solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [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 AUDIO_AUDIO_SEND_STREAM_H_ |
| 12 | #define AUDIO_AUDIO_SEND_STREAM_H_ |
solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [diff] [blame] | 13 | |
kwiberg | fffa42b | 2016-02-23 10:46:32 -0800 | [diff] [blame] | 14 | #include <memory> |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 15 | #include <vector> |
kwiberg | fffa42b | 2016-02-23 10:46:32 -0800 | [diff] [blame] | 16 | |
Niels Möller | dced9f6 | 2018-11-19 10:27:07 +0100 | [diff] [blame] | 17 | #include "audio/channel_send.h" |
Fredrik Solenberg | a8b7c7f | 2018-01-17 11:18:31 +0100 | [diff] [blame] | 18 | #include "audio/transport_feedback_packet_loss_tracker.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 19 | #include "call/audio_send_stream.h" |
| 20 | #include "call/audio_state.h" |
| 21 | #include "call/bitrate_allocator.h" |
| 22 | #include "modules/rtp_rtcp/include/rtp_rtcp.h" |
Steve Anton | 10542f2 | 2019-01-11 09:11:00 -0800 | [diff] [blame] | 23 | #include "rtc_base/constructor_magic.h" |
Sebastian Jansson | 470a5ea | 2019-01-23 12:37:49 +0100 | [diff] [blame] | 24 | #include "rtc_base/experiments/audio_allocation_settings.h" |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 25 | #include "rtc_base/race_checker.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 26 | #include "rtc_base/thread_checker.h" |
solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [diff] [blame] | 27 | |
| 28 | namespace webrtc { |
terelius | e035e2d | 2016-09-21 06:51:47 -0700 | [diff] [blame] | 29 | class RtcEventLog; |
stefan | 7de8d64 | 2017-02-07 07:14:08 -0800 | [diff] [blame] | 30 | class RtcpBandwidthObserver; |
michaelt | 9332b7d | 2016-11-30 07:51:13 -0800 | [diff] [blame] | 31 | class RtcpRttStats; |
nisse | b8f9a32 | 2017-03-27 05:36:15 -0700 | [diff] [blame] | 32 | class RtpTransportControllerSendInterface; |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 33 | |
solenberg | 1372508 | 2015-11-25 08:16:52 -0800 | [diff] [blame] | 34 | namespace internal { |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 35 | class AudioState; |
| 36 | |
mflodman | 86cc6ff | 2016-07-26 04:44:06 -0700 | [diff] [blame] | 37 | class AudioSendStream final : public webrtc::AudioSendStream, |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 38 | public webrtc::BitrateAllocatorObserver, |
Anton Sukhanov | 626015d | 2019-02-04 15:16:06 -0800 | [diff] [blame] | 39 | public webrtc::PacketFeedbackObserver, |
| 40 | public webrtc::OverheadObserver { |
solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [diff] [blame] | 41 | public: |
Sebastian Jansson | 977b335 | 2019-03-04 17:43:34 +0100 | [diff] [blame] | 42 | AudioSendStream(Clock* clock, |
| 43 | const webrtc::AudioSendStream::Config& config, |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 44 | const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
perkj | 26091b1 | 2016-09-01 01:17:40 -0700 | [diff] [blame] | 45 | rtc::TaskQueue* worker_queue, |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 46 | ProcessThread* module_process_thread, |
Niels Möller | 7d76a31 | 2018-10-26 12:57:07 +0200 | [diff] [blame] | 47 | RtpTransportControllerSendInterface* rtp_transport, |
Niels Möller | 67b011d | 2018-10-22 13:00:40 +0200 | [diff] [blame] | 48 | BitrateAllocatorInterface* bitrate_allocator, |
michaelt | 9332b7d | 2016-11-30 07:51:13 -0800 | [diff] [blame] | 49 | RtcEventLog* event_log, |
ossu | c3d4b48 | 2017-05-23 06:07:11 -0700 | [diff] [blame] | 50 | RtcpRttStats* rtcp_rtt_stats, |
Sam Zackrisson | ff05816 | 2018-11-20 17:15:13 +0100 | [diff] [blame] | 51 | const absl::optional<RtpState>& suspended_rtp_state); |
