niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1 | /* |
henrika@webrtc.org | 2919e95 | 2012-01-31 08:45:03 +0000 | [diff] [blame] | 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 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 | |
Fredrik Solenberg | a8b7c7f | 2018-01-17 11:18:31 +0100 | [diff] [blame] | 11 | #include "audio/channel.h" |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 12 | |
Henrik Lundin | 64dad83 | 2015-05-11 12:44:23 +0200 | [diff] [blame] | 13 | #include <algorithm> |
Bjorn Terelius | 440216f | 2017-09-29 21:01:42 +0200 | [diff] [blame] | 14 | #include <map> |
Elad Alon | 604c14d | 2017-10-05 12:47:06 +0000 | [diff] [blame] | 15 | #include <memory> |
Bjorn Terelius | 440216f | 2017-09-29 21:01:42 +0200 | [diff] [blame] | 16 | #include <string> |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 17 | #include <utility> |
Bjorn Terelius | 440216f | 2017-09-29 21:01:42 +0200 | [diff] [blame] | 18 | #include <vector> |
Henrik Lundin | 64dad83 | 2015-05-11 12:44:23 +0200 | [diff] [blame] | 19 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 20 | #include "api/array_view.h" |
| 21 | #include "audio/utility/audio_frame_operations.h" |
| 22 | #include "call/rtp_transport_controller_send_interface.h" |
| 23 | #include "logging/rtc_event_log/rtc_event_log.h" |
Elad Alon | 4a87e1c | 2017-10-03 16:11:34 +0200 | [diff] [blame] | 24 | #include "logging/rtc_event_log/events/rtc_event_audio_playout.h" |
Elad Alon | 4a87e1c | 2017-10-03 16:11:34 +0200 | [diff] [blame] | 25 | #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 26 | #include "modules/audio_coding/codecs/audio_format_conversion.h" |
| 27 | #include "modules/audio_device/include/audio_device.h" |
| 28 | #include "modules/audio_processing/include/audio_processing.h" |
| 29 | #include "modules/include/module_common_types.h" |
| 30 | #include "modules/pacing/packet_router.h" |
| 31 | #include "modules/rtp_rtcp/include/receive_statistics.h" |
| 32 | #include "modules/rtp_rtcp/include/rtp_payload_registry.h" |
| 33 | #include "modules/rtp_rtcp/include/rtp_receiver.h" |
| 34 | #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| 35 | #include "modules/rtp_rtcp/source/rtp_receiver_strategy.h" |
| 36 | #include "modules/utility/include/process_thread.h" |
| 37 | #include "rtc_base/checks.h" |
| 38 | #include "rtc_base/criticalsection.h" |
| 39 | #include "rtc_base/format_macros.h" |
| 40 | #include "rtc_base/location.h" |
| 41 | #include "rtc_base/logging.h" |
Elad Alon | 4a87e1c | 2017-10-03 16:11:34 +0200 | [diff] [blame] | 42 | #include "rtc_base/ptr_util.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 43 | #include "rtc_base/rate_limiter.h" |
| 44 | #include "rtc_base/task_queue.h" |
| 45 | #include "rtc_base/thread_checker.h" |
| 46 | #include "rtc_base/timeutils.h" |
| 47 | #include "system_wrappers/include/field_trial.h" |
henrika | 4580217 | 2017-09-28 09:39:34 +0200 | [diff] [blame] | 48 | #include "system_wrappers/include/metrics.h" |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 49 | |
andrew@webrtc.org | 50419b0 | 2012-11-14 19:07:54 +0000 | [diff] [blame] | 50 | namespace webrtc { |
| 51 | namespace voe { |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 52 | |
kwiberg | c8d071e | 2016-04-06 12:22:38 -0700 | [diff] [blame] | 53 | namespace { |
| 54 | |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 55 | constexpr double kAudioSampleDurationSeconds = 0.01; |
Erik Språng | 737336d | 2016-07-29 12:59:36 +0200 | [diff] [blame] | 56 | constexpr int64_t kMaxRetransmissionWindowMs = 1000; |
| 57 | constexpr int64_t kMinRetransmissionWindowMs = 30; |
| 58 | |
Fredrik Solenberg | 55900fd | 2017-11-23 20:22:55 +0100 | [diff] [blame] | 59 | // Video Sync. |
| 60 | constexpr int kVoiceEngineMinMinPlayoutDelayMs = 0; |
| 61 | constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000; |
| 62 | |
kwiberg | c8d071e | 2016-04-06 12:22:38 -0700 | [diff] [blame] | 63 | } // namespace |
| 64 | |
solenberg | 8842c3e | 2016-03-11 03:06:41 -0800 | [diff] [blame] | 65 | const int kTelephoneEventAttenuationdB = 10; |
| 66 | |
ivoc | 14d5dbe | 2016-07-04 07:06:55 -0700 | [diff] [blame] | 67 | class RtcEventLogProxy final : public webrtc::RtcEventLog { |
| 68 | public: |
| 69 | RtcEventLogProxy() : event_log_(nullptr) {} |
| 70 | |
Bjorn Terelius | de93943 | 2017-11-20 17:38:14 +0100 | [diff] [blame] | 71 | bool StartLogging(std::unique_ptr<RtcEventLogOutput> output, |
| 72 | int64_t output_period_ms) override { |
Elad Alon | 83ccca1 | 2017-10-04 13:18:26 +0200 | [diff] [blame] | 73 | RTC_NOTREACHED(); |
| 74 | return false; |
| 75 | } |
| 76 | |
ivoc | 14d5dbe | 2016-07-04 07:06:55 -0700 | [diff] [blame] | 77 | void StopLogging() override { RTC_NOTREACHED(); } |
| 78 | |
Elad Alon | 4a87e1c | 2017-10-03 16:11:34 +0200 | [diff] [blame] | 79 | void Log(std::unique_ptr<RtcEvent> event) override { |
| 80 | rtc::CritScope lock(&crit_); |
| 81 | if (event_log_) { |
| 82 | event_log_->Log(std::move(event)); |
| 83 | } |
| 84 | } |
| 85 | |
ivoc | 14d5dbe | 2016-07-04 07:06:55 -0700 | [diff] [blame] | 86 | void SetEventLog(RtcEventLog* event_log) { |
| 87 | rtc::CritScope lock(&crit_); |
| 88 | event_log_ = event_log; |
| 89 | } |
| 90 | |
| 91 | private: |
| 92 | rtc::CriticalSection crit_; |
danilchap | a37de39 | 2017-09-09 04:17:22 -0700 | [diff] [blame] | 93 | RtcEventLog* event_log_ RTC_GUARDED_BY(crit_); |
ivoc | 14d5dbe | 2016-07-04 07:06:55 -0700 | [diff] [blame] | 94 | RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogProxy); |
| 95 | }; |
| 96 | |
michaelt | 9332b7d | 2016-11-30 07:51:13 -0800 | [diff] [blame] | 97 | class RtcpRttStatsProxy final : public RtcpRttStats { |
| 98 | public: |
| 99 | RtcpRttStatsProxy() : rtcp_rtt_stats_(nullptr) {} |
| 100 | |
| 101 | void OnRttUpdate(int64_t rtt) override { |
| 102 | rtc::CritScope lock(&crit_); |
| 103 | if (rtcp_rtt_stats_) |
| 104 | rtcp_rtt_stats_->OnRttUpdate(rtt); |
| 105 | } |
| 106 | |
| 107 | int64_t LastProcessedRtt() const override { |
| 108 | rtc::CritScope lock(&crit_); |
| 109 | if (!rtcp_rtt_stats_) |
| 110 | return 0; |
| 111 | return rtcp_rtt_stats_->LastProcessedRtt(); |
| 112 | } |
| 113 | |
| 114 | void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) { |
| 115 | rtc::CritScope lock(&crit_); |
| 116 | rtcp_rtt_stats_ = rtcp_rtt_stats; |
| 117 | } |
| 118 | |
| 119 | private: |
| 120 | rtc::CriticalSection crit_; |
danilchap | a37de39 | 2017-09-09 04:17:22 -0700 | [diff] [blame] | 121 | RtcpRttStats* rtcp_rtt_stats_ RTC_GUARDED_BY(crit_); |
michaelt | 9332b7d | 2016-11-30 07:51:13 -0800 | [diff] [blame] | 122 | RTC_DISALLOW_COPY_AND_ASSIGN(RtcpRttStatsProxy); |
| 123 | }; |
| 124 | |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 125 | class TransportFeedbackProxy : public TransportFeedbackObserver { |
| 126 | public: |
| 127 | TransportFeedbackProxy() : feedback_observer_(nullptr) { |
| 128 | pacer_thread_.DetachFromThread(); |
| 129 | network_thread_.DetachFromThread(); |
| 130 | } |
| 131 | |
| 132 | void SetTransportFeedbackObserver( |
| 133 | TransportFeedbackObserver* feedback_observer) { |
| 134 | RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| 135 | rtc::CritScope lock(&crit_); |
| 136 | feedback_observer_ = feedback_observer; |
| 137 | } |
| 138 | |
| 139 | // Implements TransportFeedbackObserver. |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 140 | void AddPacket(uint32_t ssrc, |
| 141 | uint16_t sequence_number, |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 142 | size_t length, |
philipel | 8aadd50 | 2017-02-23 02:56:13 -0800 | [diff] [blame] | 143 | const PacedPacketInfo& pacing_info) override { |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 144 | RTC_DCHECK(pacer_thread_.CalledOnValidThread()); |
| 145 | rtc::CritScope lock(&crit_); |
| 146 | if (feedback_observer_) |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 147 | feedback_observer_->AddPacket(ssrc, sequence_number, length, pacing_info); |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 148 | } |
philipel | 8aadd50 | 2017-02-23 02:56:13 -0800 | [diff] [blame] | 149 | |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 150 | void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override { |
| 151 | RTC_DCHECK(network_thread_.CalledOnValidThread()); |
| 152 | rtc::CritScope lock(&crit_); |
michaelt | 9960bb1 | 2016-10-18 09:40:34 -0700 | [diff] [blame] | 153 | if (feedback_observer_) |
| 154 | feedback_observer_->OnTransportFeedback(feedback); |
Stefan Holmer | 60e4346 | 2016-09-07 09:58:20 +0200 | [diff] [blame] | 155 | } |
elad.alon | f949000 | 2017-03-06 05:32:21 -0800 | [diff] [blame] | 156 | std::vector<PacketFeedback> GetTransportFeedbackVector() const override { |
Stefan Holmer | 60e4346 | 2016-09-07 09:58:20 +0200 | [diff] [blame] | 157 | RTC_NOTREACHED(); |
elad.alon | f949000 | 2017-03-06 05:32:21 -0800 | [diff] [blame] | 158 | return std::vector<PacketFeedback>(); |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 159 | } |
| 160 | |
| 161 | private: |
| 162 | rtc::CriticalSection crit_; |
| 163 | rtc::ThreadChecker thread_checker_; |
| 164 | rtc::ThreadChecker pacer_thread_; |
| 165 | rtc::ThreadChecker network_thread_; |
danilchap | a37de39 | 2017-09-09 04:17:22 -0700 | [diff] [blame] | 166 | TransportFeedbackObserver* feedback_observer_ RTC_GUARDED_BY(&crit_); |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 167 | }; |
| 168 | |
| 169 | class TransportSequenceNumberProxy : public TransportSequenceNumberAllocator { |
| 170 | public: |
| 171 | TransportSequenceNumberProxy() : seq_num_allocator_(nullptr) { |
| 172 | pacer_thread_.DetachFromThread(); |
| 173 | } |
| 174 | |
| 175 | void SetSequenceNumberAllocator( |
| 176 | TransportSequenceNumberAllocator* seq_num_allocator) { |
| 177 | RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| 178 | rtc::CritScope lock(&crit_); |
| 179 | seq_num_allocator_ = seq_num_allocator; |
| 180 | } |
| 181 | |
| 182 | // Implements TransportSequenceNumberAllocator. |
| 183 | uint16_t AllocateSequenceNumber() override { |
| 184 | RTC_DCHECK(pacer_thread_.CalledOnValidThread()); |
| 185 | rtc::CritScope lock(&crit_); |
| 186 | if (!seq_num_allocator_) |
| 187 | return 0; |
| 188 | return seq_num_allocator_->AllocateSequenceNumber(); |
| 189 | } |
| 190 | |
| 191 | private: |
| 192 | rtc::CriticalSection crit_; |
| 193 | rtc::ThreadChecker thread_checker_; |
| 194 | rtc::ThreadChecker pacer_thread_; |
danilchap | a37de39 | 2017-09-09 04:17:22 -0700 | [diff] [blame] | 195 | TransportSequenceNumberAllocator* seq_num_allocator_ RTC_GUARDED_BY(&crit_); |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 196 | }; |
| 197 | |
| 198 | class RtpPacketSenderProxy : public RtpPacketSender { |
| 199 | public: |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 200 | RtpPacketSenderProxy() : rtp_packet_sender_(nullptr) {} |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 201 | |
| 202 | void SetPacketSender(RtpPacketSender* rtp_packet_sender) { |
| 203 | RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| 204 | rtc::CritScope lock(&crit_); |
| 205 | rtp_packet_sender_ = rtp_packet_sender; |
| 206 | } |
| 207 | |
| 208 | // Implements RtpPacketSender. |
| 209 | void InsertPacket(Priority priority, |
| 210 | uint32_t ssrc, |
| 211 | uint16_t sequence_number, |
| 212 | int64_t capture_time_ms, |
| 213 | size_t bytes, |
| 214 | bool retransmission) override { |
| 215 | rtc::CritScope lock(&crit_); |
| 216 | if (rtp_packet_sender_) { |
| 217 | rtp_packet_sender_->InsertPacket(priority, ssrc, sequence_number, |
| 218 | capture_time_ms, bytes, retransmission); |
| 219 | } |
| 220 | } |
| 221 | |
Alex Narest | 78609d5 | 2017-10-20 10:37:47 +0200 | [diff] [blame] | 222 | void SetAccountForAudioPackets(bool account_for_audio) override { |
| 223 | RTC_NOTREACHED(); |
| 224 | } |
| 225 | |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 226 | private: |
| 227 | rtc::ThreadChecker thread_checker_; |
| 228 | rtc::CriticalSection crit_; |
danilchap | a37de39 | 2017-09-09 04:17:22 -0700 | [diff] [blame] | 229 | RtpPacketSender* rtp_packet_sender_ RTC_GUARDED_BY(&crit_); |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 230 | }; |
