audio_receive_stream.h (9186B)
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 CALL_AUDIO_RECEIVE_STREAM_H_ 12 #define CALL_AUDIO_RECEIVE_STREAM_H_ 13 14 #include <cstddef> 15 #include <cstdint> 16 #include <map> 17 #include <optional> 18 #include <string> 19 20 #include "api/audio/audio_mixer.h" 21 #include "api/audio_codecs/audio_codec_pair_id.h" 22 #include "api/audio_codecs/audio_decoder_factory.h" 23 #include "api/audio_codecs/audio_format.h" 24 #include "api/call/transport.h" 25 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" 26 #include "api/crypto/crypto_options.h" 27 #include "api/crypto/frame_decryptor_interface.h" 28 #include "api/frame_transformer_interface.h" 29 #include "api/rtp_headers.h" 30 #include "api/scoped_refptr.h" 31 #include "api/units/time_delta.h" 32 #include "api/units/timestamp.h" 33 #include "call/receive_stream.h" 34 #include "call/rtp_config.h" 35 36 namespace webrtc { 37 class AudioSinkInterface; 38 39 class AudioReceiveStreamInterface : public MediaReceiveStreamInterface { 40 public: 41 struct Stats { 42 Stats(); 43 ~Stats(); 44 uint32_t remote_ssrc = 0; 45 int64_t payload_bytes_received = 0; 46 int64_t header_and_padding_bytes_received = 0; 47 uint32_t packets_received = 0; 48 // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsreceivedwithect1 49 int64_t packets_received_with_ect1 = 0; 50 // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsreceivedwithce 51 int64_t packets_received_with_ce = 0; 52 uint64_t fec_packets_received = 0; 53 uint64_t fec_packets_discarded = 0; 54 int32_t packets_lost = 0; 55 uint64_t packets_discarded = 0; 56 uint32_t nacks_sent = 0; 57 std::string codec_name; 58 std::optional<int> codec_payload_type; 59 uint32_t jitter_ms = 0; 60 uint32_t jitter_buffer_ms = 0; 61 uint32_t jitter_buffer_preferred_ms = 0; 62 uint32_t delay_estimate_ms = 0; 63 int32_t audio_level = -1; 64 // Stats below correspond to similarly-named fields in the WebRTC stats 65 // spec. https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats 66 double total_output_energy = 0.0; 67 uint64_t total_samples_received = 0; 68 double total_output_duration = 0.0; 69 uint64_t concealed_samples = 0; 70 uint64_t silent_concealed_samples = 0; 71 uint64_t concealment_events = 0; 72 double jitter_buffer_delay_seconds = 0.0; 73 uint64_t jitter_buffer_emitted_count = 0; 74 double jitter_buffer_target_delay_seconds = 0.0; 75 double jitter_buffer_minimum_delay_seconds = 0.0; 76 uint64_t inserted_samples_for_deceleration = 0; 77 uint64_t removed_samples_for_acceleration = 0; 78 double total_processing_delay_seconds = 0.0; 79 // Stats below DO NOT correspond directly to anything in the WebRTC stats 80 float expand_rate = 0.0f; 81 float speech_expand_rate = 0.0f; 82 float secondary_decoded_rate = 0.0f; 83 float secondary_discarded_rate = 0.0f; 84 float accelerate_rate = 0.0f; 85 float preemptive_expand_rate = 0.0f; 86 uint64_t delayed_packet_outage_samples = 0; 87 int32_t decoding_calls_to_silence_generator = 0; 88 int32_t decoding_calls_to_neteq = 0; 89 int32_t decoding_normal = 0; 90 // TODO(alexnarest): Consider decoding_neteq_plc for consistency 91 int32_t decoding_plc = 0; 92 int32_t decoding_codec_plc = 0; 93 int32_t decoding_cng = 0; 94 int32_t decoding_plc_cng = 0; 95 int32_t decoding_muted_output = 0; 96 int64_t capture_start_ntp_time_ms = 0; 97 // The timestamp at which the last packet was received, i.e. the time of the 98 // local clock when it was received - not the RTP timestamp of that packet. 99 // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp 100 std::optional<Timestamp> last_packet_received; 101 uint64_t jitter_buffer_flushes = 0; 102 double relative_packet_arrival_delay_seconds = 0.0; 103 int32_t interruption_count = 0; 104 int32_t total_interruption_duration_ms = 0; 105 // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp 106 std::optional<int64_t> estimated_playout_ntp_timestamp_ms; 107 // Remote outbound stats derived by the received RTCP sender reports. 108 // https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict* 109 std::optional<Timestamp> last_sender_report_timestamp; 110 // TODO: bugs.webrtc.org/370535296 - Remove the utc timestamp when linked 111 // issue is fixed. 