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Diffstat (limited to '')
| -rw-r--r-- | third_party/libwebrtc/audio/channel_send.cc | 983 |
1 files changed, 983 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/channel_send.cc b/third_party/libwebrtc/audio/channel_send.cc new file mode 100644 index 0000000000..9609ac8a31 --- /dev/null +++ b/third_party/libwebrtc/audio/channel_send.cc @@ -0,0 +1,983 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "audio/channel_send.h" + +#include <algorithm> +#include <map> +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "api/array_view.h" +#include "api/call/transport.h" +#include "api/crypto/frame_encryptor_interface.h" +#include "api/rtc_event_log/rtc_event_log.h" +#include "api/sequence_checker.h" +#include "audio/channel_send_frame_transformer_delegate.h" +#include "audio/utility/audio_frame_operations.h" +#include "call/rtp_transport_controller_send_interface.h" +#include "logging/rtc_event_log/events/rtc_event_audio_playout.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" +#include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_processing/rms_level.h" +#include "modules/pacing/packet_router.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" +#include "rtc_base/checks.h" +#include "rtc_base/event.h" +#include "rtc_base/logging.h" +#include "rtc_base/numerics/safe_conversions.h" +#include "rtc_base/race_checker.h" +#include "rtc_base/rate_limiter.h" +#include "rtc_base/synchronization/mutex.h" +#include "rtc_base/task_queue.h" +#include "rtc_base/time_utils.h" +#include "rtc_base/trace_event.h" +#include "system_wrappers/include/clock.h" +#include "system_wrappers/include/metrics.h" + +namespace webrtc { +namespace voe { + +namespace { + +constexpr int64_t kMaxRetransmissionWindowMs = 1000; +constexpr int64_t kMinRetransmissionWindowMs = 30; + +class RtpPacketSenderProxy; +class TransportSequenceNumberProxy; +class VoERtcpObserver; + +class RtcpCounterObserver : public RtcpPacketTypeCounterObserver { + public: + explicit RtcpCounterObserver(uint32_t ssrc) : ssrc_(ssrc) {} + + void RtcpPacketTypesCounterUpdated( + uint32_t ssrc, const RtcpPacketTypeCounter& packet_counter) override { + if (ssrc_ != ssrc) { + return; + } + + MutexLock lock(&mutex_); + packet_counter_ = packet_counter; + } + + RtcpPacketTypeCounter GetCounts() { + MutexLock lock(&mutex_); + return packet_counter_; + } + + private: + Mutex mutex_; + const uint32_t ssrc_; + RtcpPacketTypeCounter packet_counter_; +}; + +class ChannelSend : public ChannelSendInterface, + public AudioPacketizationCallback, // receive encoded + // packets from the ACM + public RtcpPacketTypeCounterObserver { + public: + ChannelSend(Clock* clock, + TaskQueueFactory* task_queue_factory, + Transport* rtp_transport, + RtcpRttStats* rtcp_rtt_stats, + RtcEventLog* rtc_event_log, + FrameEncryptorInterface* frame_encryptor, + const webrtc::CryptoOptions& crypto_options, + bool extmap_allow_mixed, + int rtcp_report_interval_ms, + uint32_t ssrc, + rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, + TransportFeedbackObserver* feedback_observer, + const FieldTrialsView& field_trials); + + ~ChannelSend() override; + + // Send using this encoder, with this payload type. + void SetEncoder(int payload_type, + std::unique_ptr<AudioEncoder> encoder) override; + void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> + modifier) override; + void CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) override; + + // API methods + void StartSend() override; + void StopSend() override; + + // Codecs + void OnBitrateAllocation(BitrateAllocationUpdate update) override; + int GetTargetBitrate() const override; + + // Network + void ReceivedRTCPPacket(const uint8_t* data, size_t length) override; + + // Muting, Volume and Level. + void SetInputMute(bool enable) override; + + // Stats. + ANAStats GetANAStatistics() const override; + + // Used by AudioSendStream. + RtpRtcpInterface* GetRtpRtcp() const override; + + void RegisterCngPayloadType(int payload_type, int payload_frequency) override; + + // DTMF. + bool SendTelephoneEventOutband(int event, int duration_ms) override; + void SetSendTelephoneEventPayloadType(int payload_type, + int payload_frequency) override; + + // RTP+RTCP + void SetSendAudioLevelIndicationStatus(bool