| 231 | |
mflodman@webrtc.org | 0a7d4ee | 2015-02-17 12:57:14 +0000 | [diff] [blame] | 232 | class VoERtcpObserver : public RtcpBandwidthObserver { |
minyue@webrtc.org | c1a40a7 | 2014-05-28 09:52:06 +0000 | [diff] [blame] | 233 | public: |
stefan | 7de8d64 | 2017-02-07 07:14:08 -0800 | [diff] [blame] | 234 | explicit VoERtcpObserver(Channel* owner) |
| 235 | : owner_(owner), bandwidth_observer_(nullptr) {} |
mflodman@webrtc.org | 0a7d4ee | 2015-02-17 12:57:14 +0000 | [diff] [blame] | 236 | virtual ~VoERtcpObserver() {} |
minyue@webrtc.org | c1a40a7 | 2014-05-28 09:52:06 +0000 | [diff] [blame] | 237 | |
stefan | 7de8d64 | 2017-02-07 07:14:08 -0800 | [diff] [blame] | 238 | void SetBandwidthObserver(RtcpBandwidthObserver* bandwidth_observer) { |
| 239 | rtc::CritScope lock(&crit_); |
| 240 | bandwidth_observer_ = bandwidth_observer; |
| 241 | } |
| 242 | |
mflodman@webrtc.org | 0a7d4ee | 2015-02-17 12:57:14 +0000 | [diff] [blame] | 243 | void OnReceivedEstimatedBitrate(uint32_t bitrate) override { |
stefan | 7de8d64 | 2017-02-07 07:14:08 -0800 | [diff] [blame] | 244 | rtc::CritScope lock(&crit_); |
| 245 | if (bandwidth_observer_) { |
| 246 | bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate); |
| 247 | } |
mflodman@webrtc.org | 0a7d4ee | 2015-02-17 12:57:14 +0000 | [diff] [blame] | 248 | } |
| 249 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 250 | void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks, |
| 251 | int64_t rtt, |
| 252 | int64_t now_ms) override { |
stefan | 7de8d64 | 2017-02-07 07:14:08 -0800 | [diff] [blame] | 253 | { |
| 254 | rtc::CritScope lock(&crit_); |
| 255 | if (bandwidth_observer_) { |
| 256 | bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, rtt, |
| 257 | now_ms); |
| 258 | } |
| 259 | } |
mflodman@webrtc.org | 0a7d4ee | 2015-02-17 12:57:14 +0000 | [diff] [blame] | 260 | // TODO(mflodman): Do we need to aggregate reports here or can we jut send |
| 261 | // what we get? I.e. do we ever get multiple reports bundled into one RTCP |
| 262 | // report for VoiceEngine? |
| 263 | if (report_blocks.empty()) |
| 264 | return; |
| 265 | |
| 266 | int fraction_lost_aggregate = 0; |
| 267 | int total_number_of_packets = 0; |
| 268 | |
| 269 | // If receiving multiple report blocks, calculate the weighted average based |
| 270 | // on the number of packets a report refers to. |
| 271 | for (ReportBlockList::const_iterator block_it = report_blocks.begin(); |
| 272 | block_it != report_blocks.end(); ++block_it) { |
| 273 | // Find the previous extended high sequence number for this remote SSRC, |
| 274 | // to calculate the number of RTP packets this report refers to. Ignore if |
| 275 | // we haven't seen this SSRC before. |
| 276 | std::map<uint32_t, uint32_t>::iterator seq_num_it = |
srte | 3e69e5c | 2017-08-09 06:13:45 -0700 | [diff] [blame] | 277 | extended_max_sequence_number_.find(block_it->source_ssrc); |
mflodman@webrtc.org | 0a7d4ee | 2015-02-17 12:57:14 +0000 | [diff] [blame] | 278 | int number_of_packets = 0; |
| 279 | if (seq_num_it != extended_max_sequence_number_.end()) { |
srte | 3e69e5c | 2017-08-09 06:13:45 -0700 | [diff] [blame] | 280 | number_of_packets = |
| 281 | block_it->extended_highest_sequence_number - seq_num_it->second; |
mflodman@webrtc.org | 0a7d4ee | 2015-02-17 12:57:14 +0000 | [diff] [blame] | 282 | } |
srte | 3e69e5c | 2017-08-09 06:13:45 -0700 | [diff] [blame] | 283 | fraction_lost_aggregate += number_of_packets * block_it->fraction_lost; |
mflodman@webrtc.org | 0a7d4ee | 2015-02-17 12:57:14 +0000 | [diff] [blame] | 284 | total_number_of_packets += number_of_packets; |
| 285 | |
srte | 3e69e5c | 2017-08-09 06:13:45 -0700 | [diff] [blame] | 286 | extended_max_sequence_number_[block_it->source_ssrc] = |
| 287 | block_it->extended_highest_sequence_number; |
mflodman@webrtc.org | 0a7d4ee | 2015-02-17 12:57:14 +0000 | [diff] [blame] | 288 | } |
| 289 | int weighted_fraction_lost = 0; |
| 290 | if (total_number_of_packets > 0) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 291 | weighted_fraction_lost = |
| 292 | (fraction_lost_aggregate + total_number_of_packets / 2) / |
| 293 | total_number_of_packets; |
mflodman@webrtc.org | 0a7d4ee | 2015-02-17 12:57:14 +0000 | [diff] [blame] | 294 | } |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 295 | owner_->OnUplinkPacketLossRate(weighted_fraction_lost / 255.0f); |
minyue@webrtc.org | c1a40a7 | 2014-05-28 09:52:06 +0000 | [diff] [blame] | 296 | } |
| 297 | |
| 298 | private: |
| 299 | Channel* owner_; |
mflodman@webrtc.org | 0a7d4ee | 2015-02-17 12:57:14 +0000 | [diff] [blame] | 300 | // Maps remote side ssrc to extended highest sequence number received. |
| 301 | std::map<uint32_t, uint32_t> extended_max_sequence_number_; |
stefan | 7de8d64 | 2017-02-07 07:14:08 -0800 | [diff] [blame] | 302 | rtc::CriticalSection crit_; |
danilchap | a37de39 | 2017-09-09 04:17:22 -0700 | [diff] [blame] | 303 | RtcpBandwidthObserver* bandwidth_observer_ RTC_GUARDED_BY(crit_); |
minyue@webrtc.org | c1a40a7 | 2014-05-28 09:52:06 +0000 | [diff] [blame] | 304 | }; |
| 305 | |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 306 | class Channel::ProcessAndEncodeAudioTask : public rtc::QueuedTask { |
| 307 | public: |
| 308 | ProcessAndEncodeAudioTask(std::unique_ptr<AudioFrame> audio_frame, |
| 309 | Channel* channel) |
| 310 | : audio_frame_(std::move(audio_frame)), channel_(channel) { |
| 311 | RTC_DCHECK(channel_); |
| 312 | } |
| 313 | |
| 314 | private: |
| 315 | bool Run() override { |
| 316 | RTC_DCHECK_RUN_ON(channel_->encoder_queue_); |
| 317 | channel_->ProcessAndEncodeAudioOnTaskQueue(audio_frame_.get()); |
| 318 | return true; |
| 319 | } |
| 320 | |
| 321 | std::unique_ptr<AudioFrame> audio_frame_; |
| 322 | Channel* const channel_; |
| 323 | }; |
| 324 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 325 | int32_t Channel::SendData(FrameType frameType, |
| 326 | uint8_t payloadType, |
| 327 | uint32_t timeStamp, |
| 328 | const uint8_t* payloadData, |
| 329 | size_t payloadSize, |
| 330 | const RTPFragmentationHeader* fragmentation) { |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 331 | RTC_DCHECK_RUN_ON(encoder_queue_); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 332 | if (_includeAudioLevelIndication) { |
| 333 | // Store current audio level in the RTP/RTCP module. |
| 334 | // The level will be used in combination with voice-activity state |
| 335 | // (frameType) to add an RTP header extension |
henrik.lundin | 5049942 | 2016-11-29 04:26:24 -0800 | [diff] [blame] | 336 | _rtpRtcpModule->SetAudioLevel(rms_level_.Average()); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 337 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 338 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 339 | // Push data from ACM to RTP/RTCP-module to deliver audio frame for |
| 340 | // packetization. |
| 341 | // This call will trigger Transport::SendPacket() from the RTP/RTCP module. |
Sergey Ulanov | 525df3f | 2016-08-02 17:46:41 -0700 | [diff] [blame] | 342 | if (!_rtpRtcpModule->SendOutgoingData( |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 343 | (FrameType&)frameType, payloadType, timeStamp, |
| 344 | // Leaving the time when this frame was |
| 345 | // received from the capture device as |
| 346 | // undefined for voice for now. |
Sergey Ulanov | 525df3f | 2016-08-02 17:46:41 -0700 | [diff] [blame] | 347 | -1, payloadData, payloadSize, fragmentation, nullptr, nullptr)) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 348 | RTC_LOG(LS_ERROR) |
| 349 | << "Channel::SendData() failed to send data to RTP/RTCP module"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 350 | return -1; |
| 351 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 352 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 353 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 354 | } |
| 355 | |
stefan | 1d8a506 | 2015-10-02 03:39:33 -0700 | [diff] [blame] | 356 | bool Channel::SendRtp(const uint8_t* data, |
| 357 | size_t len, |
| 358 | const PacketOptions& options) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 359 | rtc::CritScope cs(&_callbackCritSect); |
wu@webrtc.org | fb648da | 2013-10-18 21:10:51 +0000 | [diff] [blame] | 360 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 361 | if (_transportPtr == NULL) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 362 | RTC_LOG(LS_ERROR) |
| 363 | << "Channel::SendPacket() failed to send RTP packet due to" |
| 364 | << " invalid transport object"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 365 | return false; |
| 366 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 367 | |
Fredrik Solenberg | a8b7c7f | 2018-01-17 11:18:31 +0100 | [diff] [blame] | 368 | if (!_transportPtr->SendRtp(data, len, options)) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 369 | RTC_LOG(LS_ERROR) << "Channel::SendPacket() RTP transmission failed"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 370 | return false; |
| 371 | } |
| 372 | return true; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 373 | } |
| 374 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 375 | bool Channel::SendRtcp(const uint8_t* data, size_t len) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 376 | rtc::CritScope cs(&_callbackCritSect); |
| 377 | if (_transportPtr == NULL) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 378 | RTC_LOG(LS_ERROR) << "Channel::SendRtcp() failed to send RTCP packet due to" |
| 379 | << " invalid transport object"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 380 | return false; |
| 381 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 382 | |
Fredrik Solenberg | a8b7c7f | 2018-01-17 11:18:31 +0100 | [diff] [blame] | 383 | int n = _transportPtr->SendRtcp(data, len); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 384 | if (n < 0) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 385 | RTC_LOG(LS_ERROR) << "Channel::SendRtcp() transmission failed"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 386 | return false; |
| 387 | } |
| 388 | return true; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 389 | } |
| 390 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 391 | void Channel::OnIncomingSSRCChanged(uint32_t ssrc) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 392 | // Update ssrc so that NTP for AV sync can be updated. |
| 393 | _rtpRtcpModule->SetRemoteSSRC(ssrc); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 394 | } |
| 395 | |
Peter Boström | ac547a6 | 2015-09-17 23:03:57 +0200 | [diff] [blame] | 396 | void Channel::OnIncomingCSRCChanged(uint32_t CSRC, bool added) { |
Sam Zackrisson | ecc51e9 | 2017-10-02 14:32:33 +0200 | [diff] [blame] | 397 | // TODO(saza): remove. |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 398 | } |
| 399 | |
Karl Wiberg | c62f6c7 | 2017-10-04 12:38:53 +0200 | [diff] [blame] | 400 | int32_t Channel::OnInitializeDecoder(int payload_type, |
| 401 | const SdpAudioFormat& audio_format, |
| 402 | uint32_t rate) { |
| 403 | if (!audio_coding_->RegisterReceiveCodec(payload_type, audio_format)) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 404 | RTC_LOG(LS_WARNING) << "Channel::OnInitializeDecoder() invalid codec (pt=" |
| 405 | << payload_type << ", " << audio_format |
| 406 | << ") received -1"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 407 | return -1; |
| 408 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 409 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 410 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 411 | } |
| 412 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 413 | int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData, |
| 414 | size_t payloadSize, |
| 415 | const WebRtcRTPHeader* rtpHeader) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 416 | if (!channel_state_.Get().playing) { |
| 417 | // Avoid inserting into NetEQ when we are not playing. Count the |
| 418 | // packet as discarded. |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 419 | return 0; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 420 | } |
| 421 | |
