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 <utility> 16 #include <vector> 17 18 #include "audio/audio_level.h" 19 #include "audio/channel_send.h" 20 #include "call/audio_send_stream.h" 21 #include "call/audio_state.h" 22 #include "call/bitrate_allocator.h" 23 #include "modules/rtp_rtcp/include/rtp_rtcp.h" 24 #include "rtc_base/constructor_magic.h" 25 #include "rtc_base/experiments/struct_parameters_parser.h" 26 #include "rtc_base/race_checker.h" 27 #include "rtc_base/task_queue.h" 28 #include "rtc_base/thread_checker.h" 29 30 namespace webrtc { 31 class RtcEventLog; 32 class RtcpBandwidthObserver; 33 class RtcpRttStats; 34 class RtpTransportControllerSendInterface; 35 36 struct AudioAllocationConfig { 37 static constexpr char kKey[] = "WebRTC-Audio-Allocation"; 38 // Field Trial configured bitrates to use as overrides over default/user 39 // configured bitrate range when audio bitrate allocation is enabled. 40 absl::optional<DataRate> min_bitrate; 41 absl::optional<DataRate> max_bitrate; 42 DataRate priority_bitrate = DataRate::Zero(); 43 // By default the priority_bitrate is compensated for packet overhead. 44 // Use this flag to configure a raw value instead. 45 absl::optional<DataRate> priority_bitrate_raw; 46 absl::optional<double> bitrate_priority; 47 48 std::unique_ptr<StructParametersParser> Parser(); 49 AudioAllocationConfig(); 50 }; 51 namespace internal { 52 class AudioState; 53 54 class AudioSendStream final : public webrtc::AudioSendStream, 55 public webrtc::BitrateAllocatorObserver, 56 public webrtc::OverheadObserver { 57 public: 58 AudioSendStream(Clock* clock, 59 const webrtc::AudioSendStream::Config& config, 60 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, 61 TaskQueueFactory* task_queue_factory, 62 ProcessThread* module_process_thread, 63 RtpTransportControllerSendInterface* rtp_transport, 64 BitrateAllocatorInterface* bitrate_allocator, 65 RtcEventLog* event_log, 66 RtcpRttStats* rtcp_rtt_stats, 67 const absl::optional<RtpState>& suspended_rtp_state); 68 // For unit tests, which need to supply a mock ChannelSend. 69 AudioSendStream(Clock* clock, 70 const webrtc::AudioSendStream::Config& config, 71 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, 72 TaskQueueFactory* task_queue_factory, 73 RtpTransportControllerSendInterface* rtp_transport, 74 BitrateAllocatorInterface* bitrate_allocator, 75 RtcEventLog* event_log, 76 RtcpRttStats* rtcp_rtt_stats, 77 const absl::optional<RtpState>& suspended_rtp_state, 78 std::unique_ptr<voe::ChannelSendInterface> channel_send); 79 ~AudioSendStream() override; 80 81 // webrtc::AudioSendStream implementation. 82 const webrtc::AudioSendStream::Config& GetConfig() const override; 83 void Reconfigure(const webrtc::AudioSendStream::Config& config) override; 84 void Start() override; 85 void Stop() override; 86 void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override; 87 bool SendTelephoneEvent(int payload_type, 88 int payload_frequency, 89 int event, 90 int duration_ms) override; 91 void SetMuted(bool muted) override; 92 webrtc::AudioSendStream::Stats GetStats() const override; 93 webrtc::AudioSendStream::Stats GetStats( 94 bool has_remote_tracks) const override; 95 96 void DeliverRtcp(const uint8_t* packet, size_t length); 97 98 // Implements BitrateAllocatorObserver. 99 uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override; 100 101 void SetTransportOverhead(int transport_overhead_per_packet_bytes); 102 103 // OverheadObserver override reports audio packetization overhead from 104 // RTP/RTCP module or Media Transport. 105 void OnOverheadChanged(size_t overhead_bytes_per_packet_bytes) override; 106 107 RtpState GetRtpState() const; 108 const voe::ChannelSendInterface* GetChannel() const; 109 110 // Returns combined per-packet overhead. 111 size_t TestOnlyGetPerPacketOverheadBytes() const 112 RTC_LOCKS_EXCLUDED(overhead_per_packet_lock_); 113 114 private: 115 class TimedTransport; 116 // Constraints including overhead. 