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Side by Side Diff: webrtc/audio/audio_send_stream.h

Issue 1479023002: Prepare the AudioSendStream to be hooked up to send-side BWE. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Remove incorrect thread check. Created 5 years 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
11 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ 11 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
12 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ 12 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
13 13
14 #include "webrtc/audio_send_stream.h" 14 #include "webrtc/audio_send_stream.h"
15 #include "webrtc/audio_state.h" 15 #include "webrtc/audio_state.h"
16 #include "webrtc/base/thread_checker.h" 16 #include "webrtc/base/thread_checker.h"
17 #include "webrtc/base/scoped_ptr.h" 17 #include "webrtc/base/scoped_ptr.h"
18 18
19 namespace webrtc { 19 namespace webrtc {
20 class CongestionController;
20 class VoiceEngine; 21 class VoiceEngine;
21 22
22 namespace voe { 23 namespace voe {
23 class ChannelProxy; 24 class ChannelProxy;
24 } // namespace voe 25 } // namespace voe
25 26
26 namespace internal { 27 namespace internal {
27 class AudioSendStream final : public webrtc::AudioSendStream { 28 class AudioSendStream final : public webrtc::AudioSendStream {
28 public: 29 public:
29 AudioSendStream(const webrtc::AudioSendStream::Config& config, 30 AudioSendStream(const webrtc::AudioSendStream::Config& config,
30 const rtc::scoped_refptr<webrtc::AudioState>& audio_state); 31 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
32 CongestionController* congestion_controller);
31 ~AudioSendStream() override; 33 ~AudioSendStream() override;
32 34
33 // webrtc::SendStream implementation. 35 // webrtc::SendStream implementation.
34 void Start() override; 36 void Start() override;
35 void Stop() override; 37 void Stop() override;
36 void SignalNetworkState(NetworkState state) override; 38 void SignalNetworkState(NetworkState state) override;
37 bool DeliverRtcp(const uint8_t* packet, size_t length) override; 39 bool DeliverRtcp(const uint8_t* packet, size_t length) override;
38 40
39 // webrtc::AudioSendStream implementation. 41 // webrtc::AudioSendStream implementation.
40 bool SendTelephoneEvent(int payload_type, uint8_t event, 42 bool SendTelephoneEvent(int payload_type, uint8_t event,
41 uint32_t duration_ms) override; 43 uint32_t duration_ms) override;
42 webrtc::AudioSendStream::Stats GetStats() const override; 44 webrtc::AudioSendStream::Stats GetStats() const override;
43 45
44 const webrtc::AudioSendStream::Config& config() const; 46 const webrtc::AudioSendStream::Config& config() const;
45 47
46 private: 48 private:
47 VoiceEngine* voice_engine() const; 49 VoiceEngine* voice_engine() const;
48 50
49 rtc::ThreadChecker thread_checker_; 51 rtc::ThreadChecker thread_checker_;
50 const webrtc::AudioSendStream::Config config_; 52 const webrtc::AudioSendStream::Config config_;
51 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 53 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
52 rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_; 54 rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
53 55
54 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); 56 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
55 }; 57 };
56 } // namespace internal 58 } // namespace internal
57 } // namespace webrtc 59 } // namespace webrtc
58 60
59 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ 61 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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