Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 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 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 11 | #ifndef CALL_AUDIO_RECEIVE_STREAM_H_ |
| 12 | #define CALL_AUDIO_RECEIVE_STREAM_H_ |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 13 | |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 14 | #include <map> |
kwiberg | fffa42b | 2016-02-23 10:46:32 -0800 | [diff] [blame] | 15 | #include <memory> |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 16 | #include <string> |
| 17 | #include <vector> |
| 18 | |
Danil Chapovalov | b9b146c | 2018-06-15 12:28:07 +0200 | [diff] [blame] | 19 | #include "absl/types/optional.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 20 | #include "api/audio_codecs/audio_decoder_factory.h" |
| 21 | #include "api/call/transport.h" |
Steve Anton | 10542f2 | 2019-01-11 09:11:00 -0800 | [diff] [blame] | 22 | #include "api/crypto/crypto_options.h" |
Niels Möller | a837030 | 2019-09-02 15:16:49 +0200 | [diff] [blame] | 23 | #include "api/crypto/frame_decryptor_interface.h" |
Anton Sukhanov | 4f08faa | 2019-05-21 11:12:57 -0700 | [diff] [blame] | 24 | #include "api/media_transport_config.h" |
Steve Anton | 10542f2 | 2019-01-11 09:11:00 -0800 | [diff] [blame] | 25 | #include "api/rtp_parameters.h" |
Mirko Bonadei | d970807 | 2019-01-25 20:26:48 +0100 | [diff] [blame] | 26 | #include "api/scoped_refptr.h" |
Niels Möller | a837030 | 2019-09-02 15:16:49 +0200 | [diff] [blame] | 27 | #include "api/transport/rtp/rtp_source.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 28 | #include "call/rtp_config.h" |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 29 | |
| 30 | namespace webrtc { |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 31 | class AudioSinkInterface; |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 32 | |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 33 | class AudioReceiveStream { |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 34 | public: |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 35 | struct Stats { |
Paulina Hensman | 11b34f4 | 2018-04-09 14:24:52 +0200 | [diff] [blame] | 36 | Stats(); |
| 37 | ~Stats(); |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 38 | uint32_t remote_ssrc = 0; |
| 39 | int64_t bytes_rcvd = 0; |
| 40 | uint32_t packets_rcvd = 0; |
Ivo Creusen | 8d8ffdb | 2019-04-30 09:45:21 +0200 | [diff] [blame] | 41 | uint64_t fec_packets_received = 0; |
| 42 | uint64_t fec_packets_discarded = 0; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 43 | uint32_t packets_lost = 0; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 44 | std::string codec_name; |
Danil Chapovalov | b9b146c | 2018-06-15 12:28:07 +0200 | [diff] [blame] | 45 | absl::optional<int> codec_payload_type; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 46 | uint32_t jitter_ms = 0; |
| 47 | uint32_t jitter_buffer_ms = 0; |
| 48 | uint32_t jitter_buffer_preferred_ms = 0; |
| 49 | uint32_t delay_estimate_ms = 0; |
| 50 | int32_t audio_level = -1; |
Gustaf Ullberg | 9a2e906 | 2017-09-18 09:28:20 +0200 | [diff] [blame] | 51 | // Stats below correspond to similarly-named fields in the WebRTC stats |
| 52 | // spec. https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 53 | double total_output_energy = 0.0; |
Steve Anton | 2dbc69f | 2017-08-24 17:15:13 -0700 | [diff] [blame] | 54 | uint64_t total_samples_received = 0; |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 55 | double total_output_duration = 0.0; |
Steve Anton | 2dbc69f | 2017-08-24 17:15:13 -0700 | [diff] [blame] | 56 | uint64_t concealed_samples = 0; |
Ivo Creusen | 8d8ffdb | 2019-04-30 09:45:21 +0200 | [diff] [blame] | 57 | uint64_t silent_concealed_samples = 0; |
Gustaf Ullberg | 9a2e906 | 2017-09-18 09:28:20 +0200 | [diff] [blame] | 58 | uint64_t concealment_events = 0; |
Gustaf Ullberg | b0a0207 | 2017-10-02 12:00:34 +0200 | [diff] [blame] | 59 | double jitter_buffer_delay_seconds = 0.0; |
Chen Xing | 0acffb5 | 2019-01-15 15:46:29 +0100 | [diff] [blame] | 60 | uint64_t jitter_buffer_emitted_count = 0; |
Ivo Creusen | 8d8ffdb | 2019-04-30 09:45:21 +0200 | [diff] [blame] | 61 | uint64_t inserted_samples_for_deceleration = 0; |
| 62 | uint64_t removed_samples_for_acceleration = 0; |
Gustaf Ullberg | 9a2e906 | 2017-09-18 09:28:20 +0200 | [diff] [blame] | 63 | // Stats below DO NOT correspond directly to anything in the WebRTC stats |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 64 | float expand_rate = 0.0f; |
| 65 | float speech_expand_rate = 0.0f; |
