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_SEND_STREAM_H_ 12 #define CALL_AUDIO_SEND_STREAM_H_ 13 14 #include <memory> 15 #include <string> 16 #include <vector> 17 18 #include "absl/types/optional.h" 19 #include "api/audio_codecs/audio_codec_pair_id.h" 20 #include "api/audio_codecs/audio_encoder.h" 21 #include "api/audio_codecs/audio_encoder_factory.h" 22 #include "api/audio_codecs/audio_format.h" 23 #include "api/call/transport.h" 24 #include "api/crypto/crypto_options.h" 25 #include "api/crypto/frame_encryptor_interface.h" 26 #include "api/frame_transformer_interface.h" 27 #include "api/rtp_parameters.h" 28 #include "api/rtp_sender_interface.h" 29 #include "api/scoped_refptr.h" 30 #include "call/audio_sender.h" 31 #include "call/rtp_config.h" 32 #include "modules/audio_processing/include/audio_processing_statistics.h" 33 #include "modules/rtp_rtcp/include/report_block_data.h" 34 35 namespace webrtc { 36 37 class AudioSendStream : public AudioSender { 38 public: 39 struct Stats { 40 Stats(); 41 ~Stats(); 42 43 // TODO(solenberg): Harmonize naming and defaults with receive stream stats. 44 uint32_t local_ssrc = 0; 45 int64_t payload_bytes_sent = 0; 46 int64_t header_and_padding_bytes_sent = 0; 47 // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent 48 uint64_t retransmitted_bytes_sent = 0; 49 int32_t packets_sent = 0; 50 // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay 51 TimeDelta total_packet_send_delay = TimeDelta::Zero(); 52 // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent 53 uint64_t retransmitted_packets_sent = 0; 54 int32_t packets_lost = -1; 55 float fraction_lost = -1.0f; 56 std::string codec_name; 57 absl::optional<int> codec_payload_type; 58 int32_t jitter_ms = -1; 59 int64_t rtt_ms = -1; 60 int16_t audio_level = 0; 61 // See description of "totalAudioEnergy" in the WebRTC stats spec: 62 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy 63 double total_input_energy = 0.0; 64 double total_input_duration = 0.0; 65 66 ANAStats ana_statistics; 67 AudioProcessingStats apm_statistics; 68 69 int64_t target_bitrate_bps = 0; 70 // A snapshot of Report Blocks with additional data of interest to 71 // statistics. Within this list, the sender-source SSRC pair is unique and 72 // per-pair the ReportBlockData represents the latest Report Block that was 73 // received for that pair. 74 std::vector<ReportBlockData> report_block_datas; 75 uint32_t nacks_rcvd = 0; 76 }; 77 78 struct Config { 79 Config() = delete; 80 explicit Config(Transport* send_transport); 81 ~Config(); 82 std::string ToString() const; 83 84 // Send-stream specific RTP settings. 85 struct Rtp { 86 Rtp(); 87 ~Rtp(); 88 std::string ToString() const; 89 90 // Sender SSRC. 91 uint32_t ssrc = 0; 92 93 // The value to send in the RID RTP header extension if the extension is 94 // included in the list of extensions. 95 std::string rid; 96 97 // The value to send in the MID RTP header extension if the extension is 98 // included in the list of extensions. 99 std::string mid; 100 101 // Corresponds to the SDP attribute extmap-allow-mixed. 102 bool extmap_allow_mixed = false; 103 104 // RTP header extensions used for the sent stream. 105 std::vector<RtpExtension> extensions; 106 107 // RTCP CNAME, see RFC 3550. 108 std::string c_name; 109 } rtp; 110 111 // Time interval between RTCP report for audio 112 int rtcp_report_interval_ms = 5000; 113 114 // Transport for outgoing packets. The transport is expected to exist for 115 // the entire life of the AudioSendStream and is owned by the API client. 116 Transport* send_transport = nullptr; 117 118 // Bitrate limits used for variable audio bitrate streams. Set both to -1 to 119 // disable audio bitrate adaptation. 120 // Note: This is still an experimental feature and not ready for real usage. 121 int min_bitrate_bps = -1; 122 int max_bitrate_bps = -1; 123 124 double bitrate_priority = 1.0; 125 bool has_dscp = false; 126 127 // Defines whether to turn on audio network adaptor, and defines its config 128 // string. 129 absl::optional<std::string> audio_network_adaptor_config; 130 131 struct SendCodecSpec { 132 SendCodecSpec(int payload_type, const SdpAudioFormat& format); 133 ~SendCodecSpec(); 134 std::string ToString() const; 135 136 bool operator==(const SendCodecSpec& rhs) const; 137 bool operator!=(const SendCodecSpec& rhs) const { 138 return !(*this == rhs); 139 } 140 141 int payload_type; 142 SdpAudioFormat format; 143 bool nack_enabled = false; 144 bool transport_cc_enabled = false; 145 bool enable_non_sender_rtt = false; 146 absl::optional<int> cng_payload_type; 147 absl::optional<int> red_payload_type; 148 // If unset, use the encoder's default target bitrate. 149 absl::optional<int> target_bitrate_bps; 150 }; 151 152 absl::optional<SendCodecSpec> send_codec_spec; 153 rtc::scoped_refptr<AudioEncoderFactory> encoder_factory; 154 absl::optional<AudioCodecPairId> codec_pair_id; 155 156 // Track ID as specified during track creation. 157 std::string track_id; 158 159 // Per PeerConnection crypto options. 160 webrtc::CryptoOptions crypto_options; 161 162 // An optional custom frame encryptor that allows the entire frame to be 163 // encryptor in whatever way the caller choses. This is not required by 164 // default. 165 rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor; 166 167 // An optional frame transformer used by insertable streams to transform 168 // encoded frames. 169 rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer; 170 }; 171 172 virtual ~AudioSendStream() = default; 173 174 virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0; 175 176 // Reconfigure the stream according to the Configuration. 177 virtual void Reconfigure(const Config& config, 178 SetParametersCallback callback) = 0; 179 180 // Starts stream activity. 181 // When a stream is active, it can receive, process and deliver packets. 182 virtual void Start() = 0; 183 // Stops stream activity. 184 // When a stream is stopped, it can't receive, process or deliver packets. 185 virtual void Stop() = 0; 186 187 // TODO(solenberg): Make payload_type a config property instead. 188 virtual bool SendTelephoneEvent(int payload_type, 189 int payload_frequency, 190 int event, 191 int duration_ms) = 0; 192 193 virtual void SetMuted(bool muted) = 0; 194 195 virtual Stats GetStats() const = 0; 196 virtual Stats GetStats(bool has_remote_tracks) const = 0; 197 }; 198 199 } // namespace webrtc 200 201 #endif // CALL_AUDIO_SEND_STREAM_H_ 202