112 std::optional<Timestamp> last_sender_report_utc_timestamp; 113 std::optional<Timestamp> last_sender_report_remote_utc_timestamp; 114 uint64_t sender_reports_packets_sent = 0; 115 uint64_t sender_reports_bytes_sent = 0; 116 uint64_t sender_reports_reports_count = 0; 117 std::optional<TimeDelta> round_trip_time; 118 TimeDelta total_round_trip_time = TimeDelta::Zero(); 119 int round_trip_time_measurements = 0; 120 }; 121 122 struct Config { 123 Config(); 124 ~Config(); 125 126 std::string ToString() const; 127 128 // Receive-stream specific RTP settings. 129 struct Rtp : public ReceiveStreamRtpConfig { 130 Rtp(); 131 ~Rtp(); 132 133 std::string ToString() const; 134 135 // See NackConfig for description. 136 NackConfig nack; 137 RtcpMode rtcp_mode = RtcpMode::kCompound; 138 139 RtcpEventObserver* rtcp_event_observer = nullptr; 140 } rtp; 141 142 // Receive-side RTT. 143 bool enable_non_sender_rtt = false; 144 145 Transport* rtcp_send_transport = nullptr; 146 147 // NetEq settings. 148 size_t jitter_buffer_max_packets = 200; 149 bool jitter_buffer_fast_accelerate = false; 150 int jitter_buffer_min_delay_ms = 0; 151 152 // Identifier for an A/V synchronization group. Empty string to disable. 153 // TODO(pbos): Synchronize streams in a sync group, not just one video 154 // stream to one audio stream. Tracked by issue webrtc:4762. 155 std::string sync_group; 156 157 // Decoder specifications for every payload type that we can receive. 158 std::map<int, SdpAudioFormat> decoder_map; 159 160 scoped_refptr<AudioDecoderFactory> decoder_factory; 161 162 std::optional<AudioCodecPairId> codec_pair_id; 163 164 // Per PeerConnection crypto options. 165 webrtc::CryptoOptions crypto_options; 166 167 // An optional custom frame decryptor that allows the entire frame to be 168 // decrypted in whatever way the caller choses. This is not required by 169 // default. 170 // TODO(tommi): Remove this member variable from the struct. It's not 171 // a part of the AudioReceiveStreamInterface state but rather a pass through 172 // variable. 173 scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor; 174 175 // An optional frame transformer used by insertable streams to transform 176 // encoded frames. 177 // TODO(tommi): Remove this member variable from the struct. It's not 178 // a part of the AudioReceiveStreamInterface state but rather a pass through 179 // variable. 180 scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer; 181 }; 182 183 // Methods that support reconfiguring the stream post initialization. 184 virtual void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) = 0; 185 virtual void SetNackHistory(int history_ms) = 0; 186 virtual void SetNonSenderRttMeasurement(bool enabled) = 0; 187 188 // Returns true if the stream has been started. 189 virtual bool IsRunning() const = 0; 190 191 virtual Stats GetStats(bool get_and_clear_legacy_stats) const = 0; 192 Stats GetStats() { return GetStats(/*get_and_clear_legacy_stats=*/true); } 193 194 // Sets an audio sink that receives unmixed audio from the receive stream. 195 // Ownership of the sink is managed by the caller. 196 // Only one sink can be set and passing a null sink clears an existing one. 197 // NOTE: Audio must still somehow be pulled through AudioTransport for audio 198 // to stream through this sink. In practice, this happens if mixed audio 199 // is being pulled+rendered and/or if audio is being pulled for the purposes 200 // of feeding to the AEC. 201 virtual void SetSink(AudioSinkInterface* sink) = 0; 202 203 // Sets playback gain of the stream, applied when mixing, and thus after it 204 // is potentially forwarded to any attached AudioSinkInterface implementation. 205 virtual void SetGain(float gain) = 0; 206 207 // Sets a base minimum for the playout delay. Base minimum delay sets lower 208 // bound on minimum delay value determining lower bound on playout delay. 209 // 210 // Returns true if value was successfully set, false overwise. 211 virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0; 212 213 // Returns current value of base minimum delay in milliseconds. 214 virtual int GetBaseMinimumPlayoutDelayMs() const = 0; 215 216 // Synchronization source (stream identifier) to be received. 217 // This member will not change mid-stream and can be assumed to be const 218 // post initialization. 219 virtual uint32_t remote_ssrc() const = 0; 220 221 // Get the object suitable to inject into the AudioMixer 222 // (normally "this"). 223 virtual AudioMixer::Source* source() = 0; 224 225 protected: 226 virtual ~AudioReceiveStreamInterface() {} 227 }; 228 229 } // namespace webrtc 230 231 #endif // CALL_AUDIO_RECEIVE_STREAM_H_