enable, int id) override; + + void RegisterSenderCongestionControlObjects( + RtpTransportControllerSendInterface* transport, + RtcpBandwidthObserver* bandwidth_observer) override; + void ResetSenderCongestionControlObjects() override; + void SetRTCP_CNAME(absl::string_view c_name) override; + std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const override; + CallSendStatistics GetRTCPStatistics() const override; + + // ProcessAndEncodeAudio() posts a task on the shared encoder task queue, + // which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where + // the actual processing of the audio takes place. The processing mainly + // consists of encoding and preparing the result for sending by adding it to a + // send queue. + // The main reason for using a task queue here is to release the native, + // OS-specific, audio capture thread as soon as possible to ensure that it + // can go back to sleep and be prepared to deliver an new captured audio + // packet. + void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) override; + + int64_t GetRTT() const override; + + // E2EE Custom Audio Frame Encryption + void SetFrameEncryptor( + rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override; + + // Sets a frame transformer between encoder and packetizer, to transform + // encoded frames before sending them out the network. + void SetEncoderToPacketizerFrameTransformer( + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) + override; + + // RtcpPacketTypeCounterObserver. + void RtcpPacketTypesCounterUpdated( + uint32_t ssrc, + const RtcpPacketTypeCounter& packet_counter) override; + + void OnUplinkPacketLossRate(float packet_loss_rate); + + private: + // From AudioPacketizationCallback in the ACM + int32_t SendData(AudioFrameType frameType, + uint8_t payloadType, + uint32_t rtp_timestamp, + const uint8_t* payloadData, + size_t payloadSize, + int64_t absolute_capture_timestamp_ms) override; + + bool InputMute() const; + + int32_t SendRtpAudio(AudioFrameType frameType, + uint8_t payloadType, + uint32_t rtp_timestamp, + rtc::ArrayView<const uint8_t> payload, + int64_t absolute_capture_timestamp_ms) + RTC_RUN_ON(encoder_queue_); + + void OnReceivedRtt(int64_t rtt_ms); + + void InitFrameTransformerDelegate( + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer); + + // Thread checkers document and lock usage of some methods on voe::Channel to + // specific threads we know about. The goal is to eventually split up + // voe::Channel into parts with single-threaded semantics, and thereby reduce + // the need for locks. + SequenceChecker worker_thread_checker_; + // Methods accessed from audio and video threads are checked for sequential- + // only access. We don't necessarily own and control these threads, so thread + // checkers cannot be used. E.g. Chromium may transfer "ownership" from one + // audio thread to another, but access is still sequential. + rtc::RaceChecker audio_thread_race_checker_; + + mutable Mutex volume_settings_mutex_; + + const uint32_t ssrc_; + bool sending_ RTC_GUARDED_BY(&worker_thread_checker_) = false; + + RtcEventLog* const event_log_; + + std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; + std::unique_ptr<RTPSenderAudio> rtp_sender_audio_; + + std::unique_ptr<AudioCodingModule> audio_coding_; + + // This is just an offset, RTP module will add its own random offset. + uint32_t timestamp_ RTC_GUARDED_BY(audio_thread_race_checker_) = 0; + + RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_); + bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_) = false; + bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_) = false; + + // RtcpBandwidthObserver + const std::unique_ptr<VoERtcpObserver> rtcp_observer_; + + const std::unique_ptr<RtcpCounterObserver> rtcp_counter_observer_; + + PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) = + nullptr; + TransportFeedbackObserver* const feedback_observer_; + const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_; + const std::unique_ptr<RateLimiter> retransmission_rate_limiter_; + + SequenceChecker construction_thread_; + + std::atomic<bool> include_audio_level_indication_ = false; + std::atomic<bool> encoder_queue_is_active_ = false; + + // E2EE Audio Frame Encryption + rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_ + RTC_GUARDED_BY(encoder_queue_); + // E2EE Frame Encryption Options + const webrtc::CryptoOptions crypto_options_; + + // Delegates calls to a frame transformer to transform audio, and + // receives callbacks with the transformed frames; delegates calls to + // ChannelSend::SendRtpAudio to send the transformed audio. + rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> + frame_transformer_delegate_ RTC_GUARDED_BY(encoder_queue_); + + mutable Mutex rtcp_counter_mutex_; + RtcpPacketTypeCounter rtcp_packet_type_counter_ + RTC_GUARDED_BY(rtcp_counter_mutex_); + + // Defined last to ensure that there are no running tasks when the other + // members are destroyed. + rtc::TaskQueue encoder_queue_; +}; + +const int kTelephoneEventAttenuationdB = 10; + +class RtpPacketSenderProxy : public RtpPacketSender { + public: + RtpPacketSenderProxy() : rtp_packet_pacer_(nullptr) {} + + void SetPacketPacer(RtpPacketSender* rtp_packet_pacer) { + RTC_DCHECK(thread_checker_.IsCurrent()); + MutexLock lock(&mutex_); + rtp_packet_pacer_ = rtp_packet_pacer; + } + + void EnqueuePackets( + std::vector<std::unique_ptr<RtpPacketToSend>> packets) override { + MutexLock lock(&mutex_); + rtp_packet_pacer_->EnqueuePackets(std::move(packets)); + } + + void RemovePacketsForSsrc(uint32_t ssrc) override { + MutexLock lock(&mutex_); + rtp_packet_pacer_->RemovePacketsForSsrc(ssrc); + } + + private: + SequenceChecker thread_checker_; + Mutex mutex_; + RtpPacketSender* rtp_packet_pacer_ RTC_GUARDED_BY(&mutex_); +}; + +class VoERtcpObserver : public RtcpBandwidthObserver { + public: + explicit VoERtcpObserver(ChannelSend* owner) + : owner_(owner), bandwidth_observer_(nullptr) {} + ~VoERtcpObserver() override {} + + void SetBandwidthObserver(RtcpBandwidthObserver* bandwidth_observer) { + MutexLock lock(&mutex_); + bandwidth_observer_ = bandwidth_observer; + } + + void OnReceivedEstimatedBitrate(uint32_t bitrate) override { + MutexLock lock(&mutex_); + if (bandwidth_observer_) { + bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate); + } + } + + void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks, + int64_t rtt, + int64_t now_ms) override { + { + MutexLock lock(&mutex_); + if (bandwidth_observer_) { + bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, rtt, + now_ms); + } + } + // TODO(mflodman): Do we need to aggregate reports here or can we jut send + // what we get? I.e. do we ever get multiple reports bundled into one RTCP + // report for VoiceEngine? + if (report_blocks.empty()) + return; + + int fraction_lost_aggregate = 0; + int total_number_of_packets = 0; + + // If receiving multiple report blocks, calculate the weighted average based + // on the number of packets a report refers to. + for (ReportBlockList::const_iterator block_it = report_blocks.begin(); + block_it != report_blocks.end(); ++block_it) { + // Find the previous extended high sequence number for this remote SSRC, + // to calculate the number of RTP packets this report refers to. Ignore if + // we haven't seen this SSRC before. + std::map<uint32_t, uint32_t>::iterator seq_num_it = + extended_max_sequence_number_.find(block_it->source_ssrc); + int number_of_packets = 0; + if (seq_num_it != extended_max_sequence_number_.end()) { + number_of_packets = + block_it->extended_highest_sequence_number - seq_num_it->second; + } + fraction_lost_aggregate += number_of_packets * block_it->fraction_lost; + total_number_of_packets += number_of_packets; + + extended_max_sequence_number_[block_it->source_ssrc] = + block_it->extended_highest_sequence_number; + } + int weighted_fraction_lost = 0; + if (total_number_of_packets > 0) { + weighted_fraction_lost = + (fraction_lost_aggregate + total_number_of_packets / 2) / + total_number_of_packets; + } + owner_->OnUplinkPacketLossRate(weighted_fraction_lost / 255.0f); + } + + private: + ChannelSend* owner_; + // Maps remote side ssrc to extended highest sequence number received. + std::map<uint32_t, uint32_t> extended_max_sequence_number_; + Mutex mutex_; + RtcpBandwidthObserver* bandwidth_observer_ RTC_GUARDED_BY(mutex_); +}; + +int32_t ChannelSend::SendData(AudioFrameType frameType, + uint8_t payloadType, + uint32_t rtp_timestamp, + const uint8_t* payloadData, + size_t payloadSize, + int64_t