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Fredrik Solenberg23fba1f2015-04-29 13:24:011/*
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 Bonadei92ea95e2017-09-15 04:47:3111#ifndef CALL_AUDIO_RECEIVE_STREAM_H_
12#define CALL_AUDIO_RECEIVE_STREAM_H_
Fredrik Solenberg23fba1f2015-04-29 13:24:0113
Fredrik Solenberg04f49312015-06-08 11:04:5614#include <map>
kwibergfffa42b2016-02-23 18:46:3215#include <memory>
Fredrik Solenberg23fba1f2015-04-29 13:24:0116#include <string>
17#include <vector>
18
Danil Chapovalovb9b146c2018-06-15 10:28:0719#include "absl/types/optional.h"
Mirko Bonadei92ea95e2017-09-15 04:47:3120#include "api/audio_codecs/audio_decoder_factory.h"
21#include "api/call/transport.h"
Steve Anton10542f22019-01-11 17:11:0022#include "api/crypto/crypto_options.h"
Niels Möllera8370302019-09-02 13:16:4923#include "api/crypto/frame_decryptor_interface.h"
Marina Ciocea3e9af7f2020-04-01 05:46:1624#include "api/frame_transformer_interface.h"
Steve Anton10542f22019-01-11 17:11:0025#include "api/rtp_parameters.h"
Mirko Bonadeid9708072019-01-25 19:26:4826#include "api/scoped_refptr.h"
Niels Möllera8370302019-09-02 13:16:4927#include "api/transport/rtp/rtp_source.h"
Mirko Bonadei92ea95e2017-09-15 04:47:3128#include "call/rtp_config.h"
Fredrik Solenberg23fba1f2015-04-29 13:24:0129
30namespace webrtc {
Tommif888bb52015-12-12 00:37:0131class AudioSinkInterface;
Fredrik Solenberg04f49312015-06-08 11:04:5632
pbos1ba8d392016-05-02 03:18:3433class AudioReceiveStream {
Fredrik Solenberg23fba1f2015-04-29 13:24:0134 public:
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2735 struct Stats {
Paulina Hensman11b34f42018-04-09 12:24:5236 Stats();
37 ~Stats();
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2738 uint32_t remote_ssrc = 0;
Niels Möllerac0a4cb2019-10-09 13:01:3339 int64_t payload_bytes_rcvd = 0;
40 int64_t header_and_padding_bytes_rcvd = 0;
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2741 uint32_t packets_rcvd = 0;
Ivo Creusen8d8ffdb2019-04-30 07:45:2142 uint64_t fec_packets_received = 0;
43 uint64_t fec_packets_discarded = 0;
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2744 uint32_t packets_lost = 0;
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2745 std::string codec_name;
Danil Chapovalovb9b146c2018-06-15 10:28:0746 absl::optional<int> codec_payload_type;
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2747 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;
Gustaf Ullberg9a2e9062017-09-18 07:28:2052 // Stats below correspond to similarly-named fields in the WebRTC stats
53 // spec. https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
zsteine76bd3a2017-07-14 19:17:4954 double total_output_energy = 0.0;
Steve Anton2dbc69f2017-08-25 00:15:1355 uint64_t total_samples_received = 0;
zsteine76bd3a2017-07-14 19:17:4956 double total_output_duration = 0.0;
Steve Anton2dbc69f2017-08-25 00:15:1357 uint64_t concealed_samples = 0;
Ivo Creusen8d8ffdb2019-04-30 07:45:2158 uint64_t silent_concealed_samples = 0;
Gustaf Ullberg9a2e9062017-09-18 07:28:2059 uint64_t concealment_events = 0;
Gustaf Ullbergb0a02072017-10-02 10:00:3460 double jitter_buffer_delay_seconds = 0.0;
Chen Xing0acffb52019-01-15 14:46:2961 uint64_t jitter_buffer_emitted_count = 0;
Artem Titove618cc92020-03-11 10:18:5462 double jitter_buffer_target_delay_seconds = 0.0;
