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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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38 AudioSendStream(const webrtc::AudioSendStream::Config& config, | 38 AudioSendStream(const webrtc::AudioSendStream::Config& config, |
39 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 39 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
40 rtc::TaskQueue* worker_queue, | 40 rtc::TaskQueue* worker_queue, |
41 PacketRouter* packet_router, | 41 PacketRouter* packet_router, |
42 CongestionController* congestion_controller, | 42 CongestionController* congestion_controller, |
43 BitrateAllocator* bitrate_allocator, | 43 BitrateAllocator* bitrate_allocator, |
44 RtcEventLog* event_log, | 44 RtcEventLog* event_log, |
45 RtcpRttStats* rtcp_rtt_stats); | 45 RtcpRttStats* rtcp_rtt_stats); |
46 ~AudioSendStream() override; | 46 ~AudioSendStream() override; |
47 | 47 |
48 void Reconfigure(const webrtc::AudioSendStream::Config& config) override; | |
the sun
2017/04/04 23:02:54
nit: looks like it should go under the below comme
| |
49 | |
48 // webrtc::AudioSendStream implementation. | 50 // webrtc::AudioSendStream implementation. |
49 void Start() override; | 51 void Start() override; |
50 void Stop() override; | 52 void Stop() override; |
51 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, | 53 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, |
52 int duration_ms) override; | 54 int duration_ms) override; |
53 void SetMuted(bool muted) override; | 55 void SetMuted(bool muted) override; |
54 webrtc::AudioSendStream::Stats GetStats() const override; | 56 webrtc::AudioSendStream::Stats GetStats() const override; |
55 | 57 |
56 void SignalNetworkState(NetworkState state); | 58 void SignalNetworkState(NetworkState state); |
57 bool DeliverRtcp(const uint8_t* packet, size_t length); | 59 bool DeliverRtcp(const uint8_t* packet, size_t length); |
58 | 60 |
59 // Implements BitrateAllocatorObserver. | 61 // Implements BitrateAllocatorObserver. |
60 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, | 62 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, |
61 uint8_t fraction_loss, | 63 uint8_t fraction_loss, |
62 int64_t rtt, | 64 int64_t rtt, |
63 int64_t probing_interval_ms) override; | 65 int64_t probing_interval_ms) override; |
64 | 66 |
65 const webrtc::AudioSendStream::Config& config() const; | 67 const webrtc::AudioSendStream::Config& config() const; |
66 void SetTransportOverhead(int transport_overhead_per_packet); | 68 void SetTransportOverhead(int transport_overhead_per_packet); |
67 | 69 |
68 private: | 70 private: |
69 VoiceEngine* voice_engine() const; | 71 VoiceEngine* voice_engine() const; |
70 | 72 |
71 bool SetupSendCodec(); | 73 bool SetupSendCodec(const Config& new_config); |
74 bool ReconfigureSendCodec(const Config& new_config); | |
75 void ReconfigureANA(const Config& new_config); | |
76 void ReconfigureCNG(const Config& new_config); | |
77 void ReconfigureBitrateObserver(const Config& new_config); | |
78 | |
79 void ConfigureBitrateObserver(int min_bitrate_bps, int max_bitrate_bps); | |
80 void RemoveBitrateObserver(); | |
72 | 81 |
73 rtc::ThreadChecker thread_checker_; | 82 rtc::ThreadChecker thread_checker_; |
74 rtc::TaskQueue* worker_queue_; | 83 rtc::TaskQueue* worker_queue_; |
75 const webrtc::AudioSendStream::Config config_; | 84 webrtc::AudioSendStream::Config config_; |
76 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 85 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
77 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 86 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
87 RtcEventLog* const event_log_; | |
78 | 88 |
89 PacketRouter* const packet_router_; | |
79 BitrateAllocator* const bitrate_allocator_; | 90 BitrateAllocator* const bitrate_allocator_; |
80 CongestionController* const congestion_controller_; | 91 CongestionController* const congestion_controller_; |
81 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; | 92 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; |
82 | 93 |
83 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 94 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
84 }; | 95 }; |
85 } // namespace internal | 96 } // namespace internal |
86 } // namespace webrtc | 97 } // namespace webrtc |
87 | 98 |
88 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 99 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
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