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

Issue 2248713003: Revert of Add task queue to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@move_getpadding
Patch Set: Created 4 years, 4 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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26 namespace voe { 26 namespace voe {
27 class ChannelProxy; 27 class ChannelProxy;
28 } // namespace voe 28 } // namespace voe
29 29
30 namespace internal { 30 namespace internal {
31 class AudioSendStream final : public webrtc::AudioSendStream, 31 class AudioSendStream final : public webrtc::AudioSendStream,
32 public webrtc::BitrateAllocatorObserver { 32 public webrtc::BitrateAllocatorObserver {
33 public: 33 public:
34 AudioSendStream(const webrtc::AudioSendStream::Config& config, 34 AudioSendStream(const webrtc::AudioSendStream::Config& config,
35 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, 35 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
36 rtc::TaskQueue* worker_queue,
37 CongestionController* congestion_controller, 36 CongestionController* congestion_controller,
38 BitrateAllocator* bitrate_allocator); 37 BitrateAllocator* bitrate_allocator);
39 ~AudioSendStream() override; 38 ~AudioSendStream() override;
40 39
41 // webrtc::AudioSendStream implementation. 40 // webrtc::AudioSendStream implementation.
42 void Start() override; 41 void Start() override;
43 void Stop() override; 42 void Stop() override;
44 bool SendTelephoneEvent(int payload_type, int event, 43 bool SendTelephoneEvent(int payload_type, int event,
45 int duration_ms) override; 44 int duration_ms) override;
46 void SetMuted(bool muted) override; 45 void SetMuted(bool muted) override;
47 webrtc::AudioSendStream::Stats GetStats() const override; 46 webrtc::AudioSendStream::Stats GetStats() const override;
48 47
49 void SignalNetworkState(NetworkState state); 48 void SignalNetworkState(NetworkState state);
50 bool DeliverRtcp(const uint8_t* packet, size_t length); 49 bool DeliverRtcp(const uint8_t* packet, size_t length);
51 50
52 // Implements BitrateAllocatorObserver. 51 // Implements BitrateAllocatorObserver.
53 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, 52 uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
54 uint8_t fraction_loss, 53 uint8_t fraction_loss,
55 int64_t rtt) override; 54 int64_t rtt) override;
56 55
57 const webrtc::AudioSendStream::Config& config() const; 56 const webrtc::AudioSendStream::Config& config() const;
58 57
59 private: 58 private:
60 VoiceEngine* voice_engine() const; 59 VoiceEngine* voice_engine() const;
61 60
62 rtc::ThreadChecker thread_checker_; 61 rtc::ThreadChecker thread_checker_;
63 rtc::TaskQueue* worker_queue_;
64 const webrtc::AudioSendStream::Config config_; 62 const webrtc::AudioSendStream::Config config_;
65 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 63 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
66 std::unique_ptr<voe::ChannelProxy> channel_proxy_; 64 std::unique_ptr<voe::ChannelProxy> channel_proxy_;
67 65
68 BitrateAllocator* const bitrate_allocator_; 66 BitrateAllocator* const bitrate_allocator_;
69 67
70 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); 68 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
71 }; 69 };
72 } // namespace internal 70 } // namespace internal
73 } // namespace webrtc 71 } // namespace webrtc
74 72
75 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ 73 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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