| 66 | float secondary_decoded_rate = 0.0f; |
minyue-webrtc | 0e320ec | 2017-08-28 13:51:27 +0200 | [diff] [blame] | 67 | float secondary_discarded_rate = 0.0f; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 68 | float accelerate_rate = 0.0f; |
| 69 | float preemptive_expand_rate = 0.0f; |
Jakob Ivarsson | 352ce5c | 2018-11-27 12:52:16 +0100 | [diff] [blame] | 70 | uint64_t delayed_packet_outage_samples = 0; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 71 | int32_t decoding_calls_to_silence_generator = 0; |
| 72 | int32_t decoding_calls_to_neteq = 0; |
| 73 | int32_t decoding_normal = 0; |
Alex Narest | 5b5d97c | 2019-08-07 18:15:08 +0200 | [diff] [blame] | 74 | // TODO(alexnarest): Consider decoding_neteq_plc for consistency |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 75 | int32_t decoding_plc = 0; |
Alex Narest | 5b5d97c | 2019-08-07 18:15:08 +0200 | [diff] [blame] | 76 | int32_t decoding_codec_plc = 0; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 77 | int32_t decoding_cng = 0; |
| 78 | int32_t decoding_plc_cng = 0; |
henrik.lundin | 6348978 | 2016-09-20 01:47:12 -0700 | [diff] [blame] | 79 | int32_t decoding_muted_output = 0; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 80 | int64_t capture_start_ntp_time_ms = 0; |
Henrik Boström | 01738c6 | 2019-04-15 17:32:00 +0200 | [diff] [blame] | 81 | // The timestamp at which the last packet was received, i.e. the time of the |
| 82 | // local clock when it was received - not the RTP timestamp of that packet. |
| 83 | // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp |
| 84 | absl::optional<int64_t> last_packet_received_timestamp_ms; |
Ruslan Burakov | 8af8896 | 2018-11-22 17:21:10 +0100 | [diff] [blame] | 85 | uint64_t jitter_buffer_flushes = 0; |
Jakob Ivarsson | 232b3fd | 2019-03-06 09:18:40 +0100 | [diff] [blame] | 86 | double relative_packet_arrival_delay_seconds = 0.0; |
Henrik Lundin | 44125fa | 2019-04-29 17:00:46 +0200 | [diff] [blame] | 87 | int32_t interruption_count = 0; |
| 88 | int32_t total_interruption_duration_ms = 0; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 89 | }; |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 90 | |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 91 | struct Config { |
Paulina Hensman | 11b34f4 | 2018-04-09 14:24:52 +0200 | [diff] [blame] | 92 | Config(); |
| 93 | ~Config(); |
| 94 | |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 95 | std::string ToString() const; |
| 96 | |
| 97 | // Receive-stream specific RTP settings. |
| 98 | struct Rtp { |
Paulina Hensman | 11b34f4 | 2018-04-09 14:24:52 +0200 | [diff] [blame] | 99 | Rtp(); |
| 100 | ~Rtp(); |
| 101 | |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 102 | std::string ToString() const; |
| 103 | |
| 104 | // Synchronization source (stream identifier) to be received. |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 105 | uint32_t remote_ssrc = 0; |
| 106 | |
| 107 | // Sender SSRC used for sending RTCP (such as receiver reports). |
| 108 | uint32_t local_ssrc = 0; |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 109 | |
Stefan Holmer | 3842c5c | 2016-01-12 13:55:00 +0100 | [diff] [blame] | 110 | // Enable feedback for send side bandwidth estimation. |
| 111 | // See |
| 112 | // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions |
| 113 | // for details. |
| 114 | bool transport_cc = false; |
| 115 | |
solenberg | 8189b02 | 2016-06-14 12:13:00 -0700 | [diff] [blame] | 116 | // See NackConfig for description. |
| 117 | NackConfig nack; |
| 118 | |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 119 | // RTP header extensions used for the received stream. |
| 120 | std::vector<RtpExtension> extensions; |
| 121 | } rtp; |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 122 | |
solenberg | cf18b34 | 2015-10-01 08:13:42 -0700 | [diff] [blame] | 123 | Transport* rtcp_send_transport = nullptr; |
| 124 | |
Anton Sukhanov | 4f08faa | 2019-05-21 11:12:57 -0700 | [diff] [blame] | 125 | MediaTransportConfig media_transport_config; |
Niels Möller | 7d76a31 | 2018-10-26 12:57:07 +0200 | [diff] [blame] | 126 | |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 127 | // NetEq settings. |
Jakob Ivarsson | 647d5e6 | 2019-03-15 10:37:31 +0100 | [diff] [blame] | 128 | size_t jitter_buffer_max_packets = 200; |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 129 | bool jitter_buffer_fast_accelerate = false; |
Jakob Ivarsson | 10403ae | 2018-11-27 15:45:20 +0100 | [diff] [blame] | 130 | int jitter_buffer_min_delay_ms = 0; |
