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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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41 AudioSendStream(const webrtc::AudioSendStream::Config& config, | 41 AudioSendStream(const webrtc::AudioSendStream::Config& config, |
42 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 42 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
43 rtc::TaskQueue* worker_queue, | 43 rtc::TaskQueue* worker_queue, |
44 RtpTransportControllerSendInterface* transport, | 44 RtpTransportControllerSendInterface* transport, |
45 BitrateAllocator* bitrate_allocator, | 45 BitrateAllocator* bitrate_allocator, |
46 RtcEventLog* event_log, | 46 RtcEventLog* event_log, |
47 RtcpRttStats* rtcp_rtt_stats); | 47 RtcpRttStats* rtcp_rtt_stats); |
48 ~AudioSendStream() override; | 48 ~AudioSendStream() override; |
49 | 49 |
50 // webrtc::AudioSendStream implementation. | 50 // webrtc::AudioSendStream implementation. |
| 51 void Reconfigure(const webrtc::AudioSendStream::Config& config) override; |
| 52 |
51 void Start() override; | 53 void Start() override; |
52 void Stop() override; | 54 void Stop() override; |
53 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, | 55 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, |
54 int duration_ms) override; | 56 int duration_ms) override; |
55 void SetMuted(bool muted) override; | 57 void SetMuted(bool muted) override; |
56 webrtc::AudioSendStream::Stats GetStats() const override; | 58 webrtc::AudioSendStream::Stats GetStats() const override; |
57 | 59 |
58 void SignalNetworkState(NetworkState state); | 60 void SignalNetworkState(NetworkState state); |
59 bool DeliverRtcp(const uint8_t* packet, size_t length); | 61 bool DeliverRtcp(const uint8_t* packet, size_t length); |
60 | 62 |
61 // Implements BitrateAllocatorObserver. | 63 // Implements BitrateAllocatorObserver. |
62 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, | 64 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, |
63 uint8_t fraction_loss, | 65 uint8_t fraction_loss, |
64 int64_t rtt, | 66 int64_t rtt, |
65 int64_t probing_interval_ms) override; | 67 int64_t probing_interval_ms) override; |
66 | 68 |
67 // From PacketFeedbackObserver. | 69 // From PacketFeedbackObserver. |
68 void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override; | 70 void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override; |
69 void OnPacketFeedbackVector( | 71 void OnPacketFeedbackVector( |
70 const std::vector<PacketFeedback>& packet_feedback_vector) override; | 72 const std::vector<PacketFeedback>& packet_feedback_vector) override; |
71 | 73 |
72 const webrtc::AudioSendStream::Config& config() const; | 74 const webrtc::AudioSendStream::Config& config() const; |
73 void SetTransportOverhead(int transport_overhead_per_packet); | 75 void SetTransportOverhead(int transport_overhead_per_packet); |
74 | 76 |
75 private: | 77 private: |
76 VoiceEngine* voice_engine() const; | 78 VoiceEngine* voice_engine() const; |
77 | 79 |
78 bool SetupSendCodec(); | 80 // These are all static to make it less likely that (the old) config_ is |
| 81 // accessed unintentionally. |
| 82 static void ConfigureStream(AudioSendStream* stream, |
| 83 const Config& new_config, |
| 84 bool first_time); |
| 85 static bool SetupSendCodec(AudioSendStream* stream, const Config& new_config); |
| 86 static bool ReconfigureSendCodec(AudioSendStream* stream, |
| 87 const Config& new_config); |
| 88 static void ReconfigureANA(AudioSendStream* stream, const Config& new_config); |
| 89 static void ReconfigureCNG(AudioSendStream* stream, const Config& new_config); |
| 90 static void ReconfigureBitrateObserver(AudioSendStream* stream, |
| 91 const Config& new_config); |
| 92 |
| 93 void ConfigureBitrateObserver(int min_bitrate_bps, int max_bitrate_bps); |
| 94 void RemoveBitrateObserver(); |
79 | 95 |
80 rtc::ThreadChecker worker_thread_checker_; | 96 rtc::ThreadChecker worker_thread_checker_; |
81 rtc::ThreadChecker pacer_thread_checker_; | 97 rtc::ThreadChecker pacer_thread_checker_; |
82 rtc::TaskQueue* worker_queue_; | 98 rtc::TaskQueue* worker_queue_; |
83 const webrtc::AudioSendStream::Config config_; | 99 webrtc::AudioSendStream::Config config_; |
84 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 100 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
85 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 101 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
| 102 RtcEventLog* const event_log_; |
86 | 103 |
87 BitrateAllocator* const bitrate_allocator_; | 104 BitrateAllocator* const bitrate_allocator_; |
88 RtpTransportControllerSendInterface* const transport_; | 105 RtpTransportControllerSendInterface* const transport_; |
89 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; | 106 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; |
90 | 107 |
91 rtc::CriticalSection packet_loss_tracker_cs_; | 108 rtc::CriticalSection packet_loss_tracker_cs_; |
92 TransportFeedbackPacketLossTracker packet_loss_tracker_ | 109 TransportFeedbackPacketLossTracker packet_loss_tracker_ |
93 GUARDED_BY(&packet_loss_tracker_cs_); | 110 GUARDED_BY(&packet_loss_tracker_cs_); |
94 | 111 |
95 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 112 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
96 }; | 113 }; |
97 } // namespace internal | 114 } // namespace internal |
98 } // namespace webrtc | 115 } // namespace webrtc |
99 | 116 |
100 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 117 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
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