| 422 | // Push the incoming payload (parsed and ready for decoding) into the ACM |
| 423 | if (audio_coding_->IncomingPacket(payloadData, payloadSize, *rtpHeader) != |
| 424 | 0) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 425 | RTC_LOG(LS_ERROR) |
| 426 | << "Channel::OnReceivedPayloadData() unable to push data to the ACM"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 427 | return -1; |
| 428 | } |
| 429 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 430 | int64_t round_trip_time = 0; |
| 431 | _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time, NULL, NULL, |
| 432 | NULL); |
| 433 | |
| 434 | std::vector<uint16_t> nack_list = audio_coding_->GetNackList(round_trip_time); |
| 435 | if (!nack_list.empty()) { |
| 436 | // Can't use nack_list.data() since it's not supported by all |
| 437 | // compilers. |
| 438 | ResendPackets(&(nack_list[0]), static_cast<int>(nack_list.size())); |
| 439 | } |
| 440 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 441 | } |
| 442 | |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 443 | AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo( |
| 444 | int sample_rate_hz, |
| 445 | AudioFrame* audio_frame) { |
| 446 | audio_frame->sample_rate_hz_ = sample_rate_hz; |
| 447 | |
ivoc | 14d5dbe | 2016-07-04 07:06:55 -0700 | [diff] [blame] | 448 | unsigned int ssrc; |
nisse | 7d59f6b | 2017-02-21 03:40:24 -0800 | [diff] [blame] | 449 | RTC_CHECK_EQ(GetRemoteSSRC(ssrc), 0); |
Elad Alon | 4a87e1c | 2017-10-03 16:11:34 +0200 | [diff] [blame] | 450 | event_log_proxy_->Log(rtc::MakeUnique<RtcEventAudioPlayout>(ssrc)); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 451 | // Get 10ms raw PCM data from the ACM (mixer limits output frequency) |
henrik.lundin | d4ccb00 | 2016-05-17 12:21:55 -0700 | [diff] [blame] | 452 | bool muted; |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 453 | if (audio_coding_->PlayoutData10Ms(audio_frame->sample_rate_hz_, audio_frame, |
henrik.lundin | d4ccb00 | 2016-05-17 12:21:55 -0700 | [diff] [blame] | 454 | &muted) == -1) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 455 | RTC_LOG(LS_ERROR) << "Channel::GetAudioFrame() PlayoutData10Ms() failed!"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 456 | // In all likelihood, the audio in this frame is garbage. We return an |
| 457 | // error so that the audio mixer module doesn't add it to the mix. As |
| 458 | // a result, it won't be played out and the actions skipped here are |
| 459 | // irrelevant. |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 460 | return AudioMixer::Source::AudioFrameInfo::kError; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 461 | } |
henrik.lundin | a89ab96 | 2016-05-18 08:52:45 -0700 | [diff] [blame] | 462 | |
| 463 | if (muted) { |
| 464 | // TODO(henrik.lundin): We should be able to do better than this. But we |
| 465 | // will have to go through all the cases below where the audio samples may |
| 466 | // be used, and handle the muted case in some way. |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 467 | AudioFrameOperations::Mute(audio_frame); |
henrik.lundin | a89ab96 | 2016-05-18 08:52:45 -0700 | [diff] [blame] | 468 | } |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 469 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 470 | { |
| 471 | // Pass the audio buffers to an optional sink callback, before applying |
| 472 | // scaling/panning, as that applies to the mix operation. |
| 473 | // External recipients of the audio (e.g. via AudioTrack), will do their |
| 474 | // own mixing/dynamic processing. |
| 475 | rtc::CritScope cs(&_callbackCritSect); |
| 476 | if (audio_sink_) { |
| 477 | AudioSinkInterface::Data data( |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 478 | audio_frame->data(), audio_frame->samples_per_channel_, |
| 479 | audio_frame->sample_rate_hz_, audio_frame->num_channels_, |
| 480 | audio_frame->timestamp_); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 481 | audio_sink_->OnData(data); |
| 482 | } |
| 483 | } |
| 484 | |
| 485 | float output_gain = 1.0f; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 486 | { |
| 487 | rtc::CritScope cs(&volume_settings_critsect_); |
| 488 | output_gain = _outputGain; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 489 | } |
| 490 | |
| 491 | // Output volume scaling |
| 492 | if (output_gain < 0.99f || output_gain > 1.01f) { |
solenberg | 8d73f8c | 2017-03-08 01:52:20 -0800 | [diff] [blame] | 493 | // TODO(solenberg): Combine with mute state - this can cause clicks! |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 494 | AudioFrameOperations::ScaleWithSat(output_gain, audio_frame); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 495 | } |
| 496 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 497 | // Measure audio level (0-9) |
henrik.lundin | a89ab96 | 2016-05-18 08:52:45 -0700 | [diff] [blame] | 498 | // TODO(henrik.lundin) Use the |muted| information here too. |
zstein | 3c45186 | 2017-07-20 09:57:42 -0700 | [diff] [blame] | 499 | // TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| (see |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 500 | // https://crbug.com/webrtc/7517). |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 501 | _outputAudioLevel.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 502 | |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 503 | if (capture_start_rtp_time_stamp_ < 0 && audio_frame->timestamp_ != 0) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 504 | // The first frame with a valid rtp timestamp. |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 505 | capture_start_rtp_time_stamp_ = audio_frame->timestamp_; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 506 | } |
| 507 | |
| 508 | if (capture_start_rtp_time_stamp_ >= 0) { |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 509 | // audio_frame.timestamp_ should be valid from now on. |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 510 | |
| 511 | // Compute elapsed time. |
| 512 | int64_t unwrap_timestamp = |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 513 | rtp_ts_wraparound_handler_->Unwrap(audio_frame->timestamp_); |
| 514 | audio_frame->elapsed_time_ms_ = |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 515 | (unwrap_timestamp - capture_start_rtp_time_stamp_) / |
ossu | e280cde | 2016-10-12 11:04:10 -0700 | [diff] [blame] | 516 | (GetRtpTimestampRateHz() / 1000); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 517 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 518 | { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 519 | rtc::CritScope lock(&ts_stats_lock_); |
| 520 | // Compute ntp time. |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 521 | audio_frame->ntp_time_ms_ = |
| 522 | ntp_estimator_.Estimate(audio_frame->timestamp_); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 523 | // |ntp_time_ms_| won't be valid until at least 2 RTCP SRs are received. |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 524 | if (audio_frame->ntp_time_ms_ > 0) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 525 | // Compute |capture_start_ntp_time_ms_| so that |
| 526 | // |capture_start_ntp_time_ms_| + |elapsed_time_ms_| == |ntp_time_ms_| |
| 527 | capture_start_ntp_time_ms_ = |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 528 | audio_frame->ntp_time_ms_ - audio_frame->elapsed_time_ms_; |
andrew@webrtc.org | 60730cf | 2014-01-07 17:45:09 +0000 | [diff] [blame] | 529 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 530 | } |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 531 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 532 | |
Henrik Lundin | 9bde6b7 | 2017-11-02 15:01:56 +0100 | [diff] [blame] | 533 | { |
Henrik Lundin | abbff89 | 2017-11-29 09:14:04 +0100 | [diff] [blame] | 534 | RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.TargetJitterBufferDelayMs", |
| 535 | audio_coding_->TargetDelayMs()); |
Henrik Lundin | 9bde6b7 | 2017-11-02 15:01:56 +0100 | [diff] [blame] | 536 | const int jitter_buffer_delay = audio_coding_->FilteredCurrentDelayMs(); |
| 537 | rtc::CritScope lock(&video_sync_lock_); |
| 538 | RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDelayEstimateMs", |
| 539 | jitter_buffer_delay + playout_delay_ms_); |
| 540 | RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverJitterBufferDelayMs", |
| 541 | jitter_buffer_delay); |
| 542 | RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDeviceDelayMs", |
| 543 | playout_delay_ms_); |
| 544 | } |
| 545 | |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 546 | return muted ? AudioMixer::Source::AudioFrameInfo::kMuted |
| 547 | : AudioMixer::Source::AudioFrameInfo::kNormal; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 548 | } |
| 549 | |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 550 | int Channel::PreferredSampleRate() const { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 551 | // Return the bigger of playout and receive frequency in the ACM. |
solenberg | 2397b9a | 2017-09-22 06:48:10 -0700 | [diff] [blame] | 552 | return std::max(audio_coding_->ReceiveFrequency(), |
| 553 | audio_coding_->PlayoutFrequency()); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 554 | } |
| 555 | |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 556 | Channel::Channel(rtc::TaskQueue* encoder_queue, |
| 557 | ProcessThread* module_process_thread, |
| 558 | AudioDeviceModule* audio_device_module) |
| 559 | : Channel(module_process_thread, |
| 560 | audio_device_module, |
| 561 | 0, |
| 562 | false, |
| 563 | rtc::scoped_refptr<AudioDecoderFactory>()) { |
| 564 | RTC_DCHECK(encoder_queue); |
| 565 | encoder_queue_ = encoder_queue; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 566 | } |
| 567 | |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 568 | Channel::Channel(ProcessThread* module_process_thread, |
| 569 | AudioDeviceModule* audio_device_module, |
| 570 | size_t jitter_buffer_max_packets, |
| 571 | bool jitter_buffer_fast_playout, |
| 572 | rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) |
| 573 | : event_log_proxy_(new RtcEventLogProxy()), |
michaelt | 9332b7d | 2016-11-30 07:51:13 -0800 | [diff] [blame] | 574 | rtcp_rtt_stats_proxy_(new RtcpRttStatsProxy()), |
magjed | f3feeff | 2016-11-25 06:40:25 -0800 | [diff] [blame] | 575 | rtp_payload_registry_(new RTPPayloadRegistry()), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 576 | rtp_receive_statistics_( |
| 577 | ReceiveStatistics::Create(Clock::GetRealTimeClock())), |
| 578 | rtp_receiver_( |
| 579 | RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 580 | this, |
| 581 | this, |
| 582 | rtp_payload_registry_.get())), |
danilchap | 799a9d0 | 2016-09-22 03:36:27 -0700 | [diff] [blame] | 583 | telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 584 | _outputAudioLevel(), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 585 | _timeStamp(0), // This is just an offset, RTP module will add it's own |
| 586 | // random offset |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 587 | ntp_estimator_(Clock::GetRealTimeClock()), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 588 | playout_timestamp_rtp_(0), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 589 | playout_delay_ms_(0), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 590 | send_sequence_number_(0), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 591 | rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()), |
| 592 | capture_start_rtp_time_stamp_(-1), |
| 593 | capture_start_ntp_time_ms_(-1), |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 594 | _moduleProcessThreadPtr(module_process_thread), |
| 595 | _audioDeviceModulePtr(audio_device_module), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 596 | _transportPtr(NULL), |
solenberg | 1c2af8e | 2016-03-24 10:36:00 -0700 | [diff] [blame] | 597 | input_mute_(false), |
| 598 | previous_frame_muted_(false), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 599 | _outputGain(1.0f), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 600 | _includeAudioLevelIndication(false), |
nisse | 284542b | 2017-01-10 08:58:32 -0800 | [diff] [blame] | 601 | transport_overhead_per_packet_(0), |
