1 /* 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef AUDIO_AUDIO_SEND_STREAM_H_ 12 #define AUDIO_AUDIO_SEND_STREAM_H_ 13 14 #include <memory> 15 #include <vector> 16 17 #include "audio/time_interval.h" 18 #include "call/audio_send_stream.h" 19 #include "call/audio_state.h" 20 #include "call/bitrate_allocator.h" 21 #include "modules/rtp_rtcp/include/rtp_rtcp.h" 22 #include "rtc_base/constructormagic.h" 23 #include "rtc_base/thread_checker.h" 24 #include "voice_engine/transport_feedback_packet_loss_tracker.h" 25 26 namespace webrtc { 27 class VoiceEngine; 28 class RtcEventLog; 29 class RtcpBandwidthObserver; 30 class RtcpRttStats; 31 class RtpTransportControllerSendInterface; 32 33 namespace voe { 34 class ChannelProxy; 35 } // namespace voe 36 37 namespace internal { 38 class AudioSendStream final : public webrtc::AudioSendStream, 39 public webrtc::BitrateAllocatorObserver, 40 public webrtc::PacketFeedbackObserver { 41 public: 42 AudioSendStream(const webrtc::AudioSendStream::Config& config, 43 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, 44 rtc::TaskQueue* worker_queue, 45 RtpTransportControllerSendInterface* transport, 46 BitrateAllocator* bitrate_allocator, 47 RtcEventLog* event_log, 48 RtcpRttStats* rtcp_rtt_stats, 49 const rtc::Optional<RtpState>& suspended_rtp_state); 50 ~AudioSendStream() override; 51 52 // webrtc::AudioSendStream implementation. 53 const webrtc::AudioSendStream::Config& GetConfig() const override; 54 void Reconfigure(const webrtc::AudioSendStream::Config& config) override; 55 void Start() override; 56 void Stop() override; 57 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, 58 int duration_ms) override; 59 void SetMuted(bool muted) override; 60 webrtc::AudioSendStream::Stats GetStats() const override; 61 webrtc::AudioSendStream::Stats GetStats( 62 bool has_remote_tracks) const override; 63 64 void SignalNetworkState(NetworkState state); 65 bool DeliverRtcp(const uint8_t* packet, size_t length); 66 67 // Implements BitrateAllocatorObserver. 68 uint32_t OnBitrateUpdated(uint32_t bitrate_bps, 69 uint8_t fraction_loss, 70 int64_t rtt, 71 int64_t bwe_period_ms) override; 72 73 // From PacketFeedbackObserver. 74 void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override; 75 void OnPacketFeedbackVector( 76 const std::vector<PacketFeedback>& packet_feedback_vector) override; 77 78 void SetTransportOverhead(int transport_overhead_per_packet); 79 80 RtpState GetRtpState() const; 81 const TimeInterval& GetActiveLifetime() const; 82 83 private: 84 class TimedTransport; 85 86 VoiceEngine* voice_engine() const; 87 88 // These are all static to make it less likely that (the old) config_ is 89 // accessed unintentionally. 90 static void ConfigureStream(AudioSendStream* stream, 91 const Config& new_config, 92 bool first_time); 93 static bool SetupSendCodec(AudioSendStream* stream, const Config& new_config); 94 static bool ReconfigureSendCodec(AudioSendStream* stream, 95 const Config& new_config); 96 static void ReconfigureANA(AudioSendStream* stream, const Config& new_config); 97 static void ReconfigureCNG(AudioSendStream* stream, const Config& new_config); 98 static void ReconfigureBitrateObserver(AudioSendStream* stream, 99 const Config& new_config); 100 101 void ConfigureBitrateObserver(int min_bitrate_bps, int max_bitrate_bps); 102 void RemoveBitrateObserver(); 103 104 void RegisterCngPayloadType(int payload_type, int clockrate_hz); 105 106 rtc::ThreadChecker worker_thread_checker_; 107 rtc::ThreadChecker pacer_thread_checker_; 108 rtc::TaskQueue* worker_queue_; 109 webrtc::AudioSendStream::Config config_; 110 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 111 std::unique_ptr<voe::ChannelProxy> channel_proxy_; 112 RtcEventLog* const event_log_; 113 114 BitrateAllocator* const bitrate_allocator_; 115 RtpTransportControllerSendInterface* const transport_; 116 117 rtc::CriticalSection packet_loss_tracker_cs_; 118 TransportFeedbackPacketLossTracker packet_loss_tracker_ 119 RTC_GUARDED_BY(&packet_loss_tracker_cs_); 120 121 RtpRtcp* rtp_rtcp_module_; 122 rtc::Optional<RtpState> const suspended_rtp_state_; 123 124 std::unique_ptr<TimedTransport> timed_send_transport_adapter_; 125 TimeInterval active_lifetime_; 126 127 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); 128 }; 129 } // namespace internal 130 } // namespace webrtc 131 132 #endif // AUDIO_AUDIO_SEND_STREAM_H_ 133