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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 event, | 48 bool SendTelephoneEvent(int payload_type, 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 17 matching lines...) Expand all Loading... |
68 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 70 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
69 | 71 |
70 BitrateAllocator* const bitrate_allocator_; | 72 BitrateAllocator* const bitrate_allocator_; |
71 | 73 |
72 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 74 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
73 }; | 75 }; |
74 } // namespace internal | 76 } // namespace internal |
75 } // namespace webrtc | 77 } // namespace webrtc |
76 | 78 |
77 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 79 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
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