absolute_capture_timestamp_ms) { + RTC_DCHECK_RUN_ON(&encoder_queue_); + rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize); + if (frame_transformer_delegate_) { + // Asynchronously transform the payload before sending it. After the payload + // is transformed, the delegate will call SendRtpAudio to send it. + frame_transformer_delegate_->Transform( + frameType, payloadType, rtp_timestamp, rtp_rtcp_->StartTimestamp(), + payloadData, payloadSize, absolute_capture_timestamp_ms, + rtp_rtcp_->SSRC()); + return 0; + } + return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload, + absolute_capture_timestamp_ms); +} + +int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType, + uint8_t payloadType, + uint32_t rtp_timestamp, + rtc::ArrayView<const uint8_t> payload, + int64_t absolute_capture_timestamp_ms) { + if (include_audio_level_indication_.load()) { + // Store current audio level in the RTP sender. + // The level will be used in combination with voice-activity state + // (frameType) to add an RTP header extension + rtp_sender_audio_->SetAudioLevel(rms_level_.Average()); + } + + // E2EE Custom Audio Frame Encryption (This is optional). + // Keep this buffer around for the lifetime of the send call. + rtc::Buffer encrypted_audio_payload; + // We don't invoke encryptor if payload is empty, which means we are to send + // DTMF, or the encoder entered DTX. + // TODO(minyue): see whether DTMF packets should be encrypted or not. In + // current implementation, they are not. + if (!payload.empty()) { + if (frame_encryptor_ != nullptr) { + // TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline. + // Allocate a buffer to hold the maximum possible encrypted payload. + size_t max_ciphertext_size = frame_encryptor_->GetMaxCiphertextByteSize( + cricket::MEDIA_TYPE_AUDIO, payload.size()); + encrypted_audio_payload.SetSize(max_ciphertext_size); + + // Encrypt the audio payload into the buffer. + size_t bytes_written = 0; + int encrypt_status = frame_encryptor_->Encrypt( + cricket::MEDIA_TYPE_AUDIO, rtp_rtcp_->SSRC(), + /*additional_data=*/nullptr, payload, encrypted_audio_payload, + &bytes_written); + if (encrypt_status != 0) { + RTC_DLOG(LS_ERROR) + << "Channel::SendData() failed encrypt audio payload: " + << encrypt_status; + return -1; + } + // Resize the buffer to the exact number of bytes actually used. + encrypted_audio_payload.SetSize(bytes_written); + // Rewrite the payloadData and size to the new encrypted payload. + payload = encrypted_audio_payload; + } else if (crypto_options_.sframe.require_frame_encryption) { + RTC_DLOG(LS_ERROR) << "Channel::SendData() failed sending audio payload: " + "A frame encryptor is required but one is not set."; + return -1; + } + } + + // Push data from ACM to RTP/RTCP-module to deliver audio frame for + // packetization. + if (!rtp_rtcp_->OnSendingRtpFrame(rtp_timestamp, + // Leaving the time when this frame was + // received from the capture device as + // undefined for voice for now. + -1, payloadType, + /*force_sender_report=*/false)) { + return -1; + } + + // RTCPSender has it's own copy of the timestamp offset, added in + // RTCPSender::BuildSR, hence we must not add the in the offset for the above + // call. + // TODO(nisse): Delete RTCPSender:timestamp_offset_, and see if we can confine + // knowledge of the offset to a single place. + + // This call will trigger Transport::SendPacket() from the RTP/RTCP module. + if (!rtp_sender_audio_->SendAudio( + frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(), + payload.data(), payload.size(), absolute_capture_timestamp_ms)) { + RTC_DLOG(LS_ERROR) + << "ChannelSend::SendData() failed to send data to RTP/RTCP module"; + return -1; + } + + return 0; +} + +ChannelSend::ChannelSend( + Clock* clock, + TaskQueueFactory* task_queue_factory, + Transport* rtp_transport, + RtcpRttStats* rtcp_rtt_stats, + RtcEventLog* rtc_event_log, + FrameEncryptorInterface* frame_encryptor, + const webrtc::CryptoOptions& crypto_options, + bool extmap_allow_mixed, + int rtcp_report_interval_ms, + uint32_t ssrc, + rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, + TransportFeedbackObserver* feedback_observer, + const FieldTrialsView& field_trials) + : ssrc_(ssrc), + event_log_(rtc_event_log), + rtcp_observer_(new VoERtcpObserver(this)), + rtcp_counter_observer_(new