| 602 | rtp_overhead_per_packet_(0), |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 603 | rtcp_observer_(new VoERtcpObserver(this)), |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 604 | associated_send_channel_(nullptr), |
stefan | bba9dec | 2016-02-01 04:39:55 -0800 | [diff] [blame] | 605 | feedback_observer_proxy_(new TransportFeedbackProxy()), |
| 606 | seq_num_allocator_proxy_(new TransportSequenceNumberProxy()), |
ossu | 29b1a8d | 2016-06-13 07:34:51 -0700 | [diff] [blame] | 607 | rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), |
Erik Språng | 737336d | 2016-07-29 12:59:36 +0200 | [diff] [blame] | 608 | retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), |
| 609 | kMaxRetransmissionWindowMs)), |
elad.alon | 2877048 | 2017-03-28 05:03:55 -0700 | [diff] [blame] | 610 | use_twcc_plr_for_ana_( |
| 611 | webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled") { |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 612 | RTC_DCHECK(module_process_thread); |
| 613 | RTC_DCHECK(audio_device_module); |
| 614 | AudioCodingModule::Config acm_config; |
| 615 | acm_config.decoder_factory = decoder_factory; |
| 616 | acm_config.neteq_config.max_packets_in_buffer = jitter_buffer_max_packets; |
| 617 | acm_config.neteq_config.enable_fast_accelerate = jitter_buffer_fast_playout; |
henrik.lundin | a89ab96 | 2016-05-18 08:52:45 -0700 | [diff] [blame] | 618 | acm_config.neteq_config.enable_muted_state = true; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 619 | audio_coding_.reset(AudioCodingModule::Create(acm_config)); |
Henrik Lundin | 64dad83 | 2015-05-11 12:44:23 +0200 | [diff] [blame] | 620 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 621 | _outputAudioLevel.Clear(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 622 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 623 | RtpRtcp::Configuration configuration; |
| 624 | configuration.audio = true; |
| 625 | configuration.outgoing_transport = this; |
michaelt | bf65be5 | 2016-12-15 06:24:49 -0800 | [diff] [blame] | 626 | configuration.overhead_observer = this; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 627 | configuration.receive_statistics = rtp_receive_statistics_.get(); |
| 628 | configuration.bandwidth_callback = rtcp_observer_.get(); |
stefan | bba9dec | 2016-02-01 04:39:55 -0800 | [diff] [blame] | 629 | if (pacing_enabled_) { |
| 630 | configuration.paced_sender = rtp_packet_sender_proxy_.get(); |
| 631 | configuration.transport_sequence_number_allocator = |
| 632 | seq_num_allocator_proxy_.get(); |
| 633 | configuration.transport_feedback_callback = feedback_observer_proxy_.get(); |
| 634 | } |
ivoc | 14d5dbe | 2016-07-04 07:06:55 -0700 | [diff] [blame] | 635 | configuration.event_log = &(*event_log_proxy_); |
michaelt | 9332b7d | 2016-11-30 07:51:13 -0800 | [diff] [blame] | 636 | configuration.rtt_stats = &(*rtcp_rtt_stats_proxy_); |
Erik Språng | 737336d | 2016-07-29 12:59:36 +0200 | [diff] [blame] | 637 | configuration.retransmission_rate_limiter = |
| 638 | retransmission_rate_limiter_.get(); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 639 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 640 | _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration)); |
Peter Boström | 3dd5d1d | 2016-02-25 16:56:48 +0100 | [diff] [blame] | 641 | _rtpRtcpModule->SetSendingMediaStatus(false); |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 642 | |
| 643 | Init(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 644 | } |
| 645 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 646 | Channel::~Channel() { |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 647 | Terminate(); |
tommi | 0a2391f | 2017-03-21 02:31:51 -0700 | [diff] [blame] | 648 | RTC_DCHECK(!channel_state_.Get().sending); |
| 649 | RTC_DCHECK(!channel_state_.Get().playing); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 650 | } |
| 651 | |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 652 | void Channel::Init() { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 653 | channel_state_.Reset(); |
henrika@webrtc.org | 944cbeb | 2014-03-18 10:32:33 +0000 | [diff] [blame] | 654 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 655 | // --- Add modules to process thread (for periodic schedulation) |
tommi | dea489f | 2017-03-03 03:20:24 -0800 | [diff] [blame] | 656 | _moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 657 | |
| 658 | // --- ACM initialization |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 659 | int error = audio_coding_->InitializeReceiver(); |
| 660 | RTC_DCHECK_EQ(0, error); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 661 | |
| 662 | // --- RTP/RTCP module initialization |
| 663 | |
| 664 | // Ensure that RTCP is enabled by default for the created channel. |
| 665 | // Note that, the module will keep generating RTCP until it is explicitly |
| 666 | // disabled by the user. |
| 667 | // After StopListen (when no sockets exists), RTCP packets will no longer |
| 668 | // be transmitted since the Transport object will then be invalid. |
danilchap | 799a9d0 | 2016-09-22 03:36:27 -0700 | [diff] [blame] | 669 | telephone_event_handler_->SetTelephoneEventForwardToDecoder(true); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 670 | // RTCP is enabled by default. |
| 671 | _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 672 | |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 673 | // --- Register all permanent callbacks |
| 674 | error = audio_coding_->RegisterTransportCallback(this); |
| 675 | RTC_DCHECK_EQ(0, error); |
kwiberg | 1c07c70 | 2017-03-27 07:15:49 -0700 | [diff] [blame] | 676 | } |
| 677 | |
tommi | 0a2391f | 2017-03-21 02:31:51 -0700 | [diff] [blame] | 678 | void Channel::Terminate() { |
| 679 | RTC_DCHECK(construction_thread_.CalledOnValidThread()); |
| 680 | // Must be called on the same thread as Init(). |
tommi | 0a2391f | 2017-03-21 02:31:51 -0700 | [diff] [blame] | 681 | rtp_receive_statistics_->RegisterRtcpStatisticsCallback(NULL); |
| 682 | |
| 683 | StopSend(); |
| 684 | StopPlayout(); |
| 685 | |
tommi | 0a2391f | 2017-03-21 02:31:51 -0700 | [diff] [blame] | 686 | // The order to safely shutdown modules in a channel is: |
| 687 | // 1. De-register callbacks in modules |
| 688 | // 2. De-register modules in process thread |
| 689 | // 3. Destroy modules |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 690 | int error = audio_coding_->RegisterTransportCallback(NULL); |
| 691 | RTC_DCHECK_EQ(0, error); |
tommi | 0a2391f | 2017-03-21 02:31:51 -0700 | [diff] [blame] | 692 | |
tommi | 0a2391f | 2017-03-21 02:31:51 -0700 | [diff] [blame] | 693 | // De-register modules in process thread |
| 694 | if (_moduleProcessThreadPtr) |
| 695 | _moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get()); |
| 696 | |
| 697 | // End of modules shutdown |
| 698 | } |
| 699 | |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 700 | void Channel::SetSink(AudioSinkInterface* sink) { |
tommi | 31fc21f | 2016-01-21 10:37:37 -0800 | [diff] [blame] | 701 | rtc::CritScope cs(&_callbackCritSect); |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 702 | audio_sink_ = sink; |
ossu | 29b1a8d | 2016-06-13 07:34:51 -0700 | [diff] [blame] | 703 | } |
| 704 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 705 | int32_t Channel::StartPlayout() { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 706 | if (channel_state_.Get().playing) { |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 707 | return 0; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 708 | } |
| 709 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 710 | channel_state_.SetPlaying(true); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 711 | |
| 712 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 713 | } |
| 714 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 715 | int32_t Channel::StopPlayout() { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 716 | if (!channel_state_.Get().playing) { |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 717 | return 0; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 718 | } |
| 719 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 720 | channel_state_.SetPlaying(false); |
| 721 | _outputAudioLevel.Clear(); |
| 722 | |
| 723 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 724 | } |
| 725 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 726 | int32_t Channel::StartSend() { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 727 | if (channel_state_.Get().sending) { |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 728 | return 0; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 729 | } |
| 730 | channel_state_.SetSending(true); |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 731 | |
solenberg | 08b19df | 2017-02-15 00:42:31 -0800 | [diff] [blame] | 732 | // Resume the previous sequence number which was reset by StopSend(). This |
| 733 | // needs to be done before |sending| is set to true on the RTP/RTCP module. |
| 734 | if (send_sequence_number_) { |
| 735 | _rtpRtcpModule->SetSequenceNumber(send_sequence_number_); |
| 736 | } |
Peter Boström | 3dd5d1d | 2016-02-25 16:56:48 +0100 | [diff] [blame] | 737 | _rtpRtcpModule->SetSendingMediaStatus(true); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 738 | if (_rtpRtcpModule->SetSendingStatus(true) != 0) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 739 | RTC_LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending"; |
Peter Boström | 3dd5d1d | 2016-02-25 16:56:48 +0100 | [diff] [blame] | 740 | _rtpRtcpModule->SetSendingMediaStatus(false); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 741 | rtc::CritScope cs(&_callbackCritSect); |
henrika@webrtc.org | 944cbeb | 2014-03-18 10:32:33 +0000 | [diff] [blame] | 742 | channel_state_.SetSending(false); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 743 | return -1; |
| 744 | } |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 745 | { |
| 746 | // It is now OK to start posting tasks to the encoder task queue. |
| 747 | rtc::CritScope cs(&encoder_queue_lock_); |
| 748 | encoder_queue_is_active_ = true; |
| 749 | } |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 750 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 751 | } |
| 752 | |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 753 | void Channel::StopSend() { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 754 | if (!channel_state_.Get().sending) { |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 755 | return; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 756 | } |
| 757 | channel_state_.SetSending(false); |
| 758 | |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 759 | // Post a task to the encoder thread which sets an event when the task is |
| 760 | // executed. We know that no more encoding tasks will be added to the task |
| 761 | // queue for this channel since sending is now deactivated. It means that, |
| 762 | // if we wait for the event to bet set, we know that no more pending tasks |
| 763 | // exists and it is therfore guaranteed that the task queue will never try |
| 764 | // to acccess and invalid channel object. |
| 765 | RTC_DCHECK(encoder_queue_); |
henrika | 4515fa0 | 2017-05-03 08:30:15 -0700 | [diff] [blame] | 766 | |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 767 | rtc::Event flush(false, false); |
henrika | 4515fa0 | 2017-05-03 08:30:15 -0700 | [diff] [blame] | 768 | { |
| 769 | // Clear |encoder_queue_is_active_| under lock to prevent any other tasks |
| 770 | // than this final "flush task" to be posted on the queue. |
| 771 | rtc::CritScope cs(&encoder_queue_lock_); |
| 772 | encoder_queue_is_active_ = false; |
| 773 | encoder_queue_->PostTask([&flush]() { flush.Set(); }); |
| 774 | } |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 775 | flush.Wait(rtc::Event::kForever); |
| 776 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 777 | // Store the sequence number to be able to pick up the same sequence for |
| 778 | // the next StartSend(). This is needed for restarting device, otherwise |
