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

Issue 1324853003: Remove MediaChannel::GetOptions(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 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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278 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses 278 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
279 // WebRtc Voice Engine. 279 // WebRtc Voice Engine.
280 class WebRtcVoiceMediaChannel : public VoiceMediaChannel, 280 class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
281 public webrtc::Transport { 281 public webrtc::Transport {
282 public: 282 public:
283 explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine); 283 explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine);
284 ~WebRtcVoiceMediaChannel() override; 284 ~WebRtcVoiceMediaChannel() override;
285 285
286 int voe_channel() const { return voe_channel_; } 286 int voe_channel() const { return voe_channel_; }
287 bool valid() const { return voe_channel_ != -1; } 287 bool valid() const { return voe_channel_ != -1; }
288 const AudioOptions& options() const { return options_; }
288 289
289 bool SetSendParameters(const AudioSendParameters& params) override; 290 bool SetSendParameters(const AudioSendParameters& params) override;
290 bool SetRecvParameters(const AudioRecvParameters& params) override; 291 bool SetRecvParameters(const AudioRecvParameters& params) override;
291 bool SetOptions(const AudioOptions& options) override; 292 bool SetOptions(const AudioOptions& options) override;
292 bool GetOptions(AudioOptions* options) const override {
293 *options = options_;
294 return true;
295 }
296 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) override; 293 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) override;
297 bool SetSendCodecs(const std::vector<AudioCodec>& codecs) override; 294 bool SetSendCodecs(const std::vector<AudioCodec>& codecs) override;
298 bool SetRecvRtpHeaderExtensions( 295 bool SetRecvRtpHeaderExtensions(
299 const std::vector<RtpHeaderExtension>& extensions) override; 296 const std::vector<RtpHeaderExtension>& extensions) override;
300 bool SetSendRtpHeaderExtensions( 297 bool SetSendRtpHeaderExtensions(
301 const std::vector<RtpHeaderExtension>& extensions) override; 298 const std::vector<RtpHeaderExtension>& extensions) override;
302 bool SetPlayout(bool playout) override; 299 bool SetPlayout(bool playout) override;
303 bool PausePlayout(); 300 bool PausePlayout();
304 bool ResumePlayout(); 301 bool ResumePlayout();
305 bool SetSend(SendFlags send) override; 302 bool SetSend(SendFlags send) override;
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454 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 451 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
455 452
456 // Do not lock this on the VoE media processor thread; potential for deadlock 453 // Do not lock this on the VoE media processor thread; potential for deadlock
457 // exists. 454 // exists.
458 mutable rtc::CriticalSection receive_channels_cs_; 455 mutable rtc::CriticalSection receive_channels_cs_;
459 }; 456 };
460 457
461 } // namespace cricket 458 } // namespace cricket
462 459
463 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ 460 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
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