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

Issue 2705093002: Injectable audio encoders: WebRtcVoiceEngine and company (Closed)
Patch Set: Channel::GetSendCodec asks both its acm and its codec manager. Created 3 years, 7 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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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48 48
49 int id() const { return id_; } 49 int id() const { return id_; }
50 const webrtc::AudioSendStream::Config& GetConfig() const; 50 const webrtc::AudioSendStream::Config& GetConfig() const;
51 void SetStats(const webrtc::AudioSendStream::Stats& stats); 51 void SetStats(const webrtc::AudioSendStream::Stats& stats);
52 TelephoneEvent GetLatestTelephoneEvent() const; 52 TelephoneEvent GetLatestTelephoneEvent() const;
53 bool IsSending() const { return sending_; } 53 bool IsSending() const { return sending_; }
54 bool muted() const { return muted_; } 54 bool muted() const { return muted_; }
55 55
56 private: 56 private:
57 // webrtc::AudioSendStream implementation. 57 // webrtc::AudioSendStream implementation.
58 void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
59
58 void Start() override { sending_ = true; } 60 void Start() override { sending_ = true; }
59 void Stop() override { sending_ = false; } 61 void Stop() override { sending_ = false; }
60 62
61 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, 63 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
62 int duration_ms) override; 64 int duration_ms) override;
63 void SetMuted(bool muted) override; 65 void SetMuted(bool muted) override;
64 webrtc::AudioSendStream::Stats GetStats() const override; 66 webrtc::AudioSendStream::Stats GetStats() const override;
65 67
66 int id_ = -1; 68 int id_ = -1;
67 TelephoneEvent latest_telephone_event_; 69 TelephoneEvent latest_telephone_event_;
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314 316
315 int num_created_send_streams_; 317 int num_created_send_streams_;
316 int num_created_receive_streams_; 318 int num_created_receive_streams_;
317 319
318 int audio_transport_overhead_; 320 int audio_transport_overhead_;
319 int video_transport_overhead_; 321 int video_transport_overhead_;
320 }; 322 };
321 323
322 } // namespace cricket 324 } // namespace cricket
323 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ 325 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_
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