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

Issue 2530383002: Reland "Update rtt on audio only calls". (Closed)
Patch Set: Fix for memory access error in ModuleRtpRtcpImpl::Process. Created 4 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 <memory> 14 #include <memory>
15 15
16 #include "webrtc/api/call/audio_send_stream.h" 16 #include "webrtc/api/call/audio_send_stream.h"
17 #include "webrtc/api/call/audio_state.h" 17 #include "webrtc/api/call/audio_state.h"
18 #include "webrtc/base/constructormagic.h" 18 #include "webrtc/base/constructormagic.h"
19 #include "webrtc/base/thread_checker.h" 19 #include "webrtc/base/thread_checker.h"
20 #include "webrtc/call/bitrate_allocator.h" 20 #include "webrtc/call/bitrate_allocator.h"
21 21
22 namespace webrtc { 22 namespace webrtc {
23 class CongestionController; 23 class CongestionController;
24 class VoiceEngine; 24 class VoiceEngine;
25 class RtcEventLog; 25 class RtcEventLog;
26 class RtcpRttStats;
26 27
27 namespace voe { 28 namespace voe {
28 class ChannelProxy; 29 class ChannelProxy;
29 } // namespace voe 30 } // namespace voe
30 31
31 namespace internal { 32 namespace internal {
32 class AudioSendStream final : public webrtc::AudioSendStream, 33 class AudioSendStream final : public webrtc::AudioSendStream,
33 public webrtc::BitrateAllocatorObserver { 34 public webrtc::BitrateAllocatorObserver {
34 public: 35 public:
35 AudioSendStream(const webrtc::AudioSendStream::Config& config, 36 AudioSendStream(const webrtc::AudioSendStream::Config& config,
36 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, 37 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
37 rtc::TaskQueue* worker_queue, 38 rtc::TaskQueue* worker_queue,
38 CongestionController* congestion_controller, 39 CongestionController* congestion_controller,
39 BitrateAllocator* bitrate_allocator, 40 BitrateAllocator* bitrate_allocator,
40 RtcEventLog* event_log); 41 RtcEventLog* event_log,
42 RtcpRttStats* rtcp_rtt_stats);
41 ~AudioSendStream() override; 43 ~AudioSendStream() override;
42 44
43 // webrtc::AudioSendStream implementation. 45 // webrtc::AudioSendStream implementation.
44 void Start() override; 46 void Start() override;
45 void Stop() override; 47 void Stop() override;
46 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, 48 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
47 int duration_ms) override; 49 int duration_ms) override;
48 void SetMuted(bool muted) override; 50 void SetMuted(bool muted) override;
49 webrtc::AudioSendStream::Stats GetStats() const override; 51 webrtc::AudioSendStream::Stats GetStats() const override;
50 52
(...skipping 20 matching lines...) Expand all
71 std::unique_ptr<voe::ChannelProxy> channel_proxy_; 73 std::unique_ptr<voe::ChannelProxy> channel_proxy_;
72 74
73 BitrateAllocator* const bitrate_allocator_; 75 BitrateAllocator* const bitrate_allocator_;
74 76
75 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); 77 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
76 }; 78 };
77 } // namespace internal 79 } // namespace internal
78 } // namespace webrtc 80 } // namespace webrtc
79 81
80 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ 82 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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