| 779 | // it might cause libSRTP to complain about packets being replayed. |
| 780 | // TODO(xians): Remove this workaround after RtpRtcpModule's refactoring |
| 781 | // CL is landed. See issue |
| 782 | // https://code.google.com/p/webrtc/issues/detail?id=2111 . |
| 783 | send_sequence_number_ = _rtpRtcpModule->SequenceNumber(); |
| 784 | |
| 785 | // Reset sending SSRC and sequence number and triggers direct transmission |
| 786 | // of RTCP BYE |
| 787 | if (_rtpRtcpModule->SetSendingStatus(false) == -1) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 788 | RTC_LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 789 | } |
Peter Boström | 3dd5d1d | 2016-02-25 16:56:48 +0100 | [diff] [blame] | 790 | _rtpRtcpModule->SetSendingMediaStatus(false); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 791 | } |
| 792 | |
ossu | 1ffbd6c | 2017-04-06 12:05:04 -0700 | [diff] [blame] | 793 | bool Channel::SetEncoder(int payload_type, |
| 794 | std::unique_ptr<AudioEncoder> encoder) { |
| 795 | RTC_DCHECK_GE(payload_type, 0); |
| 796 | RTC_DCHECK_LE(payload_type, 127); |
ossu | 76d29f9 | 2017-06-09 07:30:13 -0700 | [diff] [blame] | 797 | // TODO(ossu): Make CodecInsts up, for now: one for the RTP/RTCP module and |
| 798 | // one for for us to keep track of sample rate and number of channels, etc. |
| 799 | |
| 800 | // The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate) |
| 801 | // as well as some other things, so we collect this info and send it along. |
| 802 | CodecInst rtp_codec; |
| 803 | rtp_codec.pltype = payload_type; |
| 804 | strncpy(rtp_codec.plname, "audio", sizeof(rtp_codec.plname)); |
| 805 | rtp_codec.plname[sizeof(rtp_codec.plname) - 1] = 0; |
ossu | 1ffbd6c | 2017-04-06 12:05:04 -0700 | [diff] [blame] | 806 | // Seems unclear if it should be clock rate or sample rate. CodecInst |
| 807 | // supposedly carries the sample rate, but only clock rate seems sensible to |
| 808 | // send to the RTP/RTCP module. |
ossu | 76d29f9 | 2017-06-09 07:30:13 -0700 | [diff] [blame] | 809 | rtp_codec.plfreq = encoder->RtpTimestampRateHz(); |
| 810 | rtp_codec.pacsize = rtc::CheckedDivExact( |
| 811 | static_cast<int>(encoder->Max10MsFramesInAPacket() * rtp_codec.plfreq), |
| 812 | 100); |
| 813 | rtp_codec.channels = encoder->NumChannels(); |
| 814 | rtp_codec.rate = 0; |
ossu | 1ffbd6c | 2017-04-06 12:05:04 -0700 | [diff] [blame] | 815 | |
ossu | 76d29f9 | 2017-06-09 07:30:13 -0700 | [diff] [blame] | 816 | if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) { |
ossu | 1ffbd6c | 2017-04-06 12:05:04 -0700 | [diff] [blame] | 817 | _rtpRtcpModule->DeRegisterSendPayload(payload_type); |
ossu | 76d29f9 | 2017-06-09 07:30:13 -0700 | [diff] [blame] | 818 | if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 819 | RTC_LOG(LS_ERROR) |
Sam Zackrisson | ecc51e9 | 2017-10-02 14:32:33 +0200 | [diff] [blame] | 820 | << "SetEncoder() failed to register codec to RTP/RTCP module"; |
ossu | 1ffbd6c | 2017-04-06 12:05:04 -0700 | [diff] [blame] | 821 | return false; |
| 822 | } |
| 823 | } |
| 824 | |
| 825 | audio_coding_->SetEncoder(std::move(encoder)); |
| 826 | return true; |
| 827 | } |
| 828 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 829 | void Channel::ModifyEncoder( |
| 830 | rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) { |
| 831 | audio_coding_->ModifyEncoder(modifier); |
| 832 | } |
| 833 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 834 | int32_t Channel::GetRecCodec(CodecInst& codec) { |
| 835 | return (audio_coding_->ReceiveCodec(&codec)); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 836 | } |
| 837 | |
minyue | 78b4d56 | 2016-11-30 04:47:39 -0800 | [diff] [blame] | 838 | void Channel::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) { |
minyue | 7e30432 | 2016-10-12 05:00:55 -0700 | [diff] [blame] | 839 | audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
michaelt | 2fedf9c | 2016-11-28 02:34:18 -0800 | [diff] [blame] | 840 | if (*encoder) { |
Oskar Sundbom | 606c882 | 2017-11-16 10:57:11 +0100 | [diff] [blame] | 841 | (*encoder)->OnReceivedUplinkBandwidth(bitrate_bps, probing_interval_ms); |
michaelt | 2fedf9c | 2016-11-28 02:34:18 -0800 | [diff] [blame] | 842 | } |
| 843 | }); |
michaelt | 566d820 | 2017-01-12 10:17:38 -0800 | [diff] [blame] | 844 | retransmission_rate_limiter_->SetMaxRate(bitrate_bps); |
Ivo Creusen | adf89b7 | 2015-04-29 16:03:33 +0200 | [diff] [blame] | 845 | } |
| 846 | |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 847 | void Channel::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) { |
| 848 | if (!use_twcc_plr_for_ana_) |
| 849 | return; |
minyue | 7e30432 | 2016-10-12 05:00:55 -0700 | [diff] [blame] | 850 | audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 851 | if (*encoder) { |
| 852 | (*encoder)->OnReceivedUplinkPacketLossFraction(packet_loss_rate); |
| 853 | } |
| 854 | }); |
| 855 | } |
| 856 | |
elad.alon | dadb4dc | 2017-03-23 15:29:50 -0700 | [diff] [blame] | 857 | void Channel::OnRecoverableUplinkPacketLossRate( |
| 858 | float recoverable_packet_loss_rate) { |
| 859 | audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| 860 | if (*encoder) { |
| 861 | (*encoder)->OnReceivedUplinkRecoverablePacketLossFraction( |
| 862 | recoverable_packet_loss_rate); |
| 863 | } |
| 864 | }); |
| 865 | } |
| 866 | |
elad.alon | d12a8e1 | 2017-03-23 11:04:48 -0700 | [diff] [blame] | 867 | void Channel::OnUplinkPacketLossRate(float packet_loss_rate) { |
| 868 | if (use_twcc_plr_for_ana_) |
| 869 | return; |
| 870 | audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| 871 | if (*encoder) { |
| 872 | (*encoder)->OnReceivedUplinkPacketLossFraction(packet_loss_rate); |
| 873 | } |
minyue | 7e30432 | 2016-10-12 05:00:55 -0700 | [diff] [blame] | 874 | }); |
minyue@webrtc.org | c1a40a7 | 2014-05-28 09:52:06 +0000 | [diff] [blame] | 875 | } |
| 876 | |
kwiberg | 1c07c70 | 2017-03-27 07:15:49 -0700 | [diff] [blame] | 877 | void Channel::SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) { |
| 878 | rtp_payload_registry_->SetAudioReceivePayloads(codecs); |
| 879 | audio_coding_->SetReceiveCodecs(codecs); |
| 880 | } |
| 881 | |
minyue | 7e30432 | 2016-10-12 05:00:55 -0700 | [diff] [blame] | 882 | bool Channel::EnableAudioNetworkAdaptor(const std::string& config_string) { |
| 883 | bool success = false; |
| 884 | audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| 885 | if (*encoder) { |
michaelt | 92aef17 | 2017-04-18 00:11:48 -0700 | [diff] [blame] | 886 | success = (*encoder)->EnableAudioNetworkAdaptor(config_string, |
| 887 | event_log_proxy_.get()); |
minyue | 7e30432 | 2016-10-12 05:00:55 -0700 | [diff] [blame] | 888 | } |
| 889 | }); |
| 890 | return success; |
| 891 | } |
| 892 | |
| 893 | void Channel::DisableAudioNetworkAdaptor() { |
| 894 | audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| 895 | if (*encoder) |
| 896 | (*encoder)->DisableAudioNetworkAdaptor(); |
| 897 | }); |
| 898 | } |
| 899 | |
| 900 | void Channel::SetReceiverFrameLengthRange(int min_frame_length_ms, |
| 901 | int max_frame_length_ms) { |
| 902 | audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| 903 | if (*encoder) { |
| 904 | (*encoder)->SetReceiverFrameLengthRange(min_frame_length_ms, |
| 905 | max_frame_length_ms); |
| 906 | } |
| 907 | }); |
| 908 | } |
| 909 | |
solenberg | 1c239d4 | 2017-09-29 06:00:28 -0700 | [diff] [blame] | 910 | void Channel::RegisterTransport(Transport* transport) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 911 | rtc::CritScope cs(&_callbackCritSect); |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 912 | _transportPtr = transport; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 913 | } |
| 914 | |
nisse | 657bab2 | 2017-02-21 06:28:10 -0800 | [diff] [blame] | 915 | void Channel::OnRtpPacket(const RtpPacketReceived& packet) { |
nisse | 657bab2 | 2017-02-21 06:28:10 -0800 | [diff] [blame] | 916 | RTPHeader header; |
| 917 | packet.GetHeader(&header); |
solenberg | 946d886 | 2017-09-21 04:02:53 -0700 | [diff] [blame] | 918 | |
| 919 | // Store playout timestamp for the received RTP packet |
| 920 | UpdatePlayoutTimestamp(false); |
| 921 | |
| 922 | header.payload_type_frequency = |
| 923 | rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType); |
| 924 | if (header.payload_type_frequency >= 0) { |
| 925 | bool in_order = IsPacketInOrder(header); |
| 926 | rtp_receive_statistics_->IncomingPacket( |
| 927 | header, packet.size(), IsPacketRetransmitted(header, in_order)); |
| 928 | rtp_payload_registry_->SetIncomingPayloadType(header); |
| 929 | |
Niels Möller | 22ec952 | 2017-10-05 08:39:15 +0200 | [diff] [blame] | 930 | ReceivePacket(packet.data(), packet.size(), header); |
solenberg | 946d886 | 2017-09-21 04:02:53 -0700 | [diff] [blame] | 931 | } |
stefan@webrtc.org | 7bb8f02 | 2013-09-06 13:40:11 +0000 | [diff] [blame] | 932 | } |
| 933 | |
| 934 | bool Channel::ReceivePacket(const uint8_t* packet, |
pkasting@chromium.org | 4591fbd | 2014-11-20 22:28:14 +0000 | [diff] [blame] | 935 | size_t packet_length, |
Niels Möller | 22ec952 | 2017-10-05 08:39:15 +0200 | [diff] [blame] | 936 | const RTPHeader& header) { |
stefan@webrtc.org | 7bb8f02 | 2013-09-06 13:40:11 +0000 | [diff] [blame] | 937 | const uint8_t* payload = packet + header.headerLength; |
pkasting@chromium.org | 4591fbd | 2014-11-20 22:28:14 +0000 | [diff] [blame] | 938 | assert(packet_length >= header.headerLength); |
| 939 | size_t payload_length = packet_length - header.headerLength; |
Karl Wiberg | 73b60b8 | 2017-09-21 15:00:58 +0200 | [diff] [blame] | 940 | const auto pl = |
| 941 | rtp_payload_registry_->PayloadTypeToPayload(header.payloadType); |
| 942 | if (!pl) { |
stefan@webrtc.org | 7bb8f02 | 2013-09-06 13:40:11 +0000 | [diff] [blame] | 943 | return false; |
wu@webrtc.org | 822fbd8 | 2013-08-15 23:38:54 +0000 | [diff] [blame] | 944 | } |
stefan@webrtc.org | 7bb8f02 | 2013-09-06 13:40:11 +0000 | [diff] [blame] | 945 | return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length, |
Niels Möller | 22ec952 | 2017-10-05 08:39:15 +0200 | [diff] [blame] | 946 | pl->typeSpecific); |
stefan@webrtc.org | 7bb8f02 | 2013-09-06 13:40:11 +0000 | [diff] [blame] | 947 | } |
| 948 | |
stefan@webrtc.org | 7bb8f02 | 2013-09-06 13:40:11 +0000 | [diff] [blame] | 949 | bool Channel::IsPacketInOrder(const RTPHeader& header) const { |
| 950 | StreamStatistician* statistician = |
| 951 | rtp_receive_statistics_->GetStatistician(header.ssrc); |
| 952 | if (!statistician) |
| 953 | return false; |
| 954 | return statistician->IsPacketInOrder(header.sequenceNumber); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 955 | } |
| 956 | |
stefan@webrtc.org | 48df381 | 2013-11-08 15:18:52 +0000 | [diff] [blame] | 957 | bool Channel::IsPacketRetransmitted(const RTPHeader& header, |
| 958 | bool in_order) const { |
stefan@webrtc.org | 7bb8f02 | 2013-09-06 13:40:11 +0000 | [diff] [blame] | 959 | StreamStatistician* statistician = |
| 960 | rtp_receive_statistics_->GetStatistician(header.ssrc); |
| 961 | if (!statistician) |
| 962 | return false; |
| 963 | // Check if this is a retransmission. |
pkasting@chromium.org | 16825b1 | 2015-01-12 21:51:21 +0000 | [diff] [blame] | 964 | int64_t min_rtt = 0; |
stefan@webrtc.org | 7bb8f02 | 2013-09-06 13:40:11 +0000 | [diff] [blame] | 965 | _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 966 | return !in_order && statistician->IsRetransmitOfOldPacket(header, min_rtt); |
wu@webrtc.org | 822fbd8 | 2013-08-15 23:38:54 +0000 | [diff] [blame] | 967 | } |
| 968 | |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 969 | int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) { |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 970 | // Store playout timestamp for the received RTCP packet |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame] | 971 | UpdatePlayoutTimestamp(true); |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 972 | |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 973 | // Deliver RTCP packet to RTP/RTCP module for parsing |
nisse | 479d3d7 | 2017-09-13 07:53:37 -0700 | [diff] [blame] | 974 | _rtpRtcpModule->IncomingRtcpPacket(data, length); |
wu@webrtc.org | 82c4b85 | 2014-05-20 22:55:01 +0000 | [diff] [blame] | 975 | |