RtcpCounterObserver(ssrc)), + feedback_observer_(feedback_observer), + rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()), + retransmission_rate_limiter_( + new RateLimiter(clock, kMaxRetransmissionWindowMs)), + frame_encryptor_(frame_encryptor), + crypto_options_(crypto_options), + encoder_queue_(task_queue_factory->CreateTaskQueue( + "AudioEncoder", + TaskQueueFactory::Priority::NORMAL)) { + audio_coding_.reset(AudioCodingModule::Create(AudioCodingModule::Config())); + + RtpRtcpInterface::Configuration configuration; + configuration.bandwidth_callback = rtcp_observer_.get(); + configuration.transport_feedback_callback = feedback_observer_; + configuration.clock = clock; + configuration.audio = true; + configuration.outgoing_transport = rtp_transport; + + configuration.paced_sender = rtp_packet_pacer_proxy_.get(); + + configuration.event_log = event_log_; + configuration.rtt_stats = rtcp_rtt_stats; + configuration.rtcp_packet_type_counter_observer = + rtcp_counter_observer_.get(); + configuration.retransmission_rate_limiter = + retransmission_rate_limiter_.get(); + configuration.extmap_allow_mixed = extmap_allow_mixed; + configuration.rtcp_report_interval_ms = rtcp_report_interval_ms; + configuration.rtcp_packet_type_counter_observer = this; + + configuration.local_media_ssrc = ssrc; + + rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration); + rtp_rtcp_->SetSendingMediaStatus(false); + + rtp_sender_audio_ = std::make_unique<RTPSenderAudio>(configuration.clock, + rtp_rtcp_->RtpSender()); + + // Ensure that RTCP is enabled by default for the created channel. + rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound); + + int error = audio_coding_->RegisterTransportCallback(this); + RTC_DCHECK_EQ(0, error); + if (frame_transformer) + InitFrameTransformerDelegate(std::move(frame_transformer)); +} + +ChannelSend::~ChannelSend() { + RTC_DCHECK(construction_thread_.IsCurrent()); + + // Resets the delegate's callback to ChannelSend::SendRtpAudio. + if (frame_transformer_delegate_) + frame_transformer_delegate_->Reset(); + + StopSend(); + int error = audio_coding_->RegisterTransportCallback(NULL); + RTC_DCHECK_EQ(0, error); +} + +void ChannelSend::StartSend() { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + RTC_DCHECK(!sending_); + sending_ = true; + + RTC_DCHECK(packet_router_); + packet_router_->AddSendRtpModule(rtp_rtcp_.get(), /*remb_candidate=*/false); + rtp_rtcp_->SetSendingMediaStatus(true); + int ret = rtp_rtcp_->SetSendingStatus(true); + RTC_DCHECK_EQ(0, ret); + + // It is now OK to start processing on the encoder task queue. + encoder_queue_is_active_.store(true); +} + +void ChannelSend::StopSend() { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + if (!sending_) { + return; + } + sending_ = false; + encoder_queue_is_active_.store(false); + + // Wait until all pending encode tasks are executed and clear any remaining + // buffers in the encoder. + rtc::Event flush; + encoder_queue_.PostTask([this, &flush]() { + RTC_DCHECK_RUN_ON(&encoder_queue_); + CallEncoder([](AudioEncoder* encoder) { encoder->Reset(); }); + flush.Set(); + }); + flush.Wait(rtc::Event::kForever); + + // Reset sending SSRC and sequence number and triggers direct transmission + // of RTCP BYE + if (rtp_rtcp_->SetSendingStatus(false) == -1) { + RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending"; + } + rtp_rtcp_->SetSendingMediaStatus(false); + + RTC_DCHECK(packet_router_); + packet_router_->RemoveSendRtpModule(rtp_rtcp_.get()); + rtp_packet_pacer_proxy_->RemovePacketsForSsrc(rtp_rtcp_->SSRC()); +} + +void ChannelSend::SetEncoder(int payload_type, + std::unique_ptr<AudioEncoder> encoder) { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + RTC_DCHECK_GE(payload_type, 0); + RTC_DCHECK_LE(payload_type, 127); + + // The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate) + // as well as some other things, so we collect this info and send it along. + rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, + encoder->RtpTimestampRateHz()); + rtp_sender_audio_->RegisterAudioPayload("audio", payload_type, + encoder->RtpTimestampRateHz(), + encoder->NumChannels(), 0); + + audio_coding_->SetEncoder(std::move(encoder)); +} + +void ChannelSend::ModifyEncoder( + rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) { + // This