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