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 <map> 15 #include <memory> 16 #include <string> 17 #include <vector> 18 19 #include "absl/types/optional.h" 20 #include "api/audio_codecs/audio_decoder_factory.h" 21 #include "api/call/transport.h" 22 #include "api/crypto/crypto_options.h" 23 #include "api/crypto/frame_decryptor_interface.h" 24 #include "api/frame_transformer_interface.h" 25 #include "api/rtp_parameters.h" 26 #include "api/scoped_refptr.h" 27 #include "api/transport/rtp/rtp_source.h" 28 #include "call/rtp_config.h" 29 30 namespace webrtc { 31 class AudioSinkInterface; 32 33 class AudioReceiveStream { 34 public: 35 struct Stats { 36 Stats(); 37 ~Stats(); 38 uint32_t remote_ssrc = 0; 39 int64_t payload_bytes_rcvd = 0; 40 int64_t header_and_padding_bytes_rcvd = 0; 41 uint32_t packets_rcvd = 0; 42 uint64_t fec_packets_received = 0; 43 uint64_t fec_packets_discarded = 0; 44 uint32_t packets_lost = 0; 45 std::string codec_name; 46 absl::optional<int> codec_payload_type; 47 uint32_t jitter_ms = 0; 48 uint32_t jitter_buffer_ms = 0; 49 uint32_t jitter_buffer_preferred_ms = 0; 50 uint32_t delay_estimate_ms = 0; 51 int32_t audio_level = -1; 52 // Stats below correspond to similarly-named fields in the WebRTC stats 53 // spec. https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats 54 double total_output_energy = 0.0; 55 uint64_t total_samples_received = 0; 56 double total_output_duration = 0.0; 57 uint64_t concealed_samples = 0; 58 uint64_t silent_concealed_samples = 0; 59 uint64_t concealment_events = 0; 60 double jitter_buffer_delay_seconds = 0.0; 61 uint64_t jitter_buffer_emitted_count = 0; 62 double jitter_buffer_target_delay_seconds = 0.0; 63 uint64_t inserted_samples_for_deceleration = 0; 64 uint64_t removed_samples_for_acceleration = 0; 65 // Stats below DO NOT correspond directly to anything in the WebRTC stats 66 float expand_rate = 0.0f; 67 float speech_expand_rate = 0.0f; 68 float secondary_decoded_rate = 0.0f; 69 float secondary_discarded_rate = 0.0f; 70 float accelerate_rate = 0.0f; 71 float preemptive_expand_rate = 0.0f; 72 uint64_t delayed_packet_outage_samples = 0; 73 int32_t decoding_calls_to_silence_generator = 0; 74 int32_t decoding_calls_to_neteq = 0; 75 int32_t decoding_normal = 0; 76 // TODO(alexnarest): Consider decoding_neteq_plc for consistency 77 int32_t decoding_plc = 0; 78 int32_t decoding_codec_plc = 0; 79 int32_t decoding_cng = 0; 80 int32_t decoding_plc_cng = 0; 81 int32_t decoding_muted_output = 0; 82 int64_t capture_start_ntp_time_ms = 0; 83 // The timestamp at which the last packet was received, i.e. the time of the 84 // local clock when it was received - not the RTP timestamp of that packet. 85 // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp 86 absl::optional<int64_t> last_packet_received_timestamp_ms; 87 uint64_t jitter_buffer_flushes = 0; 88 double relative_packet_arrival_delay_seconds = 0.0; 89 int32_t interruption_count = 0; 90 int32_t total_interruption_duration_ms = 0; 91 // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp 92 absl::optional<int64_t> estimated_playout_ntp_timestamp_ms; 93 }; 94 95 struct Config { 96 Config(); 97 ~Config(); 98 99 std::string ToString() const; 100 101 // Receive-stream specific RTP settings. 102 struct Rtp { 103 Rtp(); 104 ~Rtp(); 105 106 std::string ToString() const; 107 108 // Synchronization source (stream identifier) to be received. 109 uint32_t remote_ssrc = 0; 110 111 // Sender SSRC used for sending RTCP (such as receiver reports). 112 uint32_t local_ssrc = 0; 113 114 // Enable feedback for send side bandwidth estimation. 115 // See 116 // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions 117 // for details. 118 bool transport_cc = false; 119 120 // See NackConfig for description. 121 NackConfig nack; 122 123 // RTP header extensions used for the received stream. 124 std::vector<RtpExtension> extensions; 125 } rtp; 126 127 Transport* rtcp_send_transport = nullptr; 128 129 // NetEq settings. 130 size_t jitter_buffer_max_packets = 200; 131 bool jitter_buffer_fast_accelerate = false; 132 int jitter_buffer_min_delay_ms = 0; 133 bool jitter_buffer_enable_rtx_handling = false; 134 135 // Identifier for an A/V synchronization group. Empty string to disable. 136 // TODO(pbos): Synchronize streams in a sync group, not just one video 137 // stream to one audio stream. Tracked by issue webrtc:4762. 138 std::string sync_group; 139 140 // Decoder specifications for every payload type that we can receive. 141 std::map<int, SdpAudioFormat> decoder_map; 142 143 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory; 144 145 absl::optional<AudioCodecPairId> codec_pair_id; 146 147 // Per PeerConnection crypto options. 148 webrtc::CryptoOptions crypto_options; 149 150 // An optional custom frame decryptor that allows the entire frame to be 151 // decrypted in whatever way the caller choses. This is not required by 152 // default. 153 rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor; 154 155 // An optional frame transformer used by insertable streams to transform 156 // encoded frames. 157 rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer; 158 }; 159 160 // Reconfigure the stream according to the Configuration. 161 virtual void Reconfigure(const Config& config) = 0; 162 163 // Starts stream activity. 164 // When a stream is active, it can receive, process and deliver packets. 165 virtual void Start() = 0; 166 // Stops stream activity. 167 // When a stream is stopped, it can't receive, process or deliver packets. 168 virtual void Stop() = 0; 169 170 virtual Stats GetStats() const = 0; 171 172 // Sets an audio sink that receives unmixed audio from the receive stream. 173 // Ownership of the sink is managed by the caller. 174 // Only one sink can be set and passing a null sink clears an existing one. 175 // NOTE: Audio must still somehow be pulled through AudioTransport for audio 176 // to stream through this sink. In practice, this happens if mixed audio 177 // is being pulled+rendered and/or if audio is being pulled for the purposes 178 // of feeding to the AEC. 179 virtual void SetSink(AudioSinkInterface* sink) = 0; 180 181 // Sets playback gain of the stream, applied when mixing, and thus after it 182 // is potentially forwarded to any attached AudioSinkInterface implementation. 183 virtual void SetGain(float gain) = 0; 184 185 // Sets a base minimum for the playout delay. Base minimum delay sets lower 186 // bound on minimum delay value determining lower bound on playout delay. 187 // 188 // Returns true if value was successfully set, false overwise. 189 virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0; 190 191 // Returns current value of base minimum delay in milliseconds. 192 virtual int GetBaseMinimumPlayoutDelayMs() const = 0; 193 194 virtual std::vector<RtpSource> GetSources() const = 0; 195 196 protected: ~AudioReceiveStream()197 virtual ~AudioReceiveStream() {} 198 }; 199 } // namespace webrtc 200 201 #endif // CALL_AUDIO_RECEIVE_STREAM_H_ 202