Minyue | 2013aec | 2015-05-13 14:14:42 +0200 | [diff] [blame] | 976 | int64_t rtt = GetRTT(true); |
| 977 | if (rtt == 0) { |
| 978 | // Waiting for valid RTT. |
| 979 | return 0; |
| 980 | } |
Erik Språng | 737336d | 2016-07-29 12:59:36 +0200 | [diff] [blame] | 981 | |
| 982 | int64_t nack_window_ms = rtt; |
| 983 | if (nack_window_ms < kMinRetransmissionWindowMs) { |
| 984 | nack_window_ms = kMinRetransmissionWindowMs; |
| 985 | } else if (nack_window_ms > kMaxRetransmissionWindowMs) { |
| 986 | nack_window_ms = kMaxRetransmissionWindowMs; |
| 987 | } |
| 988 | retransmission_rate_limiter_->SetWindowSize(nack_window_ms); |
| 989 | |
minyue | 7e30432 | 2016-10-12 05:00:55 -0700 | [diff] [blame] | 990 | // Invoke audio encoders OnReceivedRtt(). |
| 991 | audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| 992 | if (*encoder) |
| 993 | (*encoder)->OnReceivedRtt(rtt); |
| 994 | }); |
| 995 | |
Minyue | 2013aec | 2015-05-13 14:14:42 +0200 | [diff] [blame] | 996 | uint32_t ntp_secs = 0; |
| 997 | uint32_t ntp_frac = 0; |
| 998 | uint32_t rtp_timestamp = 0; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 999 | if (0 != |
| 1000 | _rtpRtcpModule->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL, |
| 1001 | &rtp_timestamp)) { |
Minyue | 2013aec | 2015-05-13 14:14:42 +0200 | [diff] [blame] | 1002 | // Waiting for RTCP. |
| 1003 | return 0; |
| 1004 | } |
| 1005 | |
stefan@webrtc.org | 8e24d87 | 2014-09-02 18:58:24 +0000 | [diff] [blame] | 1006 | { |
tommi | 31fc21f | 2016-01-21 10:37:37 -0800 | [diff] [blame] | 1007 | rtc::CritScope lock(&ts_stats_lock_); |
minyue@webrtc.org | 2c0cdbc | 2014-10-09 10:52:43 +0000 | [diff] [blame] | 1008 | ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); |
stefan@webrtc.org | 8e24d87 | 2014-09-02 18:58:24 +0000 | [diff] [blame] | 1009 | } |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 1010 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1011 | } |
| 1012 | |
solenberg | 8d73f8c | 2017-03-08 01:52:20 -0800 | [diff] [blame] | 1013 | int Channel::GetSpeechOutputLevel() const { |
| 1014 | return _outputAudioLevel.Level(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1015 | } |
| 1016 | |
solenberg | 8d73f8c | 2017-03-08 01:52:20 -0800 | [diff] [blame] | 1017 | int Channel::GetSpeechOutputLevelFullRange() const { |
| 1018 | return _outputAudioLevel.LevelFullRange(); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1019 | } |
| 1020 | |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 1021 | double Channel::GetTotalOutputEnergy() const { |
zstein | 3c45186 | 2017-07-20 09:57:42 -0700 | [diff] [blame] | 1022 | return _outputAudioLevel.TotalEnergy(); |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 1023 | } |
| 1024 | |
| 1025 | double Channel::GetTotalOutputDuration() const { |
zstein | 3c45186 | 2017-07-20 09:57:42 -0700 | [diff] [blame] | 1026 | return _outputAudioLevel.TotalDuration(); |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 1027 | } |
| 1028 | |
solenberg | 8d73f8c | 2017-03-08 01:52:20 -0800 | [diff] [blame] | 1029 | void Channel::SetInputMute(bool enable) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1030 | rtc::CritScope cs(&volume_settings_critsect_); |
solenberg | 1c2af8e | 2016-03-24 10:36:00 -0700 | [diff] [blame] | 1031 | input_mute_ = enable; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1032 | } |
| 1033 | |
solenberg | 1c2af8e | 2016-03-24 10:36:00 -0700 | [diff] [blame] | 1034 | bool Channel::InputMute() const { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1035 | rtc::CritScope cs(&volume_settings_critsect_); |
solenberg | 1c2af8e | 2016-03-24 10:36:00 -0700 | [diff] [blame] | 1036 | return input_mute_; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1037 | } |
| 1038 | |
solenberg | 8d73f8c | 2017-03-08 01:52:20 -0800 | [diff] [blame] | 1039 | void Channel::SetChannelOutputVolumeScaling(float scaling) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1040 | rtc::CritScope cs(&volume_settings_critsect_); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1041 | _outputGain = scaling; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1042 | } |
| 1043 | |
solenberg | 8842c3e | 2016-03-11 03:06:41 -0800 | [diff] [blame] | 1044 | int Channel::SendTelephoneEventOutband(int event, int duration_ms) { |
solenberg | 8842c3e | 2016-03-11 03:06:41 -0800 | [diff] [blame] | 1045 | RTC_DCHECK_LE(0, event); |
| 1046 | RTC_DCHECK_GE(255, event); |
| 1047 | RTC_DCHECK_LE(0, duration_ms); |
| 1048 | RTC_DCHECK_GE(65535, duration_ms); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1049 | if (!Sending()) { |
| 1050 | return -1; |
| 1051 | } |
solenberg | 8842c3e | 2016-03-11 03:06:41 -0800 | [diff] [blame] | 1052 | if (_rtpRtcpModule->SendTelephoneEventOutband( |
| 1053 | event, duration_ms, kTelephoneEventAttenuationdB) != 0) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1054 | RTC_LOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1055 | return -1; |
| 1056 | } |
| 1057 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1058 | } |
| 1059 | |
solenberg | ffbbcac | 2016-11-17 05:25:37 -0800 | [diff] [blame] | 1060 | int Channel::SetSendTelephoneEventPayloadType(int payload_type, |
| 1061 | int payload_frequency) { |
solenberg | 31642aa | 2016-03-14 08:00:37 -0700 | [diff] [blame] | 1062 | RTC_DCHECK_LE(0, payload_type); |
| 1063 | RTC_DCHECK_GE(127, payload_type); |
| 1064 | CodecInst codec = {0}; |
solenberg | 31642aa | 2016-03-14 08:00:37 -0700 | [diff] [blame] | 1065 | codec.pltype = payload_type; |
solenberg | ffbbcac | 2016-11-17 05:25:37 -0800 | [diff] [blame] | 1066 | codec.plfreq = payload_frequency; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1067 | memcpy(codec.plname, "telephone-event", 16); |
| 1068 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) { |
| 1069 | _rtpRtcpModule->DeRegisterSendPayload(codec.pltype); |
| 1070 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1071 | RTC_LOG(LS_ERROR) |
| 1072 | << "SetSendTelephoneEventPayloadType() failed to register " |
| 1073 | "send payload type"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1074 | return -1; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1075 | } |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1076 | } |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1077 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1078 | } |
| 1079 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1080 | int Channel::SetLocalSSRC(unsigned int ssrc) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1081 | if (channel_state_.Get().sending) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1082 | RTC_LOG(LS_ERROR) << "SetLocalSSRC() already sending"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1083 | return -1; |
| 1084 | } |
| 1085 | _rtpRtcpModule->SetSSRC(ssrc); |
| 1086 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1087 | } |
| 1088 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1089 | int Channel::GetRemoteSSRC(unsigned int& ssrc) { |
| 1090 | ssrc = rtp_receiver_->SSRC(); |
| 1091 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1092 | } |
| 1093 | |
wu@webrtc.org | ebdb0e3 | 2014-03-06 23:49:08 +0000 | [diff] [blame] | 1094 | int Channel::SetSendAudioLevelIndicationStatus(bool enable, unsigned char id) { |
andrew@webrtc.org | f3930e9 | 2013-09-18 22:37:32 +0000 | [diff] [blame] | 1095 | _includeAudioLevelIndication = enable; |
wu@webrtc.org | ebdb0e3 | 2014-03-06 23:49:08 +0000 | [diff] [blame] | 1096 | return SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1097 | } |
andrew@webrtc.org | f3930e9 | 2013-09-18 22:37:32 +0000 | [diff] [blame] | 1098 | |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 1099 | void Channel::EnableSendTransportSequenceNumber(int id) { |
| 1100 | int ret = |
| 1101 | SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id); |
| 1102 | RTC_DCHECK_EQ(0, ret); |
| 1103 | } |
| 1104 | |
stefan | bba9dec | 2016-02-01 04:39:55 -0800 | [diff] [blame] | 1105 | void Channel::RegisterSenderCongestionControlObjects( |
nisse | b8f9a32 | 2017-03-27 05:36:15 -0700 | [diff] [blame] | 1106 | RtpTransportControllerSendInterface* transport, |
stefan | 7de8d64 | 2017-02-07 07:14:08 -0800 | [diff] [blame] | 1107 | RtcpBandwidthObserver* bandwidth_observer) { |
nisse | b8f9a32 | 2017-03-27 05:36:15 -0700 | [diff] [blame] | 1108 | RtpPacketSender* rtp_packet_sender = transport->packet_sender(); |
| 1109 | TransportFeedbackObserver* transport_feedback_observer = |
| 1110 | transport->transport_feedback_observer(); |
| 1111 | PacketRouter* packet_router = transport->packet_router(); |
| 1112 | |
stefan | bba9dec | 2016-02-01 04:39:55 -0800 | [diff] [blame] | 1113 | RTC_DCHECK(rtp_packet_sender); |
| 1114 | RTC_DCHECK(transport_feedback_observer); |
kwiberg | ee89e78 | 2017-08-09 17:22:01 -0700 | [diff] [blame] | 1115 | RTC_DCHECK(packet_router); |
| 1116 | RTC_DCHECK(!packet_router_); |
stefan | 7de8d64 | 2017-02-07 07:14:08 -0800 | [diff] [blame] | 1117 | rtcp_observer_->SetBandwidthObserver(bandwidth_observer); |
stefan | bba9dec | 2016-02-01 04:39:55 -0800 | [diff] [blame] | 1118 | feedback_observer_proxy_->SetTransportFeedbackObserver( |
| 1119 | transport_feedback_observer); |
| 1120 | seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router); |
| 1121 | rtp_packet_sender_proxy_->SetPacketSender(rtp_packet_sender); |
| 1122 | _rtpRtcpModule->SetStorePacketsStatus(true, 600); |
eladalon | 822ff2b | 2017-08-01 06:30:28 -0700 | [diff] [blame] | 1123 | constexpr bool remb_candidate = false; |
| 1124 | packet_router->AddSendRtpModule(_rtpRtcpModule.get(), remb_candidate); |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 1125 | packet_router_ = packet_router; |
| 1126 | } |
| 1127 | |
stefan | bba9dec | 2016-02-01 04:39:55 -0800 | [diff] [blame] | 1128 | void Channel::RegisterReceiverCongestionControlObjects( |
| 1129 | PacketRouter* packet_router) { |
kwiberg | ee89e78 | 2017-08-09 17:22:01 -0700 | [diff] [blame] | 1130 | RTC_DCHECK(packet_router); |
| 1131 | RTC_DCHECK(!packet_router_); |
eladalon | 822ff2b | 2017-08-01 06:30:28 -0700 | [diff] [blame] | 1132 | constexpr bool remb_candidate = false; |
| 1133 | packet_router->AddReceiveRtpModule(_rtpRtcpModule.get(), remb_candidate); |
stefan | bba9dec | 2016-02-01 04:39:55 -0800 | [diff] [blame] | 1134 | packet_router_ = packet_router; |
| 1135 | } |
| 1136 | |
nisse | fdbfdc9 | 2017-03-31 05:44:52 -0700 | [diff] [blame] | 1137 | void Channel::ResetSenderCongestionControlObjects() { |
stefan | bba9dec | 2016-02-01 04:39:55 -0800 | [diff] [blame] | 1138 | RTC_DCHECK(packet_router_); |
| 1139 | _rtpRtcpModule->SetStorePacketsStatus(false, 600); |
stefan | 7de8d64 | 2017-02-07 07:14:08 -0800 | [diff] [blame] | 1140 | rtcp_observer_->SetBandwidthObserver(nullptr); |
stefan | bba9dec | 2016-02-01 04:39:55 -0800 | [diff] [blame] | 1141 | feedback_observer_proxy_->SetTransportFeedbackObserver(nullptr); |
| 1142 | seq_num_allocator_proxy_->SetSequenceNumberAllocator(nullptr); |
nisse | fdbfdc9 | 2017-03-31 05:44:52 -0700 | [diff] [blame] | 1143 | packet_router_->RemoveSendRtpModule(_rtpRtcpModule.get()); |
stefan | bba9dec | 2016-02-01 04:39:55 -0800 | [diff] [blame] | 1144 | packet_router_ = nullptr; |
| 1145 | rtp_packet_sender_proxy_->SetPacketSender(nullptr); |
| 1146 | } |
| 1147 | |
nisse | fdbfdc9 | 2017-03-31 05:44:52 -0700 | [diff] [blame] | 1148 | void Channel::ResetReceiverCongestionControlObjects() { |
| 1149 | RTC_DCHECK(packet_router_); |
| 1150 | packet_router_->RemoveReceiveRtpModule(_rtpRtcpModule.get()); |
| 1151 | packet_router_ = nullptr; |
| 1152 | } |
| 1153 | |
pbos@webrtc.org | d16e839 | 2014-12-19 13:49:55 +0000 | [diff] [blame] | 1154 | void Channel::SetRTCPStatus(bool enable) { |
pbos | da903ea | 2015-10-02 02:36:56 -0700 | [diff] [blame] | 1155 | _rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1156 | } |
| 1157 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1158 | int Channel::SetRTCP_CNAME(const char cName[256]) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1159 | if (_rtpRtcpModule->SetCNAME(cName) != 0) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1160 | RTC_LOG(LS_ERROR) << "SetRTCP_CNAME() failed to set RTCP CNAME"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1161 | return -1; |