Niels Möller | dced9f6 | 2018-11-19 10:27:07 +0100 | [diff] [blame] | 52 | // For unit tests, which need to supply a mock ChannelSend. |
Sebastian Jansson | 977b335 | 2019-03-04 17:43:34 +0100 | [diff] [blame] | 53 | AudioSendStream(Clock* clock, |
| 54 | const webrtc::AudioSendStream::Config& config, |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 55 | const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| 56 | rtc::TaskQueue* worker_queue, |
Niels Möller | 7d76a31 | 2018-10-26 12:57:07 +0200 | [diff] [blame] | 57 | RtpTransportControllerSendInterface* rtp_transport, |
Niels Möller | 67b011d | 2018-10-22 13:00:40 +0200 | [diff] [blame] | 58 | BitrateAllocatorInterface* bitrate_allocator, |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 59 | RtcEventLog* event_log, |
| 60 | RtcpRttStats* rtcp_rtt_stats, |
Danil Chapovalov | b9b146c | 2018-06-15 12:28:07 +0200 | [diff] [blame] | 61 | const absl::optional<RtpState>& suspended_rtp_state, |
Niels Möller | dced9f6 | 2018-11-19 10:27:07 +0100 | [diff] [blame] | 62 | std::unique_ptr<voe::ChannelSendInterface> channel_send); |
solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [diff] [blame] | 63 | ~AudioSendStream() override; |
| 64 | |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 65 | // webrtc::AudioSendStream implementation. |
eladalon | abbc430 | 2017-07-26 02:09:44 -0700 | [diff] [blame] | 66 | const webrtc::AudioSendStream::Config& GetConfig() const override; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 67 | void Reconfigure(const webrtc::AudioSendStream::Config& config) override; |
solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [diff] [blame] | 68 | void Start() override; |
| 69 | void Stop() override; |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 70 | void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 71 | bool SendTelephoneEvent(int payload_type, |
| 72 | int payload_frequency, |
| 73 | int event, |
solenberg | 8842c3e | 2016-03-11 03:06:41 -0800 | [diff] [blame] | 74 | int duration_ms) override; |
solenberg | 9421853 | 2016-06-16 10:53:22 -0700 | [diff] [blame] | 75 | void SetMuted(bool muted) override; |
solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [diff] [blame] | 76 | webrtc::AudioSendStream::Stats GetStats() const override; |
Ivo Creusen | 56d4609 | 2017-11-24 17:29:59 +0100 | [diff] [blame] | 77 | webrtc::AudioSendStream::Stats GetStats( |
| 78 | bool has_remote_tracks) const override; |
solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [diff] [blame] | 79 | |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 80 | void SignalNetworkState(NetworkState state); |
Niels Möller | 8fb1a6a | 2019-03-05 14:29:42 +0100 | [diff] [blame] | 81 | void DeliverRtcp(const uint8_t* packet, size_t length); |
mflodman | 86cc6ff | 2016-07-26 04:44:06 -0700 | [diff] [blame] | 82 | |
| 83 | // Implements BitrateAllocatorObserver. |
Sebastian Jansson | c0e4d45 | 2018-10-25 15:08:32 +0200 | [diff] [blame] | 84 | uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override; |
mflodman | 86cc6ff | 2016-07-26 04:44:06 -0700 | [diff] [blame] | 85 | |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 86 | // From PacketFeedbackObserver. |
| 87 | void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override; |
| 88 | void OnPacketFeedbackVector( |
| 89 | const std::vector<PacketFeedback>& packet_feedback_vector) override; |
| 90 | |
Anton Sukhanov | 626015d | 2019-02-04 15:16:06 -0800 | [diff] [blame] | 91 | void SetTransportOverhead(int transport_overhead_per_packet_bytes); |
| 92 | |
| 93 | // OverheadObserver override reports audio packetization overhead from |