Ivo Creusen8d8ffdb2019-04-30 07:45:2163 uint64_t inserted_samples_for_deceleration = 0;
64 uint64_t removed_samples_for_acceleration = 0;
Gustaf Ullberg9a2e9062017-09-18 07:28:2065 // Stats below DO NOT correspond directly to anything in the WebRTC stats
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2766 float expand_rate = 0.0f;
67 float speech_expand_rate = 0.0f;
68 float secondary_decoded_rate = 0.0f;
minyue-webrtc0e320ec2017-08-28 11:51:2769 float secondary_discarded_rate = 0.0f;
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2770 float accelerate_rate = 0.0f;
71 float preemptive_expand_rate = 0.0f;
Jakob Ivarsson352ce5c2018-11-27 11:52:1672 uint64_t delayed_packet_outage_samples = 0;
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2773 int32_t decoding_calls_to_silence_generator = 0;
74 int32_t decoding_calls_to_neteq = 0;
75 int32_t decoding_normal = 0;
Alex Narest5b5d97c2019-08-07 16:15:0876 // TODO(alexnarest): Consider decoding_neteq_plc for consistency
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2777 int32_t decoding_plc = 0;
Alex Narest5b5d97c2019-08-07 16:15:0878 int32_t decoding_codec_plc = 0;
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2779 int32_t decoding_cng = 0;
80 int32_t decoding_plc_cng = 0;
henrik.lundin63489782016-09-20 08:47:1281 int32_t decoding_muted_output = 0;
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2782 int64_t capture_start_ntp_time_ms = 0;
Henrik Boström01738c62019-04-15 15:32:0083 // 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;
Ruslan Burakov8af88962018-11-22 16:21:1087 uint64_t jitter_buffer_flushes = 0;
Jakob Ivarsson232b3fd2019-03-06 08:18:4088 double relative_packet_arrival_delay_seconds = 0.0;
Henrik Lundin44125fa2019-04-29 15:00:4689 int32_t interruption_count = 0;
90 int32_t total_interruption_duration_ms = 0;
Åsa Perssonfcf79cc2019-10-22 13:23:4491 // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
92 absl::optional<int64_t> estimated_playout_ntp_timestamp_ms;
Fredrik Solenberg4f4ec0a2015-10-22 08:49:2793 };
Fredrik Solenberg04f49312015-06-08 11:04:5694
Fredrik Solenberg23fba1f2015-04-29 13:24:0195 struct Config {
Paulina Hensman11b34f42018-04-09 12:24:5296 Config();
97 ~Config();
98
Fredrik Solenberg23fba1f2015-04-29 13:24:0199 std::string ToString() const;
100
101 // Receive-stream specific RTP settings.
102 struct Rtp {
Paulina Hensman11b34f42018-04-09 12:24:52103 Rtp();
104 ~Rtp();
105
Fredrik Solenberg23fba1f2015-04-29 13:24:01106 std::string ToString() const;
107
108 // Synchronization source (stream identifier) to be received.
Fredrik Solenberg04f49312015-06-08 11:04:56109 uint32_t remote_ssrc = 0;
110
111 // Sender SSRC used for sending RTCP (such as receiver reports).
112 uint32_t local_ssrc = 0;
Fredrik Solenberg23fba1f2015-04-29 13:24:01113
Stefan Holmer3842c5c2016-01-12 12:55:00114 // 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
solenberg8189b022016-06-14 19:13:00120 // See NackConfig for description.
121 NackConfig nack;
122
Fredrik Solenberg23fba1f2015-04-29 13:24:01123 // RTP header extensions used for the received stream.
124 std::vector<RtpExtension> extensions;
125 } rtp;
Fredrik Solenberg04f49312015-06-08 11:04:56126
solenbergcf18b342015-10-01 15:13:42127 Transport* rtcp_send_transport = nullptr;
128
Fredrik Solenberg8f5787a2018-01-11 12:52:30129 // NetEq settings.