| 1162 | } |
| 1163 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1164 | } |
| 1165 | |
henrika@webrtc.org | 8a2fc88 | 2012-08-22 08:53:55 +0000 | [diff] [blame] | 1166 | int Channel::GetRemoteRTCPReportBlocks( |
| 1167 | std::vector<ReportBlock>* report_blocks) { |
| 1168 | if (report_blocks == NULL) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1169 | RTC_LOG(LS_ERROR) << "GetRemoteRTCPReportBlock()s invalid report_blocks."; |
henrika@webrtc.org | 8a2fc88 | 2012-08-22 08:53:55 +0000 | [diff] [blame] | 1170 | return -1; |
| 1171 | } |
| 1172 | |
| 1173 | // Get the report blocks from the latest received RTCP Sender or Receiver |
| 1174 | // Report. Each element in the vector contains the sender's SSRC and a |
| 1175 | // report block according to RFC 3550. |
| 1176 | std::vector<RTCPReportBlock> rtcp_report_blocks; |
| 1177 | if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks) != 0) { |
henrika@webrtc.org | 8a2fc88 | 2012-08-22 08:53:55 +0000 | [diff] [blame] | 1178 | return -1; |
| 1179 | } |
| 1180 | |
| 1181 | if (rtcp_report_blocks.empty()) |
| 1182 | return 0; |
| 1183 | |
| 1184 | std::vector<RTCPReportBlock>::const_iterator it = rtcp_report_blocks.begin(); |
| 1185 | for (; it != rtcp_report_blocks.end(); ++it) { |
| 1186 | ReportBlock report_block; |
srte | 3e69e5c | 2017-08-09 06:13:45 -0700 | [diff] [blame] | 1187 | report_block.sender_SSRC = it->sender_ssrc; |
| 1188 | report_block.source_SSRC = it->source_ssrc; |
| 1189 | report_block.fraction_lost = it->fraction_lost; |
| 1190 | report_block.cumulative_num_packets_lost = it->packets_lost; |
| 1191 | report_block.extended_highest_sequence_number = |
| 1192 | it->extended_highest_sequence_number; |
henrika@webrtc.org | 8a2fc88 | 2012-08-22 08:53:55 +0000 | [diff] [blame] | 1193 | report_block.interarrival_jitter = it->jitter; |
srte | 3e69e5c | 2017-08-09 06:13:45 -0700 | [diff] [blame] | 1194 | report_block.last_SR_timestamp = it->last_sender_report_timestamp; |
| 1195 | report_block.delay_since_last_SR = it->delay_since_last_sender_report; |
henrika@webrtc.org | 8a2fc88 | 2012-08-22 08:53:55 +0000 | [diff] [blame] | 1196 | report_blocks->push_back(report_block); |
| 1197 | } |
| 1198 | return 0; |
| 1199 | } |
| 1200 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1201 | int Channel::GetRTPStatistics(CallStatistics& stats) { |
| 1202 | // --- RtcpStatistics |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1203 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1204 | // The jitter statistics is updated for each received RTP packet and is |
| 1205 | // based on received packets. |
| 1206 | RtcpStatistics statistics; |
| 1207 | StreamStatistician* statistician = |
| 1208 | rtp_receive_statistics_->GetStatistician(rtp_receiver_->SSRC()); |
Peter Boström | 59013bc | 2016-02-12 11:35:08 +0100 | [diff] [blame] | 1209 | if (statistician) { |
| 1210 | statistician->GetStatistics(&statistics, |
| 1211 | _rtpRtcpModule->RTCP() == RtcpMode::kOff); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1212 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1213 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1214 | stats.fractionLost = statistics.fraction_lost; |
srte | 186d9c3 | 2017-08-04 05:03:53 -0700 | [diff] [blame] | 1215 | stats.cumulativeLost = statistics.packets_lost; |
| 1216 | stats.extendedMax = statistics.extended_highest_sequence_number; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1217 | stats.jitterSamples = statistics.jitter; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1218 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1219 | // --- RTT |
| 1220 | stats.rttMs = GetRTT(true); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1221 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1222 | // --- Data counters |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1223 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1224 | size_t bytesSent(0); |
| 1225 | uint32_t packetsSent(0); |
| 1226 | size_t bytesReceived(0); |
| 1227 | uint32_t packetsReceived(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1228 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1229 | if (statistician) { |
| 1230 | statistician->GetDataCounters(&bytesReceived, &packetsReceived); |
| 1231 | } |
wu@webrtc.org | 822fbd8 | 2013-08-15 23:38:54 +0000 | [diff] [blame] | 1232 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1233 | if (_rtpRtcpModule->DataCountersRTP(&bytesSent, &packetsSent) != 0) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1234 | RTC_LOG(LS_WARNING) |
| 1235 | << "GetRTPStatistics() failed to retrieve RTP datacounters" |
| 1236 | << " => output will not be complete"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1237 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1238 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1239 | stats.bytesSent = bytesSent; |
| 1240 | stats.packetsSent = packetsSent; |
| 1241 | stats.bytesReceived = bytesReceived; |
| 1242 | stats.packetsReceived = packetsReceived; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1243 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1244 | // --- Timestamps |
| 1245 | { |
| 1246 | rtc::CritScope lock(&ts_stats_lock_); |
| 1247 | stats.capture_start_ntp_time_ms_ = capture_start_ntp_time_ms_; |
| 1248 | } |
| 1249 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1250 | } |
| 1251 | |
pwestin@webrtc.org | db24995 | 2013-06-05 15:33:20 +0000 | [diff] [blame] | 1252 | void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) { |
| 1253 | // None of these functions can fail. |
Stefan Holmer | b86d4e4 | 2015-12-07 10:26:18 +0100 | [diff] [blame] | 1254 | // If pacing is enabled we always store packets. |
| 1255 | if (!pacing_enabled_) |
| 1256 | _rtpRtcpModule->SetStorePacketsStatus(enable, maxNumberOfPackets); |
stefan@webrtc.org | 7bb8f02 | 2013-09-06 13:40:11 +0000 | [diff] [blame] | 1257 | rtp_receive_statistics_->SetMaxReorderingThreshold(maxNumberOfPackets); |
pwestin@webrtc.org | d30859e | 2013-06-06 21:09:01 +0000 | [diff] [blame] | 1258 | if (enable) |
andrew@webrtc.org | eb524d9 | 2013-09-23 23:02:24 +0000 | [diff] [blame] | 1259 | audio_coding_->EnableNack(maxNumberOfPackets); |
pwestin@webrtc.org | d30859e | 2013-06-06 21:09:01 +0000 | [diff] [blame] | 1260 | else |
andrew@webrtc.org | eb524d9 | 2013-09-23 23:02:24 +0000 | [diff] [blame] | 1261 | audio_coding_->DisableNack(); |
pwestin@webrtc.org | db24995 | 2013-06-05 15:33:20 +0000 | [diff] [blame] | 1262 | } |
| 1263 | |
pwestin@webrtc.org | d30859e | 2013-06-06 21:09:01 +0000 | [diff] [blame] | 1264 | // Called when we are missing one or more packets. |
| 1265 | int Channel::ResendPackets(const uint16_t* sequence_numbers, int length) { |
pwestin@webrtc.org | db24995 | 2013-06-05 15:33:20 +0000 | [diff] [blame] | 1266 | return _rtpRtcpModule->SendNACK(sequence_numbers, length); |
| 1267 | } |
| 1268 | |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 1269 | void Channel::ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) { |
henrika | 4515fa0 | 2017-05-03 08:30:15 -0700 | [diff] [blame] | 1270 | // Avoid posting any new tasks if sending was already stopped in StopSend(). |
| 1271 | rtc::CritScope cs(&encoder_queue_lock_); |
| 1272 | if (!encoder_queue_is_active_) { |
| 1273 | return; |
| 1274 | } |
henrika | 4580217 | 2017-09-28 09:39:34 +0200 | [diff] [blame] | 1275 | // Profile time between when the audio frame is added to the task queue and |
| 1276 | // when the task is actually executed. |
| 1277 | audio_frame->UpdateProfileTimeStamp(); |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 1278 | encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>( |
| 1279 | new ProcessAndEncodeAudioTask(std::move(audio_frame), this))); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1280 | } |
| 1281 | |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 1282 | void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) { |
| 1283 | RTC_DCHECK_RUN_ON(encoder_queue_); |
| 1284 | RTC_DCHECK_GT(audio_input->samples_per_channel_, 0); |
| 1285 | RTC_DCHECK_LE(audio_input->num_channels_, 2); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1286 | |
henrika | 4580217 | 2017-09-28 09:39:34 +0200 | [diff] [blame] | 1287 | // Measure time between when the audio frame is added to the task queue and |
| 1288 | // when the task is actually executed. Goal is to keep track of unwanted |
| 1289 | // extra latency added by the task queue. |
| 1290 | RTC_HISTOGRAM_COUNTS_10000("WebRTC.Audio.EncodingTaskQueueLatencyMs", |
| 1291 | audio_input->ElapsedProfileTimeMs()); |
| 1292 | |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 1293 | bool is_muted = InputMute(); |
| 1294 | AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1295 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1296 | if (_includeAudioLevelIndication) { |
| 1297 | size_t length = |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 1298 | audio_input->samples_per_channel_ * audio_input->num_channels_; |
yujo | 36b1a5f | 2017-06-12 12:45:32 -0700 | [diff] [blame] | 1299 | RTC_CHECK_LE(length, AudioFrame::kMaxDataSizeBytes); |
solenberg | 1c2af8e | 2016-03-24 10:36:00 -0700 | [diff] [blame] | 1300 | if (is_muted && previous_frame_muted_) { |
henrik.lundin | 5049942 | 2016-11-29 04:26:24 -0800 | [diff] [blame] | 1301 | rms_level_.AnalyzeMuted(length); |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1302 | } else { |
henrik.lundin | 5049942 | 2016-11-29 04:26:24 -0800 | [diff] [blame] | 1303 | rms_level_.Analyze( |
yujo | 36b1a5f | 2017-06-12 12:45:32 -0700 | [diff] [blame] | 1304 | rtc::ArrayView<const int16_t>(audio_input->data(), length)); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1305 | } |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1306 | } |
solenberg | 1c2af8e | 2016-03-24 10:36:00 -0700 | [diff] [blame] | 1307 | previous_frame_muted_ = is_muted; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1308 | |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 1309 | // Add 10ms of raw (PCM) audio data to the encoder @ 32kHz. |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1310 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1311 | // The ACM resamples internally. |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 1312 | audio_input->timestamp_ = _timeStamp; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1313 | // This call will trigger AudioPacketizationCallback::SendData if encoding |
| 1314 | // is done and payload is ready for packetization and transmission. |
| 1315 | // Otherwise, it will return without invoking the callback. |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 1316 | if (audio_coding_->Add10MsData(*audio_input) < 0) { |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1317 | RTC_LOG(LS_ERROR) << "ACM::Add10MsData() failed."; |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 1318 | return; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1319 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1320 | |
henrika | ec6fbd2 | 2017-03-31 05:43:36 -0700 | [diff] [blame] | 1321 | _timeStamp += static_cast<uint32_t>(audio_input->samples_per_channel_); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1322 | } |
| 1323 | |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1324 | void Channel::SetAssociatedSendChannel(Channel* channel) { |
| 1325 | RTC_DCHECK_NE(this, channel); |
solenberg | 7602aab | 2016-11-14 11:30:07 -0800 | [diff] [blame] | 1326 | rtc::CritScope lock(&assoc_send_channel_lock_); |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1327 | associated_send_channel_ = channel; |
Minyue | 2013aec | 2015-05-13 14:14:42 +0200 | [diff] [blame] | 1328 | } |
| 1329 | |
ivoc | 14d5dbe | 2016-07-04 07:06:55 -0700 | [diff] [blame] | 1330 | void Channel::SetRtcEventLog(RtcEventLog* event_log) { |
| 1331 | event_log_proxy_->SetEventLog(event_log); |
| 1332 | } |
| 1333 | |
michaelt | 9332b7d | 2016-11-30 07:51:13 -0800 | [diff] [blame] | 1334 | void Channel::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) { |