method can be called on the worker thread, module process thread + // or network thread. Audio coding is thread safe, so we do not need to + // enforce the calling thread. + audio_coding_->ModifyEncoder(modifier); +} + +void ChannelSend::CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) { + ModifyEncoder([modifier](std::unique_ptr<AudioEncoder>* encoder_ptr) { + if (*encoder_ptr) { + modifier(encoder_ptr->get()); + } else { + RTC_DLOG(LS_WARNING) << "Trying to call unset encoder."; + } + }); +} + +void ChannelSend::OnBitrateAllocation(BitrateAllocationUpdate update) { + // This method can be called on the worker thread, module process thread + // or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged. + // TODO(solenberg): Figure out a good way to check this or enforce calling + // rules. + // RTC_DCHECK(worker_thread_checker_.IsCurrent() || + // module_process_thread_checker_.IsCurrent()); + CallEncoder([&](AudioEncoder* encoder) { + encoder->OnReceivedUplinkAllocation(update); + }); + retransmission_rate_limiter_->SetMaxRate(update.target_bitrate.bps()); +} + +int ChannelSend::GetTargetBitrate() const { + return audio_coding_->GetTargetBitrate(); +} + +void ChannelSend::OnUplinkPacketLossRate(float packet_loss_rate) { + CallEncoder([&](AudioEncoder* encoder) { + encoder->OnReceivedUplinkPacketLossFraction(packet_loss_rate); + }); +} + +void ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + + // Deliver RTCP packet to RTP/RTCP module for parsing + rtp_rtcp_->IncomingRtcpPacket(data, length); + + int64_t rtt = GetRTT(); + if (rtt == 0) { + // Waiting for valid RTT. + return; + } + + int64_t nack_window_ms = rtt; + if (nack_window_ms < kMinRetransmissionWindowMs) { + nack_window_ms = kMinRetransmissionWindowMs; + } else if (nack_window_ms > kMaxRetransmissionWindowMs) { + nack_window_ms = kMaxRetransmissionWindowMs; + } + retransmission_rate_limiter_->SetWindowSize(nack_window_ms); + + OnReceivedRtt(rtt); +} + +void ChannelSend::SetInputMute(bool enable) { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + MutexLock lock(&volume_settings_mutex_); + input_mute_ = enable; +} + +bool ChannelSend::InputMute() const { + MutexLock lock(&volume_settings_mutex_); + return input_mute_; +} + +bool ChannelSend::SendTelephoneEventOutband(int event, int duration_ms) { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + RTC_DCHECK_LE(0, event); + RTC_DCHECK_GE(255, event); + RTC_DCHECK_LE(0, duration_ms); + RTC_DCHECK_GE(65535, duration_ms); + if (!sending_) { + return false; + } + if (rtp_sender_audio_->SendTelephoneEvent( + event, duration_ms, kTelephoneEventAttenuationdB) != 0) { + RTC_DLOG(LS_ERROR) << "SendTelephoneEvent() failed to send event"; + return false; + } + return true; +} + +void ChannelSend::RegisterCngPayloadType(int payload_type, + int payload_frequency) { + rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, payload_frequency); + rtp_sender_audio_->RegisterAudioPayload("CN", payload_type, payload_frequency, + 1, 0); +} + +void ChannelSend::SetSendTelephoneEventPayloadType(int payload_type, + int payload_frequency) { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + RTC_DCHECK_LE(0, payload_type); + RTC_DCHECK_GE(127, payload_type); + rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, payload_frequency); + rtp_sender_audio_->RegisterAudioPayload("telephone-event", payload_type, + payload_frequency, 0, 0); +} + +void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + include_audio_level_indication_.store(enable); + if (enable) { + rtp_rtcp_->RegisterRtpHeaderExtension(webrtc::AudioLevel::Uri(), id); + } else { + rtp_rtcp_->DeregisterSendRtpHeaderExtension(webrtc::AudioLevel::Uri()); + } +} + +void ChannelSend::RegisterSenderCongestionControlObjects( + RtpTransportControllerSendInterface* transport, + RtcpBandwidthObserver* bandwidth_observer) { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + RtpPacketSender* rtp_packet_pacer = transport->packet_sender(); + PacketRouter* packet_router = transport->packet_router(); + + RTC_DCHECK(rtp_packet_pacer); + RTC_DCHECK(packet_router); + RTC_DCHECK(!packet_router_); + rtcp_observer_->SetBandwidthObserver(bandwidth_observer); + rtp_packet_pacer_proxy_->SetPacketPacer(rtp_packet_pacer); + rtp_rtcp_->SetStorePacketsStatus(true, 