Jakob Ivarsson647d5e62019-03-15 09:37:31130 size_t jitter_buffer_max_packets = 200;
Fredrik Solenberg8f5787a2018-01-11 12:52:30131 bool jitter_buffer_fast_accelerate = false;
Jakob Ivarsson10403ae2018-11-27 14:45:20132 int jitter_buffer_min_delay_ms = 0;
Jakob Ivarsson53eae872019-01-10 14:58:36133 bool jitter_buffer_enable_rtx_handling = false;
Fredrik Solenberg8f5787a2018-01-11 12:52:30134
pbos8fc7fa72015-07-15 15:02:58135 // 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
kwibergd32bf752017-01-19 15:03:59140 // Decoder specifications for every payload type that we can receive.
141 std::map<int, SdpAudioFormat> decoder_map;
ossu29b1a8d2016-06-13 14:34:51142
143 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
Karl Wiberg08126342018-03-20 18:18:55144
Danil Chapovalovb9b146c2018-06-15 10:28:07145 absl::optional<AudioCodecPairId> codec_pair_id;
Benjamin Wright84583f62018-10-04 21:22:34146
Benjamin Wrightbfb444c2018-10-15 17:20:24147 // Per PeerConnection crypto options.
148 webrtc::CryptoOptions crypto_options;
149
Benjamin Wright84583f62018-10-04 21:22:34150 // 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;
Marina Ciocea3e9af7f2020-04-01 05:46:16154
155 // An optional frame transformer used by insertable streams to transform
156 // encoded frames.
157 rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
Fredrik Solenberg23fba1f2015-04-29 13:24:01158 };
159
Fredrik Solenberg3b903d02018-01-10 14:17:10160 // Reconfigure the stream according to the Configuration.
161 virtual void Reconfigure(const Config& config) = 0;
162
pbos1ba8d392016-05-02 03:18:34163 // 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
Niels Möller6b4d9622020-09-14 08:47:50170 virtual Stats GetStats(bool get_and_clear_legacy_stats) const = 0;
171 Stats GetStats() { return GetStats(/*get_and_clear_legacy_stats=*/true); }
Tommif888bb52015-12-12 00:37:01172
173 // Sets an audio sink that receives unmixed audio from the receive stream.
Fredrik Solenberg8f5787a2018-01-11 12:52:30174 // Ownership of the sink is managed by the caller.
deadbeef884f5852016-01-15 17:20:04175 // Only one sink can be set and passing a null sink clears an existing one.
Tommif888bb52015-12-12 00:37:01176 // NOTE: Audio must still somehow be pulled through AudioTransport for audio
177 // to stream through this sink. In practice, this happens if mixed audio
178 // is being pulled+rendered and/or if audio is being pulled for the purposes
179 // of feeding to the AEC.
Fredrik Solenberg8f5787a2018-01-11 12:52:30180 virtual void SetSink(AudioSinkInterface* sink) = 0;
pbos1ba8d392016-05-02 03:18:34181
solenberg217fb662016-06-17 15:30:54182 // Sets playback gain of the stream, applied when mixing, and thus after it
183 // is potentially forwarded to any attached AudioSinkInterface implementation.
184 virtual void SetGain(float gain) = 0;
185
Ruslan Burakov3b50f9f2019-02-06 08:45:56186 // Sets a base minimum for the playout delay. Base minimum delay sets lower
187 // bound on minimum delay value determining lower bound on playout delay.
188 //
189 // Returns true if value was successfully set, false overwise.
190 virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
191
192 // Returns current value of base minimum delay in milliseconds.
193 virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
194
hbos8d609f62017-04-10 14:39:05195 virtual std::vector<RtpSource> GetSources() const = 0;
196
pbos1ba8d392016-05-02 03:18:34197 protected:
198 virtual ~AudioReceiveStream() {}
Fredrik Solenberg23fba1f2015-04-29 13:24:01199};
Fredrik Solenberg23fba1f2015-04-29 13:24:01200} // namespace webrtc
201
Mirko Bonadei92ea95e2017-09-15 04:47:31202#endif // CALL_AUDIO_RECEIVE_STREAM_H_