Jakob Ivarsson | 53eae87 | 2019-01-10 15:58:36 +0100 | [diff] [blame] | 131 | bool jitter_buffer_enable_rtx_handling = false; |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 132 | |
pbos | 8fc7fa7 | 2015-07-15 08:02:58 -0700 | [diff] [blame] | 133 | // Identifier for an A/V synchronization group. Empty string to disable. |
| 134 | // TODO(pbos): Synchronize streams in a sync group, not just one video |
| 135 | // stream to one audio stream. Tracked by issue webrtc:4762. |
| 136 | std::string sync_group; |
| 137 | |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 138 | // Decoder specifications for every payload type that we can receive. |
| 139 | std::map<int, SdpAudioFormat> decoder_map; |
ossu | 29b1a8d | 2016-06-13 07:34:51 -0700 | [diff] [blame] | 140 | |
| 141 | rtc::scoped_refptr<AudioDecoderFactory> decoder_factory; |
Karl Wiberg | 0812634 | 2018-03-20 19:18:55 +0100 | [diff] [blame] | 142 | |
Danil Chapovalov | b9b146c | 2018-06-15 12:28:07 +0200 | [diff] [blame] | 143 | absl::optional<AudioCodecPairId> codec_pair_id; |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 144 | |
Benjamin Wright | bfb444c | 2018-10-15 10:20:24 -0700 | [diff] [blame] | 145 | // Per PeerConnection crypto options. |
| 146 | webrtc::CryptoOptions crypto_options; |
| 147 | |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 148 | // An optional custom frame decryptor that allows the entire frame to be |
| 149 | // decrypted in whatever way the caller choses. This is not required by |
| 150 | // default. |
| 151 | rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor; |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 152 | }; |
| 153 | |
Fredrik Solenberg | 3b903d0 | 2018-01-10 15:17:10 +0100 | [diff] [blame] | 154 | // Reconfigure the stream according to the Configuration. |
| 155 | virtual void Reconfigure(const Config& config) = 0; |
| 156 | |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 157 | // Starts stream activity. |
| 158 | // When a stream is active, it can receive, process and deliver packets. |
| 159 | virtual void Start() = 0; |
| 160 | // Stops stream activity. |
| 161 | // When a stream is stopped, it can't receive, process or deliver packets. |
| 162 | virtual void Stop() = 0; |
| 163 | |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 164 | virtual Stats GetStats() const = 0; |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 165 | |
| 166 | // Sets an audio sink that receives unmixed audio from the receive stream. |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 167 | // Ownership of the sink is managed by the caller. |
deadbeef | 884f585 | 2016-01-15 09:20:04 -0800 | [diff] [blame] | 168 | // Only one sink can be set and passing a null sink clears an existing one. |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 169 | // NOTE: Audio must still somehow be pulled through AudioTransport for audio |
| 170 | // to stream through this sink. In practice, this happens if mixed audio |
| 171 | // is being pulled+rendered and/or if audio is being pulled for the purposes |
| 172 | // of feeding to the AEC. |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 173 | virtual void SetSink(AudioSinkInterface* sink) = 0; |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 174 | |
solenberg | 217fb66 | 2016-06-17 08:30:54 -0700 | [diff] [blame] | 175 | // Sets playback gain of the stream, applied when mixing, and thus after it |
| 176 | // is potentially forwarded to any attached AudioSinkInterface implementation. |
| 177 | virtual void SetGain(float gain) = 0; |
| 178 | |
Ruslan Burakov | 3b50f9f | 2019-02-06 09:45:56 +0100 | [diff] [blame] | 179 | // Sets a base minimum for the playout delay. Base minimum delay sets lower |
| 180 | // bound on minimum delay value determining lower bound on playout delay. |
| 181 | // |
| 182 | // Returns true if value was successfully set, false overwise. |
| 183 | virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0; |
| 184 | |
| 185 | // Returns current value of base minimum delay in milliseconds. |
| 186 | virtual int GetBaseMinimumPlayoutDelayMs() const = 0; |
| 187 | |
hbos | 8d609f6 | 2017-04-10 07:39:05 -0700 | [diff] [blame] | 188 | virtual std::vector<RtpSource> GetSources() const = 0; |
| 189 | |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 190 | protected: |
| 191 | virtual ~AudioReceiveStream() {} |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 192 | }; |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 193 | } // namespace webrtc |
| 194 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 195 | #endif // CALL_AUDIO_RECEIVE_STREAM_H_ |