 2004 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,
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74 virtual bool Init(rtc::Thread* worker_thread) = 0; 74 virtual bool Init(rtc::Thread* worker_thread) = 0;
75 // Shuts down the engine. 75 // Shuts down the engine.
76 virtual void Terminate() = 0; 76 virtual void Terminate() = 0;
77 // TODO(solenberg): Remove once VoE API refactoring is done. 77 // TODO(solenberg): Remove once VoE API refactoring is done.
78 virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0; 78 virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
79 79
80 // MediaChannel creation 80 // MediaChannel creation
81 // Creates a voice media channel. Returns NULL on failure. 81 // Creates a voice media channel. Returns NULL on failure.
82 virtual VoiceMediaChannel* CreateChannel( 82 virtual VoiceMediaChannel* CreateChannel(
83 webrtc::Call* call, 83 webrtc::Call* call,
84 const AudioOptions& options) = 0; 84 const MediaChannelOptions& options,
85 const AudioOptions& audio_options) = 0;
85 // Creates a video media channel, paired with the specified voice channel. 86 // Creates a video media channel, paired with the specified voice channel.
86 // Returns NULL on failure. 87 // Returns NULL on failure.
87 virtual VideoMediaChannel* CreateVideoChannel( 88 virtual VideoMediaChannel* CreateVideoChannel(
88 webrtc::Call* call, 89 webrtc::Call* call,
89 const VideoOptions& options) = 0; 90 const MediaChannelOptions& options,
91 const VideoOptions& video_options) = 0;
90 92
91 // Device configuration 93 // Device configuration
92 // Gets the current speaker volume, as a value between 0 and 255. 94 // Gets the current speaker volume, as a value between 0 and 255.
93 virtual bool GetOutputVolume(int* level) = 0; 95 virtual bool GetOutputVolume(int* level) = 0;
94 // Sets the current speaker volume, as a value between 0 and 255. 96 // Sets the current speaker volume, as a value between 0 and 255.
95 virtual bool SetOutputVolume(int level) = 0; 97 virtual bool SetOutputVolume(int level) = 0;
96 98
97 // Gets the current microphone level, as a value between 0 and 10. 99 // Gets the current microphone level, as a value between 0 and 10.
98 virtual int GetInputLevel() = 0; 100 virtual int GetInputLevel() = 0;
99 101
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149 return true; 151 return true;
150 } 152 }
151 virtual void Terminate() { 153 virtual void Terminate() {
152 voice_.Terminate(); 154 voice_.Terminate();
153 } 155 }
154 156
155 virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const { 157 virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
156 return voice_.GetAudioState(); 158 return voice_.GetAudioState();
157 } 159 }
158 virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call, 160 virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
159 const AudioOptions& options) { 161 const MediaChannelOptions& options,
160 return voice_.CreateChannel(call, options); 162 const AudioOptions& audio_options) {
163 return voice_.CreateChannel(call, options, audio_options);
161 } 164 }
162 virtual VideoMediaChannel* CreateVideoChannel(webrtc::Call* call, 165 virtual VideoMediaChannel* CreateVideoChannel(
163 const VideoOptions& options) { 166 webrtc::Call* call,
164 return video_.CreateChannel(call, options); 167 const MediaChannelOptions& options,
168 const VideoOptions& video_options) {
169 return video_.CreateChannel(call, options, video_options);
165 } 170 }
166 171
167 virtual bool GetOutputVolume(int* level) { 172 virtual bool GetOutputVolume(int* level) {
168 return voice_.GetOutputVolume(level); 173 return voice_.GetOutputVolume(level);
169 } 174 }
170 virtual bool SetOutputVolume(int level) { 175 virtual bool SetOutputVolume(int level) {
171 return voice_.SetOutputVolume(level); 176 return voice_.SetOutputVolume(level);
172 } 177 }
173 178
174 virtual int GetInputLevel() { 179 virtual int GetInputLevel() {
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215 class DataEngineInterface { 220 class DataEngineInterface {
216 public: 221 public:
217 virtual ~DataEngineInterface() {} 222 virtual ~DataEngineInterface() {}
218 virtual DataMediaChannel* CreateChannel(DataChannelType type) = 0; 223 virtual DataMediaChannel* CreateChannel(DataChannelType type) = 0;
219 virtual const std::vector<DataCodec>& data_codecs() = 0; 224 virtual const std::vector<DataCodec>& data_codecs() = 0;
220 }; 225 };
221 226
222 } // namespace cricket 227 } // namespace cricket
223 228
224 #endif // TALK_MEDIA_BASE_MEDIAENGINE_H_ 229 #endif // TALK_MEDIA_BASE_MEDIAENGINE_H_
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