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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 class PacketRouter; | 27 class PacketRouter; |
27 | 28 |
28 namespace voe { | 29 namespace voe { |
29 class ChannelProxy; | 30 class ChannelProxy; |
30 } // namespace voe | 31 } // namespace voe |
31 | 32 |
32 namespace internal { | 33 namespace internal { |
33 class AudioSendStream final : public webrtc::AudioSendStream, | 34 class AudioSendStream final : public webrtc::AudioSendStream, |
34 public webrtc::BitrateAllocatorObserver { | 35 public webrtc::BitrateAllocatorObserver { |
35 public: | 36 public: |
36 AudioSendStream(const webrtc::AudioSendStream::Config& config, | 37 AudioSendStream(const webrtc::AudioSendStream::Config& config, |
37 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 38 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
38 rtc::TaskQueue* worker_queue, | 39 rtc::TaskQueue* worker_queue, |
39 PacketRouter* packet_router, | 40 PacketRouter* packet_router, |
40 CongestionController* congestion_controller, | 41 CongestionController* congestion_controller, |
41 BitrateAllocator* bitrate_allocator, | 42 BitrateAllocator* bitrate_allocator, |
42 RtcEventLog* event_log); | 43 RtcEventLog* event_log, |
| 44 RtcpRttStats* rtcp_rtt_stats); |
43 ~AudioSendStream() override; | 45 ~AudioSendStream() override; |
44 | 46 |
45 // webrtc::AudioSendStream implementation. | 47 // webrtc::AudioSendStream implementation. |
46 void Start() override; | 48 void Start() override; |
47 void Stop() override; | 49 void Stop() override; |
48 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, | 50 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, |
49 int duration_ms) override; | 51 int duration_ms) override; |
50 void SetMuted(bool muted) override; | 52 void SetMuted(bool muted) override; |
51 webrtc::AudioSendStream::Stats GetStats() const override; | 53 webrtc::AudioSendStream::Stats GetStats() const override; |
52 | 54 |
(...skipping 21 matching lines...) Expand all Loading... |
74 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 76 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
75 | 77 |
76 BitrateAllocator* const bitrate_allocator_; | 78 BitrateAllocator* const bitrate_allocator_; |
77 | 79 |
78 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 80 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
79 }; | 81 }; |
80 } // namespace internal | 82 } // namespace internal |
81 } // namespace webrtc | 83 } // namespace webrtc |
82 | 84 |
83 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 85 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
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