| 94 | // RTP/RTCP module or Media Transport. |
| 95 | void OnOverheadChanged(size_t overhead_bytes_per_packet_bytes) override; |
solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [diff] [blame] | 96 | |
ossu | c3d4b48 | 2017-05-23 06:07:11 -0700 | [diff] [blame] | 97 | RtpState GetRtpState() const; |
Niels Möller | dced9f6 | 2018-11-19 10:27:07 +0100 | [diff] [blame] | 98 | const voe::ChannelSendInterface* GetChannel() const; |
ossu | c3d4b48 | 2017-05-23 06:07:11 -0700 | [diff] [blame] | 99 | |
Anton Sukhanov | 626015d | 2019-02-04 15:16:06 -0800 | [diff] [blame] | 100 | // Returns combined per-packet overhead. |
| 101 | size_t TestOnlyGetPerPacketOverheadBytes() const |
| 102 | RTC_LOCKS_EXCLUDED(overhead_per_packet_lock_); |
| 103 | |
solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [diff] [blame] | 104 | private: |
saza | c58f8c0 | 2017-07-19 00:39:19 -0700 | [diff] [blame] | 105 | class TimedTransport; |
| 106 | |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 107 | internal::AudioState* audio_state(); |
| 108 | const internal::AudioState* audio_state() const; |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 109 | |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 110 | void StoreEncoderProperties(int sample_rate_hz, size_t num_channels); |
| 111 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 112 | // These are all static to make it less likely that (the old) config_ is |
| 113 | // accessed unintentionally. |
| 114 | static void ConfigureStream(AudioSendStream* stream, |
| 115 | const Config& new_config, |
| 116 | bool first_time); |
| 117 | static bool SetupSendCodec(AudioSendStream* stream, const Config& new_config); |
| 118 | static bool ReconfigureSendCodec(AudioSendStream* stream, |
| 119 | const Config& new_config); |
| 120 | static void ReconfigureANA(AudioSendStream* stream, const Config& new_config); |
| 121 | static void ReconfigureCNG(AudioSendStream* stream, const Config& new_config); |
| 122 | static void ReconfigureBitrateObserver(AudioSendStream* stream, |
| 123 | const Config& new_config); |
| 124 | |
Seth Hampson | 24722b3 | 2017-12-22 09:36:42 -0800 | [diff] [blame] | 125 | void ConfigureBitrateObserver(int min_bitrate_bps, |
| 126 | int max_bitrate_bps, |
Sebastian Jansson | 79f0d4d | 2019-01-23 09:41:43 +0100 | [diff] [blame] | 127 | double bitrate_priority); |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 128 | void RemoveBitrateObserver(); |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 129 | |
Anton Sukhanov | 626015d | 2019-02-04 15:16:06 -0800 | [diff] [blame] | 130 | // Sets per-packet overhead on encoded (for ANA) based on current known values |
| 131 | // of transport and packetization overheads. |
| 132 | void UpdateOverheadForEncoder() |
| 133 | RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_); |
| 134 | |
| 135 | // Returns combined per-packet overhead. |
| 136 | size_t GetPerPacketOverheadBytes() const |
| 137 | RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_); |
| 138 | |
ossu | 3b9ff38 | 2017-04-27 08:03:42 -0700 | [diff] [blame] | 139 | void RegisterCngPayloadType(int payload_type, int clockrate_hz); |
Sebastian Jansson | 977b335 | 2019-03-04 17:43:34 +0100 | [diff] [blame] | 140 | Clock* clock_; |
ossu | 3b9ff38 | 2017-04-27 08:03:42 -0700 | [diff] [blame] | 141 | |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 142 | rtc::ThreadChecker worker_thread_checker_; |
| 143 | rtc::ThreadChecker pacer_thread_checker_; |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 144 | rtc::RaceChecker audio_capture_race_checker_; |
perkj | 26091b1 | 2016-09-01 01:17:40 -0700 | [diff] [blame] | 145 | rtc::TaskQueue* worker_queue_; |