117 struct TargetAudioBitrateConstraints { 118 DataRate min; 119 DataRate max; 120 }; 121 122 internal::AudioState* audio_state(); 123 const internal::AudioState* audio_state() const; 124 125 void StoreEncoderProperties(int sample_rate_hz, size_t num_channels); 126 127 void ConfigureStream(const Config& new_config, bool first_time); 128 bool SetupSendCodec(const Config& new_config); 129 bool ReconfigureSendCodec(const Config& new_config); 130 void ReconfigureANA(const Config& new_config); 131 void ReconfigureCNG(const Config& new_config); 132 void ReconfigureBitrateObserver(const Config& new_config); 133 134 void ConfigureBitrateObserver() RTC_RUN_ON(worker_queue_); 135 void RemoveBitrateObserver(); 136 137 // Returns bitrate constraints, maybe including overhead when enabled by 138 // field trial. 139 TargetAudioBitrateConstraints GetMinMaxBitrateConstraints() const 140 RTC_RUN_ON(worker_queue_); 141 142 // Sets per-packet overhead on encoded (for ANA) based on current known values 143 // of transport and packetization overheads. 144 void UpdateOverheadForEncoder() 145 RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_); 146 147 // Returns combined per-packet overhead. 148 size_t GetPerPacketOverheadBytes() const 149 RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_); 150 151 void RegisterCngPayloadType(int payload_type, int clockrate_hz); 152 Clock* clock_; 153 154 rtc::ThreadChecker worker_thread_checker_; 155 rtc::ThreadChecker pacer_thread_checker_; 156 rtc::RaceChecker audio_capture_race_checker_; 157 rtc::TaskQueue* worker_queue_; 158 159 const bool audio_send_side_bwe_; 160 const bool allocate_audio_without_feedback_; 161 const bool force_no_audio_feedback_ = allocate_audio_without_feedback_; 162 const bool enable_audio_alr_probing_; 163 const bool send_side_bwe_with_overhead_; 164 const AudioAllocationConfig allocation_settings_; 165 166 webrtc::AudioSendStream::Config config_; 167 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 168 const std::unique_ptr<voe::ChannelSendInterface> channel_send_; 169 RtcEventLog* const event_log_; 170 const bool use_legacy_overhead_calculation_; 171 172 int encoder_sample_rate_hz_ = 0; 173 size_t encoder_num_channels_ = 0; 174 bool sending_ = false; 175 rtc::CriticalSection audio_level_lock_; 176 // Keeps track of audio level, total audio energy and total samples duration. 177 // https://w3c.github.io/webrtc-stats/#dom-rtcaudiohandlerstats-totalaudioenergy 178 webrtc::voe::AudioLevel audio_level_; 179 180 BitrateAllocatorInterface* const bitrate_allocator_ 181 RTC_GUARDED_BY(worker_queue_); 182 RtpTransportControllerSendInterface* const rtp_transport_; 183 184 RtpRtcp* const rtp_rtcp_module_; 185 absl::optional<RtpState> const suspended_rtp_state_; 186 187 // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is 188 // reserved for padding and MUST NOT be used as a local identifier. 189 // So it should be safe to use 0 here to indicate "not configured". 190 struct ExtensionIds { 191 int audio_level = 0; 192 int abs_send_time = 0; 193 int abs_capture_time = 0; 194 int transport_sequence_number = 0; 195 int mid = 0; 196 int rid = 0; 197 int repaired_rid = 0; 198 }; 199 static ExtensionIds FindExtensionIds( 200 const std::vector<RtpExtension>& extensions); 201 static int TransportSeqNumId(const Config& config); 202 203 rtc::CriticalSection overhead_per_packet_lock_; 204 205 // Current transport overhead (ICE, TURN, etc.) 206 size_t transport_overhead_per_packet_bytes_ 207 RTC_GUARDED_BY(overhead_per_packet_lock_) = 0; 208 209 // Current audio packetization overhead (RTP or Media Transport). 210 size_t audio_overhead_per_packet_bytes_ 211 RTC_GUARDED_BY(overhead_per_packet_lock_) = 0; 212 213 bool registered_with_allocator_ RTC_GUARDED_BY(worker_queue_) = false; 214 size_t total_packet_overhead_bytes_ RTC_GUARDED_BY(worker_queue_) = 0; 215 absl::optional<std::pair<TimeDelta, TimeDelta>> frame_length_range_ 216 RTC_GUARDED_BY(worker_queue_); 217 218 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); 219 }; 220 } // namespace internal 221 } // namespace webrtc 222 223 #endif // AUDIO_AUDIO_SEND_STREAM_H_ 224