| /* |
| * 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/strings/string_builder.h" |
| #include "rtc_base/synchronization/mutex.h" |
| #include "rtc_base/system/no_unique_address.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 ChannelSend : public ChannelSendInterface, |
| public AudioPacketizationCallback, // receive encoded |
| // packets from the ACM |
| public RtcpPacketTypeCounterObserver, |
| public ReportBlockDataObserver { |
| 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, |
| RtpTransportControllerSendInterface* transport_controller, |
| const FieldTrialsView& field_trials); |
| |
| ~ChannelSend() override; |
| |
| // Send using this encoder, with this payload type. |
| void SetEncoder(int payload_type, |
| const SdpAudioFormat& encoder_format, |
| 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) override; |
| void ResetSenderCongestionControlObjects() override; |
| void SetRTCP_CNAME(absl::string_view c_name) override; |
| std::vector<ReportBlockData> 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; |
| |
| // ReportBlockDataObserver. |
| void OnReportBlockDataUpdated(ReportBlockData report_block) override; |
| |
| 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_without_offset, |
| rtc::ArrayView<const uint8_t> payload, |
| int64_t absolute_capture_timestamp_ms, |
| rtc::ArrayView<const uint32_t> csrcs, |
| absl::optional<uint8_t> audio_level_dbov) |
| RTC_RUN_ON(encoder_queue_checker_); |
| |
| 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. |
| RTC_NO_UNIQUE_ADDRESS 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; |
| absl::optional<int64_t> last_capture_timestamp_ms_ |
| RTC_GUARDED_BY(audio_thread_race_checker_); |
| |
| RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_checker_); |
| bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_) = false; |
| bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_checker_) = false; |
| |
| PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) = |
| nullptr; |
| const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_; |
| const std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
| |
| RTC_NO_UNIQUE_ADDRESS SequenceChecker construction_thread_; |
| |
| std::atomic<bool> include_audio_level_indication_ = false; |
| std::atomic<bool> encoder_queue_is_active_ = false; |
| std::atomic<bool> first_frame_ = true; |
| |
| // E2EE Audio Frame Encryption |
| rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_ |
| RTC_GUARDED_BY(encoder_queue_checker_); |
| // 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_checker_); |
| |
| mutable Mutex rtcp_counter_mutex_; |
| RtcpPacketTypeCounter rtcp_packet_type_counter_ |
| RTC_GUARDED_BY(rtcp_counter_mutex_); |
| |
| std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue_; |
| RTC_NO_UNIQUE_ADDRESS SequenceChecker encoder_queue_checker_; |
| |
| SdpAudioFormat encoder_format_; |
| }; |
| |
| 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: |
| RTC_NO_UNIQUE_ADDRESS SequenceChecker thread_checker_; |
| Mutex mutex_; |
| RtpPacketSender* rtp_packet_pacer_ 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_checker_); |
| rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize); |
| |
| absl::optional<uint8_t> audio_level_dbov; |
| if (include_audio_level_indication_.load()) { |
| // Take the averaged audio levels from rms_level_ and reset it before |
| // invoking any async transformer. |
| audio_level_dbov = rms_level_.Average(); |
| } |
| |
| if (frame_transformer_delegate_) { |
| // Asynchronously transform the payload before sending it. After the payload |
| // is transformed, the delegate will call SendRtpAudio to send it. |
| char buf[1024]; |
| rtc::SimpleStringBuilder mime_type(buf); |
| mime_type << MediaTypeToString(cricket::MEDIA_TYPE_AUDIO) << "/" |
| << encoder_format_.name; |
| frame_transformer_delegate_->Transform( |
| frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(), |
| payloadData, payloadSize, absolute_capture_timestamp_ms, |
| rtp_rtcp_->SSRC(), mime_type.str(), audio_level_dbov); |
| return 0; |
| } |
| return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload, |
| absolute_capture_timestamp_ms, /*csrcs=*/{}, |
| audio_level_dbov); |
| } |
| |
| int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType, |
| uint8_t payloadType, |
| uint32_t rtp_timestamp_without_offset, |
| rtc::ArrayView<const uint8_t> payload, |
| int64_t absolute_capture_timestamp_ms, |
| rtc::ArrayView<const uint32_t> csrcs, |
| absl::optional<uint8_t> audio_level_dbov) { |
| // 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_without_offset, |
| // 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. |
| RTPSenderAudio::RtpAudioFrame frame = { |
| .type = frameType, |
| .payload = payload, |
| .payload_id = payloadType, |
| .rtp_timestamp = |
| rtp_timestamp_without_offset + rtp_rtcp_->StartTimestamp(), |
| .csrcs = csrcs}; |
| if (absolute_capture_timestamp_ms > 0) { |
| frame.capture_time = Timestamp::Millis(absolute_capture_timestamp_ms); |
| } |
