1 /* 2 * Copyright (c) 2012 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_CHANNEL_RECEIVE_H_ 12 #define AUDIO_CHANNEL_RECEIVE_H_ 13 14 #include <map> 15 #include <memory> 16 #include <utility> 17 #include <vector> 18 19 #include "absl/types/optional.h" 20 #include "api/audio/audio_mixer.h" 21 #include "api/audio_codecs/audio_decoder_factory.h" 22 #include "api/call/audio_sink.h" 23 #include "api/call/transport.h" 24 #include "api/crypto/crypto_options.h" 25 #include "api/frame_transformer_interface.h" 26 #include "api/neteq/neteq_factory.h" 27 #include "api/transport/rtp/rtp_source.h" 28 #include "call/rtp_packet_sink_interface.h" 29 #include "call/syncable.h" 30 #include "modules/audio_coding/include/audio_coding_module_typedefs.h" 31 #include "modules/rtp_rtcp/source/source_tracker.h" 32 #include "system_wrappers/include/clock.h" 33 34 // TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence 35 // warnings about use of unsigned short. 36 // These need cleanup, in a separate cl. 37 38 namespace rtc { 39 class TimestampWrapAroundHandler; 40 } 41 42 namespace webrtc { 43 44 class AudioDeviceModule; 45 class FrameDecryptorInterface; 46 class PacketRouter; 47 class RateLimiter; 48 class ReceiveStatistics; 49 class RtcEventLog; 50 class RtpPacketReceived; 51 class RtpRtcp; 52 53 struct CallReceiveStatistics { 54 int cumulativeLost; 55 unsigned int jitterSamples; 56 int64_t payload_bytes_rcvd = 0; 57 int64_t header_and_padding_bytes_rcvd = 0; 58 int packetsReceived; 59 uint32_t nacks_sent = 0; 60 // The capture NTP time (in local timebase) of the first played out audio 61 // frame. 62 int64_t capture_start_ntp_time_ms_; 63 // The timestamp at which the last packet was received, i.e. the time of the 64 // local clock when it was received - not the RTP timestamp of that packet. 65 // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp 66 absl::optional<int64_t> last_packet_received_timestamp_ms; 67 // Remote outbound stats derived by the received RTCP sender reports. 68 // Note that the timestamps below correspond to the time elapsed since the 69 // Unix epoch. 70 // https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict* 71 absl::optional<int64_t> last_sender_report_timestamp_ms; 72 absl::optional<int64_t> last_sender_report_remote_timestamp_ms; 73 uint32_t sender_reports_packets_sent = 0; 74 uint64_t sender_reports_bytes_sent = 0; 75 uint64_t sender_reports_reports_count = 0; 76 absl::optional<TimeDelta> round_trip_time; 77 TimeDelta total_round_trip_time = TimeDelta::Zero(); 78 int round_trip_time_measurements; 79 }; 80 81 namespace voe { 82 83 class ChannelSendInterface; 84 85 // Interface class needed for AudioReceiveStreamInterface tests that use a 86 // MockChannelReceive. 87 88 class ChannelReceiveInterface : public RtpPacketSinkInterface { 89 public: 90 virtual ~ChannelReceiveInterface() = default; 91 92 virtual void SetSink(AudioSinkInterface* sink) = 0; 93 94 virtual void SetReceiveCodecs( 95 const std::map<int, SdpAudioFormat>& codecs) = 0; 96 97 virtual void StartPlayout() = 0; 98 virtual void StopPlayout() = 0; 99 100 // Payload type and format of last received RTP packet, if any. 101 virtual absl::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec() 102 const = 0; 103 104 virtual void ReceivedRTCPPacket(const uint8_t* data, size_t length) = 0; 105 106 virtual void SetChannelOutputVolumeScaling(float scaling) = 0; 107 virtual int GetSpeechOutputLevelFullRange() const = 0; 108 // See description of "totalAudioEnergy" in the WebRTC stats spec: 109 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy 110 virtual double GetTotalOutputEnergy() const = 0; 111 virtual double GetTotalOutputDuration() const = 0; 112 113 // Stats. 114 virtual NetworkStatistics GetNetworkStatistics( 115 bool get_and_clear_legacy_stats) const = 0; 116 virtual AudioDecodingCallStats GetDecodingCallStatistics() const = 0; 117 118 // Audio+Video Sync. 119 virtual uint32_t GetDelayEstimate() const = 0; 120 virtual bool SetMinimumPlayoutDelay(int delay_ms) = 0; 121 virtual bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp, 122 int64_t* time_ms) const = 0; 123 virtual void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms, 124 int64_t time_ms) = 0; 125 virtual absl::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs( 126 int64_t now_ms) const = 0; 127 128 // Audio quality. 129 // Base minimum delay sets lower bound on minimum delay value which 130 // determines minimum delay until audio playout. 131 virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0; 132 virtual int GetBaseMinimumPlayoutDelayMs() const = 0; 133 134 // Produces the transport-related timestamps; current_delay_ms is left unset. 135 virtual absl::optional<Syncable::Info> GetSyncInfo() const = 0; 136 137 virtual void RegisterReceiverCongestionControlObjects( 138 PacketRouter* packet_router) = 0; 139 virtual void ResetReceiverCongestionControlObjects() = 0; 140 141 virtual CallReceiveStatistics GetRTCPStatistics() const = 0; 142 virtual void SetNACKStatus(bool enable, int max_packets) = 0; 143 virtual void SetNonSenderRttMeasurement(bool enabled) = 0; 144 145 virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( 146 int sample_rate_hz, 147 AudioFrame* audio_frame) = 0; 148 149 virtual int PreferredSampleRate() const = 0; 150 151 // Sets the source tracker to notify about "delivered" packets when output is 152 // muted. 153 virtual void SetSourceTracker(SourceTracker* source_tracker) = 0; 154 155 // Associate to a send channel. 156 // Used for obtaining RTT for a receive-only channel. 157 virtual void SetAssociatedSendChannel( 158 const ChannelSendInterface* channel) = 0; 159 160 // Sets a frame transformer between the depacketizer and the decoder, to 161 // transform the received frames before decoding them. 162 virtual void SetDepacketizerToDecoderFrameTransformer( 163 rtc::scoped_refptr<webrtc::FrameTransformerInterface> 164 frame_transformer) = 0; 165 166 virtual void SetFrameDecryptor( 167 rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0; 168 169 virtual void OnLocalSsrcChange(uint32_t local_ssrc) = 0; 170 virtual uint32_t GetLocalSsrc() const = 0; 171 }; 172 173 std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive( 174 Clock* clock, 175 NetEqFactory* neteq_factory, 176 AudioDeviceModule* audio_device_module, 177 Transport* rtcp_send_transport, 178 RtcEventLog* rtc_event_log, 179 uint32_t local_ssrc, 180 uint32_t remote_ssrc, 181 size_t jitter_buffer_max_packets, 182 bool jitter_buffer_fast_playout, 183 int jitter_buffer_min_delay_ms, 184 bool enable_non_sender_rtt, 185 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory, 186 absl::optional<AudioCodecPairId> codec_pair_id, 187 rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor, 188 const webrtc::CryptoOptions& crypto_options, 189 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer); 190 191 } // namespace voe 192 } // namespace webrtc 193 194 #endif // AUDIO_CHANNEL_RECEIVE_H_ 195