Sebastian Jansson | 470a5ea | 2019-01-23 12:37:49 +0100 | [diff] [blame] | 146 | const AudioAllocationSettings allocation_settings_; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 147 | webrtc::AudioSendStream::Config config_; |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 148 | rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
Niels Möller | dced9f6 | 2018-11-19 10:27:07 +0100 | [diff] [blame] | 149 | const std::unique_ptr<voe::ChannelSendInterface> channel_send_; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 150 | RtcEventLog* const event_log_; |
solenberg | 85a0496 | 2015-10-27 03:35:21 -0700 | [diff] [blame] | 151 | |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 152 | int encoder_sample_rate_hz_ = 0; |
| 153 | size_t encoder_num_channels_ = 0; |
| 154 | bool sending_ = false; |
| 155 | |
Niels Möller | 67b011d | 2018-10-22 13:00:40 +0200 | [diff] [blame] | 156 | BitrateAllocatorInterface* const bitrate_allocator_; |
Niels Möller | 7d76a31 | 2018-10-26 12:57:07 +0200 | [diff] [blame] | 157 | RtpTransportControllerSendInterface* const rtp_transport_; |
mflodman | 86cc6ff | 2016-07-26 04:44:06 -0700 | [diff] [blame] | 158 | |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 159 | rtc::CriticalSection packet_loss_tracker_cs_; |
| 160 | TransportFeedbackPacketLossTracker packet_loss_tracker_ |
danilchap | a37de39 | 2017-09-09 04:17:22 -0700 | [diff] [blame] | 161 | RTC_GUARDED_BY(&packet_loss_tracker_cs_); |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 162 | |
ossu | c3d4b48 | 2017-05-23 06:07:11 -0700 | [diff] [blame] | 163 | RtpRtcp* rtp_rtcp_module_; |
Danil Chapovalov | b9b146c | 2018-06-15 12:28:07 +0200 | [diff] [blame] | 164 | absl::optional<RtpState> const suspended_rtp_state_; |
ossu | c3d4b48 | 2017-05-23 06:07:11 -0700 | [diff] [blame] | 165 | |
Alex Narest | cedd351 | 2017-12-07 20:54:55 +0100 | [diff] [blame] | 166 | // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is |
| 167 | // reserved for padding and MUST NOT be used as a local identifier. |
| 168 | // So it should be safe to use 0 here to indicate "not configured". |
| 169 | struct ExtensionIds { |
| 170 | int audio_level = 0; |
| 171 | int transport_sequence_number = 0; |
Steve Anton | bb50ce5 | 2018-03-26 10:24:32 -0700 | [diff] [blame] | 172 | int mid = 0; |
Amit Hilbuch | 77938e6 | 2018-12-21 09:23:38 -0800 | [diff] [blame] | 173 | int rid = 0; |
| 174 | int repaired_rid = 0; |
Alex Narest | cedd351 | 2017-12-07 20:54:55 +0100 | [diff] [blame] | 175 | }; |
| 176 | static ExtensionIds FindExtensionIds( |
| 177 | const std::vector<RtpExtension>& extensions); |
Sebastian Jansson | 470a5ea | 2019-01-23 12:37:49 +0100 | [diff] [blame] | 178 | static int TransportSeqNumId(const Config& config); |
Alex Narest | cedd351 | 2017-12-07 20:54:55 +0100 | [diff] [blame] | 179 | |
Anton Sukhanov | 626015d | 2019-02-04 15:16:06 -0800 | [diff] [blame] | 180 | rtc::CriticalSection overhead_per_packet_lock_; |
| 181 | |
| 182 | // Current transport overhead (ICE, TURN, etc.) |
| 183 | size_t transport_overhead_per_packet_bytes_ |
| 184 | RTC_GUARDED_BY(overhead_per_packet_lock_) = 0; |
| 185 | |
| 186 | // Current audio packetization overhead (RTP or Media Transport). |
| 187 | size_t audio_overhead_per_packet_bytes_ |
| 188 | RTC_GUARDED_BY(overhead_per_packet_lock_) = 0; |
| 189 | |
solenberg | 85a0496 | 2015-10-27 03:35:21 -0700 | [diff] [blame] | 190 | RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [diff] [blame] | 191 | }; |
| 192 | } // namespace internal |
| 193 | } // namespace webrtc |
| 194 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 195 | #endif // AUDIO_AUDIO_SEND_STREAM_H_ |