| if (include_audio_level_indication_.load() && audio_level_dbov) { |
| frame.audio_level_dbov = *audio_level_dbov; |
| } |
| if (!rtp_sender_audio_->SendAudio(frame)) { |
| 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, |
| RtpTransportControllerSendInterface* transport_controller, |
| const FieldTrialsView& field_trials) |
| : ssrc_(ssrc), |
| event_log_(rtc_event_log), |
| 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)), |
| encoder_queue_checker_(encoder_queue_.get()), |
| encoder_format_("x-unknown", 0, 0) { |
| audio_coding_ = AudioCodingModule::Create(); |
| |
| RtpRtcpInterface::Configuration configuration; |
| configuration.report_block_data_observer = this; |
| configuration.network_link_rtcp_observer = |
| transport_controller->GetRtcpObserver(); |
| configuration.clock = (clock ? clock : Clock::GetRealTimeClock()); |
| 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; |
| if (field_trials.IsDisabled("WebRTC-DisableRtxRateLimiter")) { |
| 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; |
| configuration.field_trials = &field_trials; |
| |
| 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); |
| |
| // Delete the encoder task queue first to ensure that there are no running |
| // tasks when the other members are destroyed. |
| encoder_queue_ = nullptr; |
| } |
| |
| 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. |
| first_frame_.store(true); |
| 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_checker_); |
| 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, |
| const SdpAudioFormat& encoder_format, |
| 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); |
| |
| encoder_format_ = encoder_format; |
| 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::OnReportBlockDataUpdated(ReportBlockData report_block) { |
| float packet_loss_rate = report_block.fraction_lost(); |
| 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(rtc::MakeArrayView(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(AudioLevelExtension::Uri(), id); |
| } else { |
| rtp_rtcp_->DeregisterSendRtpHeaderExtension(AudioLevelExtension::Uri()); |
| } |
| } |
| |
| void ChannelSend::RegisterSenderCongestionControlObjects( |
| RtpTransportControllerSendInterface* transport) { |
| 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_); |
| 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); |
| 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<ReportBlockData> 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. |
| return rtp_rtcp_->GetLatestReportBlockData(); |
| } |
| |
| CallSendStatistics ChannelSend::GetRTCPStatistics() const { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| CallSendStatistics stats = {0}; |
| stats.rttMs = GetRTT(); |
| |
| 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_received = 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); |
| |
| if (!encoder_queue_is_active_.load()) { |
| return; |
| } |
| |
| // Update `timestamp_` based on the capture timestamp for the first frame |
| // after sending is resumed. |
| if (first_frame_.load()) { |
| first_frame_.store(false); |
| if (last_capture_timestamp_ms_ && |
| audio_frame->absolute_capture_timestamp_ms()) { |
| int64_t diff_ms = *audio_frame->absolute_capture_timestamp_ms() - |
| *last_capture_timestamp_ms_; |
| // Truncate to whole frames and subtract one since `timestamp_` was |
| // incremented after the last frame. |
| int64_t diff_frames = diff_ms * audio_frame->sample_rate_hz() / 1000 / |
| audio_frame->samples_per_channel() - |
| 1; |
| timestamp_ += std::max<int64_t>( |
| diff_frames * audio_frame->samples_per_channel(), 0); |
| } |
| } |
| |
| audio_frame->timestamp_ = timestamp_; |
| timestamp_ += audio_frame->samples_per_channel_; |
| last_capture_timestamp_ms_ = audio_frame->absolute_capture_timestamp_ms(); |
| |
| // 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_checker_); |
| 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_checker_); |
| 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_checker_); |
| 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_checker_); |
| 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_with_offset, |
| rtc::ArrayView<const uint8_t> payload, |
| int64_t absolute_capture_timestamp_ms, |
| rtc::ArrayView<const uint32_t> csrcs, |
| absl::optional<uint8_t> audio_level_dbov) { |
| RTC_DCHECK_RUN_ON(&encoder_queue_checker_); |
| return SendRtpAudio( |
| frameType, payloadType, |
| rtp_timestamp_with_offset - rtp_rtcp_->StartTimestamp(), payload, |
| absolute_capture_timestamp_ms, csrcs, audio_level_dbov); |
| }; |
| frame_transformer_delegate_ = |
| rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>( |
| std::move(send_audio_callback), std::move(frame_transformer), |
| encoder_queue_.get()); |
| 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, |
| RtpTransportControllerSendInterface* transport_controller, |
| 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), |
| transport_controller, field_trials); |
| } |
| |
| } // namespace voe |
| } // namespace webrtc |