| 1335 | rtcp_rtt_stats_proxy_->SetRtcpRttStats(rtcp_rtt_stats); |
| 1336 | } |
| 1337 | |
nisse | 284542b | 2017-01-10 08:58:32 -0800 | [diff] [blame] | 1338 | void Channel::UpdateOverheadForEncoder() { |
hbos | 3fd31fe | 2017-02-28 05:43:16 -0800 | [diff] [blame] | 1339 | size_t overhead_per_packet = |
| 1340 | transport_overhead_per_packet_ + rtp_overhead_per_packet_; |
nisse | 284542b | 2017-01-10 08:58:32 -0800 | [diff] [blame] | 1341 | audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| 1342 | if (*encoder) { |
hbos | 3fd31fe | 2017-02-28 05:43:16 -0800 | [diff] [blame] | 1343 | (*encoder)->OnReceivedOverhead(overhead_per_packet); |
nisse | 284542b | 2017-01-10 08:58:32 -0800 | [diff] [blame] | 1344 | } |
| 1345 | }); |
| 1346 | } |
| 1347 | |
| 1348 | void Channel::SetTransportOverhead(size_t transport_overhead_per_packet) { |
hbos | 3fd31fe | 2017-02-28 05:43:16 -0800 | [diff] [blame] | 1349 | rtc::CritScope cs(&overhead_per_packet_lock_); |
nisse | 284542b | 2017-01-10 08:58:32 -0800 | [diff] [blame] | 1350 | transport_overhead_per_packet_ = transport_overhead_per_packet; |
| 1351 | UpdateOverheadForEncoder(); |
michaelt | 79e0588 | 2016-11-08 02:50:09 -0800 | [diff] [blame] | 1352 | } |
| 1353 | |
hbos | 3fd31fe | 2017-02-28 05:43:16 -0800 | [diff] [blame] | 1354 | // TODO(solenberg): Make AudioSendStream an OverheadObserver instead. |
michaelt | bf65be5 | 2016-12-15 06:24:49 -0800 | [diff] [blame] | 1355 | void Channel::OnOverheadChanged(size_t overhead_bytes_per_packet) { |
hbos | 3fd31fe | 2017-02-28 05:43:16 -0800 | [diff] [blame] | 1356 | rtc::CritScope cs(&overhead_per_packet_lock_); |
nisse | 284542b | 2017-01-10 08:58:32 -0800 | [diff] [blame] | 1357 | rtp_overhead_per_packet_ = overhead_bytes_per_packet; |
| 1358 | UpdateOverheadForEncoder(); |
michaelt | bf65be5 | 2016-12-15 06:24:49 -0800 | [diff] [blame] | 1359 | } |
| 1360 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1361 | int Channel::GetNetworkStatistics(NetworkStatistics& stats) { |
| 1362 | return audio_coding_->GetNetworkStatistics(&stats); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1363 | } |
| 1364 | |
wu@webrtc.org | 24301a6 | 2013-12-13 19:17:43 +0000 | [diff] [blame] | 1365 | void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const { |
| 1366 | audio_coding_->GetDecodingCallStatistics(stats); |
| 1367 | } |
| 1368 | |
ivoc | e1198e0 | 2017-09-08 08:13:19 -0700 | [diff] [blame] | 1369 | ANAStats Channel::GetANAStatistics() const { |
| 1370 | return audio_coding_->GetANAStats(); |
| 1371 | } |
| 1372 | |
solenberg | 358057b | 2015-11-27 10:46:42 -0800 | [diff] [blame] | 1373 | uint32_t Channel::GetDelayEstimate() const { |
solenberg | 08b19df | 2017-02-15 00:42:31 -0800 | [diff] [blame] | 1374 | rtc::CritScope lock(&video_sync_lock_); |
| 1375 | return audio_coding_->FilteredCurrentDelayMs() + playout_delay_ms_; |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1376 | } |
| 1377 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1378 | int Channel::SetMinimumPlayoutDelay(int delayMs) { |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1379 | if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) || |
| 1380 | (delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1381 | RTC_LOG(LS_ERROR) << "SetMinimumPlayoutDelay() invalid min delay"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1382 | return -1; |
| 1383 | } |
| 1384 | if (audio_coding_->SetMinimumPlayoutDelay(delayMs) != 0) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1385 | RTC_LOG(LS_ERROR) |
| 1386 | << "SetMinimumPlayoutDelay() failed to set min playout delay"; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1387 | return -1; |
| 1388 | } |
| 1389 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1390 | } |
| 1391 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame] | 1392 | int Channel::GetPlayoutTimestamp(unsigned int& timestamp) { |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1393 | uint32_t playout_timestamp_rtp = 0; |
| 1394 | { |
tommi | 31fc21f | 2016-01-21 10:37:37 -0800 | [diff] [blame] | 1395 | rtc::CritScope lock(&video_sync_lock_); |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1396 | playout_timestamp_rtp = playout_timestamp_rtp_; |
| 1397 | } |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1398 | if (playout_timestamp_rtp == 0) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1399 | RTC_LOG(LS_ERROR) << "GetPlayoutTimestamp() failed to retrieve timestamp"; |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame] | 1400 | return -1; |
| 1401 | } |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1402 | timestamp = playout_timestamp_rtp; |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame] | 1403 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1404 | } |
| 1405 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1406 | int Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule, |
| 1407 | RtpReceiver** rtp_receiver) const { |
| 1408 | *rtpRtcpModule = _rtpRtcpModule.get(); |
| 1409 | *rtp_receiver = rtp_receiver_.get(); |
| 1410 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1411 | } |
| 1412 | |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1413 | void Channel::UpdatePlayoutTimestamp(bool rtcp) { |
henrik.lundin | 96bd502 | 2016-04-06 04:13:56 -0700 | [diff] [blame] | 1414 | jitter_buffer_playout_timestamp_ = audio_coding_->PlayoutTimestamp(); |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1415 | |
henrik.lundin | 96bd502 | 2016-04-06 04:13:56 -0700 | [diff] [blame] | 1416 | if (!jitter_buffer_playout_timestamp_) { |
| 1417 | // This can happen if this channel has not received any RTP packets. In |
| 1418 | // this case, NetEq is not capable of computing a playout timestamp. |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1419 | return; |
| 1420 | } |
| 1421 | |
| 1422 | uint16_t delay_ms = 0; |
| 1423 | if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1424 | RTC_LOG(LS_WARNING) << "Channel::UpdatePlayoutTimestamp() failed to read" |
| 1425 | << " playout delay from the ADM"; |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1426 | return; |
| 1427 | } |
| 1428 | |
henrik.lundin | 96bd502 | 2016-04-06 04:13:56 -0700 | [diff] [blame] | 1429 | RTC_DCHECK(jitter_buffer_playout_timestamp_); |
| 1430 | uint32_t playout_timestamp = *jitter_buffer_playout_timestamp_; |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1431 | |
| 1432 | // Remove the playout delay. |
ossu | e280cde | 2016-10-12 11:04:10 -0700 | [diff] [blame] | 1433 | playout_timestamp -= (delay_ms * (GetRtpTimestampRateHz() / 1000)); |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1434 | |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1435 | { |
tommi | 31fc21f | 2016-01-21 10:37:37 -0800 | [diff] [blame] | 1436 | rtc::CritScope lock(&video_sync_lock_); |
solenberg | 81d93f3 | 2017-02-14 03:44:57 -0800 | [diff] [blame] | 1437 | if (!rtcp) { |
henrik.lundin | 96bd502 | 2016-04-06 04:13:56 -0700 | [diff] [blame] | 1438 | playout_timestamp_rtp_ = playout_timestamp; |
deadbeef | 7437588 | 2015-08-13 12:09:10 -0700 | [diff] [blame] | 1439 | } |
| 1440 | playout_delay_ms_ = delay_ms; |
| 1441 | } |
| 1442 | } |
| 1443 | |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1444 | int Channel::SetSendRtpHeaderExtension(bool enable, |
| 1445 | RTPExtensionType type, |
wu@webrtc.org | ebdb0e3 | 2014-03-06 23:49:08 +0000 | [diff] [blame] | 1446 | unsigned char id) { |
| 1447 | int error = 0; |
| 1448 | _rtpRtcpModule->DeregisterSendRtpHeaderExtension(type); |
| 1449 | if (enable) { |
| 1450 | error = _rtpRtcpModule->RegisterSendRtpHeaderExtension(type, id); |
| 1451 | } |
| 1452 | return error; |
| 1453 | } |
minyue@webrtc.org | c1a40a7 | 2014-05-28 09:52:06 +0000 | [diff] [blame] | 1454 | |
ossu | e280cde | 2016-10-12 11:04:10 -0700 | [diff] [blame] | 1455 | int Channel::GetRtpTimestampRateHz() const { |
| 1456 | const auto format = audio_coding_->ReceiveFormat(); |
| 1457 | // Default to the playout frequency if we've not gotten any packets yet. |
| 1458 | // TODO(ossu): Zero clockrate can only happen if we've added an external |
| 1459 | // decoder for a format we don't support internally. Remove once that way of |
| 1460 | // adding decoders is gone! |
| 1461 | return (format && format->clockrate_hz != 0) |
| 1462 | ? format->clockrate_hz |
| 1463 | : audio_coding_->PlayoutFrequency(); |
wu@webrtc.org | 94454b7 | 2014-06-05 20:34:08 +0000 | [diff] [blame] | 1464 | } |
| 1465 | |
Minyue | 2013aec | 2015-05-13 14:14:42 +0200 | [diff] [blame] | 1466 | int64_t Channel::GetRTT(bool allow_associate_channel) const { |
pbos | da903ea | 2015-10-02 02:36:56 -0700 | [diff] [blame] | 1467 | RtcpMode method = _rtpRtcpModule->RTCP(); |
| 1468 | if (method == RtcpMode::kOff) { |
minyue@webrtc.org | 2b58a44 | 2014-09-11 07:51:53 +0000 | [diff] [blame] | 1469 | return 0; |
| 1470 | } |
| 1471 | std::vector<RTCPReportBlock> report_blocks; |
| 1472 | _rtpRtcpModule->RemoteRTCPStat(&report_blocks); |
Minyue | 2013aec | 2015-05-13 14:14:42 +0200 | [diff] [blame] | 1473 | |
| 1474 | int64_t rtt = 0; |
minyue@webrtc.org | 2b58a44 | 2014-09-11 07:51:53 +0000 | [diff] [blame] | 1475 | if (report_blocks.empty()) { |
Minyue | 2013aec | 2015-05-13 14:14:42 +0200 | [diff] [blame] | 1476 | if (allow_associate_channel) { |
tommi | 31fc21f | 2016-01-21 10:37:37 -0800 | [diff] [blame] | 1477 | rtc::CritScope lock(&assoc_send_channel_lock_); |
Minyue | 2013aec | 2015-05-13 14:14:42 +0200 | [diff] [blame] | 1478 | // Tries to get RTT from an associated channel. This is important for |
| 1479 | // receive-only channels. |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1480 | if (associated_send_channel_) { |
Minyue | 2013aec | 2015-05-13 14:14:42 +0200 | [diff] [blame] | 1481 | // To prevent infinite recursion and deadlock, calling GetRTT of |
| 1482 | // associate channel should always use "false" for argument: |
| 1483 | // |allow_associate_channel|. |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1484 | rtt = associated_send_channel_->GetRTT(false); |
Minyue | 2013aec | 2015-05-13 14:14:42 +0200 | [diff] [blame] | 1485 | } |
| 1486 | } |
| 1487 | return rtt; |
minyue@webrtc.org | 2b58a44 | 2014-09-11 07:51:53 +0000 | [diff] [blame] | 1488 | } |
| 1489 | |
| 1490 | uint32_t remoteSSRC = rtp_receiver_->SSRC(); |
| 1491 | std::vector<RTCPReportBlock>::const_iterator it = report_blocks.begin(); |
| 1492 | for (; it != report_blocks.end(); ++it) { |
srte | 3e69e5c | 2017-08-09 06:13:45 -0700 | [diff] [blame] | 1493 | if (it->sender_ssrc == remoteSSRC) |
minyue@webrtc.org | 2b58a44 | 2014-09-11 07:51:53 +0000 | [diff] [blame] | 1494 | break; |
| 1495 | } |
| 1496 | if (it == report_blocks.end()) { |
| 1497 | // We have not received packets with SSRC matching the report blocks. |
| 1498 | // To calculate RTT we try with the SSRC of the first report block. |
| 1499 | // This is very important for send-only channels where we don't know |
| 1500 | // the SSRC of the other end. |
srte | 3e69e5c | 2017-08-09 06:13:45 -0700 | [diff] [blame] | 1501 | remoteSSRC = report_blocks[0].sender_ssrc; |
minyue@webrtc.org | 2b58a44 | 2014-09-11 07:51:53 +0000 | [diff] [blame] | 1502 | } |
Minyue | 2013aec | 2015-05-13 14:14:42 +0200 | [diff] [blame] | 1503 | |
pkasting@chromium.org | 16825b1 | 2015-01-12 21:51:21 +0000 | [diff] [blame] | 1504 | int64_t avg_rtt = 0; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1505 | int64_t max_rtt = 0; |
pkasting@chromium.org | 16825b1 | 2015-01-12 21:51:21 +0000 | [diff] [blame] | 1506 | int64_t min_rtt = 0; |
kwiberg | 55b97fe | 2016-01-28 05:22:45 -0800 | [diff] [blame] | 1507 | if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
| 1508 | 0) { |
minyue@webrtc.org | 2b58a44 | 2014-09-11 07:51:53 +0000 | [diff] [blame] | 1509 | return 0; |
| 1510 | } |
pkasting@chromium.org | 16825b1 | 2015-01-12 21:51:21 +0000 | [diff] [blame] | 1511 | return rtt; |
minyue@webrtc.org | 2b58a44 | 2014-09-11 07:51:53 +0000 | [diff] [blame] | 1512 | } |
| 1513 | |
pbos@webrtc.org | d900e8b | 2013-07-03 15:12:26 +0000 | [diff] [blame] | 1514 | } // namespace voe |
| 1515 | } // namespace webrtc |