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Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.h

Issue 1670153003: Introduce struct MediaConfig, with construction-time settings. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed test nit; use reference. Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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45 45
46 WebRtcVoiceEngine(); 46 WebRtcVoiceEngine();
47 // Dependency injection for testing. 47 // Dependency injection for testing.
48 explicit WebRtcVoiceEngine(VoEWrapper* voe_wrapper); 48 explicit WebRtcVoiceEngine(VoEWrapper* voe_wrapper);
49 ~WebRtcVoiceEngine(); 49 ~WebRtcVoiceEngine();
50 bool Init(rtc::Thread* worker_thread); 50 bool Init(rtc::Thread* worker_thread);
51 void Terminate(); 51 void Terminate();
52 52
53 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const; 53 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
54 VoiceMediaChannel* CreateChannel(webrtc::Call* call, 54 VoiceMediaChannel* CreateChannel(webrtc::Call* call,
55 const MediaConfig& config,
55 const AudioOptions& options); 56 const AudioOptions& options);
56 57
57 bool GetOutputVolume(int* level); 58 bool GetOutputVolume(int* level);
58 bool SetOutputVolume(int level); 59 bool SetOutputVolume(int level);
59 int GetInputLevel(); 60 int GetInputLevel();
60 61
61 const std::vector<AudioCodec>& codecs(); 62 const std::vector<AudioCodec>& codecs();
62 RtpCapabilities GetCapabilities() const; 63 RtpCapabilities GetCapabilities() const;
63 64
64 // For tracking WebRtc channels. Needed because we have to pause them 65 // For tracking WebRtc channels. Needed because we have to pause them
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133 134
134 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcVoiceEngine); 135 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcVoiceEngine);
135 }; 136 };
136 137
137 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses 138 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
138 // WebRtc Voice Engine. 139 // WebRtc Voice Engine.
139 class WebRtcVoiceMediaChannel final : public VoiceMediaChannel, 140 class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
140 public webrtc::Transport { 141 public webrtc::Transport {
141 public: 142 public:
142 WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine, 143 WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
144 const MediaConfig& config,
143 const AudioOptions& options, 145 const AudioOptions& options,
144 webrtc::Call* call); 146 webrtc::Call* call);
145 ~WebRtcVoiceMediaChannel() override; 147 ~WebRtcVoiceMediaChannel() override;
146 148
147 const AudioOptions& options() const { return options_; } 149 const AudioOptions& options() const { return options_; }
148 150
151 rtc::DiffServCodePoint PreferredDscp() const override;
152
149 bool SetSendParameters(const AudioSendParameters& params) override; 153 bool SetSendParameters(const AudioSendParameters& params) override;
150 bool SetRecvParameters(const AudioRecvParameters& params) override; 154 bool SetRecvParameters(const AudioRecvParameters& params) override;
151 bool SetPlayout(bool playout) override; 155 bool SetPlayout(bool playout) override;
152 bool PausePlayout(); 156 bool PausePlayout();
153 bool ResumePlayout(); 157 bool ResumePlayout();
154 bool SetSend(SendFlags send) override; 158 bool SetSend(SendFlags send) override;
155 bool PauseSend(); 159 bool PauseSend();
156 bool ResumeSend(); 160 bool ResumeSend();
157 bool SetAudioSend(uint32_t ssrc, 161 bool SetAudioSend(uint32_t ssrc,
158 bool enable, 162 bool enable,
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266 270
267 class WebRtcAudioReceiveStream; 271 class WebRtcAudioReceiveStream;
268 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; 272 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
269 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 273 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
270 274
271 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); 275 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
272 }; 276 };
273 } // namespace cricket 277 } // namespace cricket
274 278
275 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 279 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
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