600); + packet_router_ = packet_router; +} + +void ChannelSend::ResetSenderCongestionControlObjects() { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + RTC_DCHECK(packet_router_); + rtp_rtcp_->SetStorePacketsStatus(false, 600); + rtcp_observer_->SetBandwidthObserver(nullptr); + packet_router_ = nullptr; + rtp_packet_pacer_proxy_->SetPacketPacer(nullptr); +} + +void ChannelSend::SetRTCP_CNAME(absl::string_view c_name) { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + // Note: SetCNAME() accepts a c string of length at most 255. + const std::string c_name_limited(c_name.substr(0, 255)); + int ret = rtp_rtcp_->SetCNAME(c_name_limited.c_str()) != 0; + RTC_DCHECK_EQ(0, ret) << "SetRTCP_CNAME() failed to set RTCP CNAME"; +} + +std::vector<ReportBlock> ChannelSend::GetRemoteRTCPReportBlocks() const { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + // Get the report blocks from the latest received RTCP Sender or Receiver + // Report. Each element in the vector contains the sender's SSRC and a + // report block according to RFC 3550. + std::vector<ReportBlock> report_blocks; + for (const ReportBlockData& data : rtp_rtcp_->GetLatestReportBlockData()) { + ReportBlock report_block; + report_block.sender_SSRC = data.report_block().sender_ssrc; + report_block.source_SSRC = data.report_block().source_ssrc; + report_block.fraction_lost = data.report_block().fraction_lost; + report_block.cumulative_num_packets_lost = data.report_block().packets_lost; + report_block.extended_highest_sequence_number = + data.report_block().extended_highest_sequence_number; + report_block.interarrival_jitter = data.report_block().jitter; + report_block.last_SR_timestamp = + data.report_block().last_sender_report_timestamp; + report_block.delay_since_last_SR = + data.report_block().delay_since_last_sender_report; + report_blocks.push_back(report_block); + } + return report_blocks; +} + +CallSendStatistics ChannelSend::GetRTCPStatistics() const { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + CallSendStatistics stats = {0}; + stats.rttMs = GetRTT(); + stats.rtcp_packet_type_counts = rtcp_counter_observer_->GetCounts(); + + StreamDataCounters rtp_stats; + StreamDataCounters rtx_stats; + rtp_rtcp_->GetSendStreamDataCounters(&rtp_stats, &rtx_stats); + stats.payload_bytes_sent = + rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes; + stats.header_and_padding_bytes_sent = + rtp_stats.transmitted.padding_bytes + rtp_stats.transmitted.header_bytes + + rtx_stats.transmitted.padding_bytes + rtx_stats.transmitted.header_bytes; + + // TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up in + // separate outbound-rtp stream objects. + stats.retransmitted_bytes_sent = rtp_stats.retransmitted.payload_bytes; + stats.packetsSent = + rtp_stats.transmitted.packets + rtx_stats.transmitted.packets; + stats.total_packet_send_delay = rtp_stats.transmitted.total_packet_delay; + stats.retransmitted_packets_sent = rtp_stats.retransmitted.packets; + stats.report_block_datas = rtp_rtcp_->GetLatestReportBlockData(); + + { + MutexLock lock(&rtcp_counter_mutex_); + stats.nacks_rcvd = rtcp_packet_type_counter_.nack_packets; + } + + return stats; +} + +void ChannelSend::RtcpPacketTypesCounterUpdated( + uint32_t ssrc, + const RtcpPacketTypeCounter& packet_counter) { + if (ssrc != ssrc_) { + return; + } + MutexLock lock(&rtcp_counter_mutex_); + rtcp_packet_type_counter_ = packet_counter; +} + +void ChannelSend::ProcessAndEncodeAudio( + std::unique_ptr<AudioFrame> audio_frame) { + TRACE_EVENT0("webrtc", "ChannelSend::ProcessAndEncodeAudio"); + + RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_); + RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0); + RTC_DCHECK_LE(audio_frame->num_channels_, 8); + + audio_frame->timestamp_ = timestamp_; + timestamp_ += audio_frame->samples_per_channel_; + if (!encoder_queue_is_active_.load()) { + return; + } + + // Profile time between when the audio frame is added to the task queue and + // when the task is actually executed. + audio_frame->UpdateProfileTimeStamp(); + encoder_queue_.PostTask( + [this, audio_frame = std::move(audio_frame)]() mutable { + RTC_DCHECK_RUN_ON(&encoder_queue_); + if (!encoder_queue_is_active_.load()) { + return; + } + // Measure time between when the audio frame is added to the task queue + // and when the task is actually executed. Goal is to keep track of + // unwanted extra latency added by the task queue. + RTC_HISTOGRAM_COUNTS_10000("WebRTC.Audio.EncodingTaskQueueLatencyMs", + audio_frame->ElapsedProfileTimeMs()); + + bool is_muted = InputMute(); + AudioFrameOperations::Mute(audio_frame.get(), previous_frame_muted_, + is_muted); + + if (include_audio_level_indication_.load()) { + size_t length = + audio_frame->samples_per_channel_ * audio_frame->num_channels_; + RTC_CHECK_LE(length, AudioFrame::kMaxDataSizeBytes); + if (is_muted && previous_frame_muted_) { + rms_level_.AnalyzeMuted(length); + } else { + rms_level_.Analyze( + rtc::ArrayView<const int16_t>(audio_frame->data(), length)); + } + } + previous_frame_muted_ = is_muted; + + // This call will trigger AudioPacketizationCallback::SendData if + // encoding is done and payload is ready for packetization and + // transmission. Otherwise, it will return without invoking the + // callback. + if (audio_coding_->Add10MsData(*audio_frame) < 0) { + RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed."; + return; + } + }); +} + +ANAStats ChannelSend::GetANAStatistics() const { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + return audio_coding_->GetANAStats(); +} + +RtpRtcpInterface* ChannelSend::GetRtpRtcp() const { + return rtp_rtcp_.get(); +} + +int64_t ChannelSend::GetRTT() const { + std::vector<ReportBlockData> report_blocks = + rtp_rtcp_->GetLatestReportBlockData(); + if (report_blocks.empty()) { + return 0; + } + + // We don't know in advance the remote ssrc used by the other end's receiver + // reports, so use the first report block for the RTT. + return report_blocks.front().last_rtt_ms(); +} + +void ChannelSend::SetFrameEncryptor( + rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + encoder_queue_.PostTask([this, frame_encryptor]() mutable { + RTC_DCHECK_RUN_ON(&encoder_queue_); + frame_encryptor_ = std::move(frame_encryptor); + }); +} + +void ChannelSend::SetEncoderToPacketizerFrameTransformer( + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + if (!frame_transformer) + return; + + encoder_queue_.PostTask( + [this, frame_transformer = std::move(frame_transformer)]() mutable { + RTC_DCHECK_RUN_ON(&encoder_queue_); + InitFrameTransformerDelegate(std::move(frame_transformer)); + }); +} + +void ChannelSend::OnReceivedRtt(int64_t rtt_ms) { + // Invoke audio encoders OnReceivedRtt(). + CallEncoder( + [rtt_ms](AudioEncoder* encoder) { encoder->OnReceivedRtt(rtt_ms); }); +} + +void ChannelSend::InitFrameTransformerDelegate( + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) { + RTC_DCHECK_RUN_ON(&encoder_queue_); + RTC_DCHECK(frame_transformer); + RTC_DCHECK(!frame_transformer_delegate_); + + // Pass a callback to ChannelSend::SendRtpAudio, to be called by the delegate + // to send the transformed audio. + ChannelSendFrameTransformerDelegate::SendFrameCallback send_audio_callback = + [this](AudioFrameType frameType, uint8_t payloadType, + uint32_t rtp_timestamp, rtc::ArrayView<const uint8_t> payload, + int64_t absolute_capture_timestamp_ms) { + RTC_DCHECK_RUN_ON(&encoder_queue_); + return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload, + absolute_capture_timestamp_ms); + }; + frame_transformer_delegate_ = + rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>( + std::move(send_audio_callback), std::move(frame_transformer), + &encoder_queue_); + frame_transformer_delegate_->Init(); +} + +} // namespace + +std::unique_ptr<ChannelSendInterface> CreateChannelSend( + Clock* clock, + TaskQueueFactory* task_queue_factory, + Transport* rtp_transport, + RtcpRttStats* rtcp_rtt_stats, + RtcEventLog* rtc_event_log, + FrameEncryptorInterface* frame_encryptor, + const webrtc::CryptoOptions& crypto_options, + bool extmap_allow_mixed, + int rtcp_report_interval_ms, + uint32_t ssrc, + rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, + TransportFeedbackObserver* feedback_observer, + const FieldTrialsView& field_trials) { + return std::make_unique<ChannelSend>( + clock, task_queue_factory, rtp_transport, rtcp_rtt_stats, rtc_event_log, + frame_encryptor, crypto_options, extmap_allow_mixed, + rtcp_report_interval_ms, ssrc, std::move(frame_transformer), + feedback_observer, field_trials); +} + +} // namespace voe +} // namespace webrtc |