| /* |
| * 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 "absl/memory/memory.h" |
| #include "api/array_view.h" |
| #include "api/call/transport.h" |
| #include "api/crypto/frame_encryptor_interface.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 "logging/rtc_event_log/rtc_event_log.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/utility/include/process_thread.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/critical_section.h" |
| #include "rtc_base/event.h" |
| #include "rtc_base/format_macros.h" |
| #include "rtc_base/location.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/task_queue.h" |
| #include "rtc_base/thread_checker.h" |
| #include "rtc_base/time_utils.h" |
| #include "system_wrappers/include/clock.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "system_wrappers/include/metrics.h" |
| |
| namespace webrtc { |
| namespace voe { |
| |
| namespace { |
| |
| constexpr int64_t kMaxRetransmissionWindowMs = 1000; |
| constexpr int64_t kMinRetransmissionWindowMs = 30; |
| |
| MediaTransportEncodedAudioFrame::FrameType |
| MediaTransportFrameTypeForWebrtcFrameType(webrtc::FrameType frame_type) { |
| switch (frame_type) { |
| case kAudioFrameSpeech: |
| return MediaTransportEncodedAudioFrame::FrameType::kSpeech; |
| break; |
| |
| case kAudioFrameCN: |
| return MediaTransportEncodedAudioFrame::FrameType:: |
| kDiscontinuousTransmission; |
| break; |
| |
| default: |
| RTC_CHECK(false) << "Unexpected frame type=" << frame_type; |
| break; |
| } |
| } |
| |
| class RtpPacketSenderProxy; |
| class TransportFeedbackProxy; |
| class TransportSequenceNumberProxy; |
| class VoERtcpObserver; |
| |
| class ChannelSend |
| : public ChannelSendInterface, |
| public AudioPacketizationCallback, // receive encoded packets from the |
| // ACM |
| public TargetTransferRateObserver { |
| public: |
| // TODO(nisse): Make OnUplinkPacketLossRate public, and delete friend |
| // declaration. |
| friend class VoERtcpObserver; |
| |
| ChannelSend(Clock* clock, |
| rtc::TaskQueue* encoder_queue, |
| ProcessThread* module_process_thread, |
| MediaTransportInterface* media_transport, |
| OverheadObserver* overhead_observer, |
| 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); |
| |
| ~ChannelSend() override; |
| |
| // Send using this encoder, with this payload type. |
| bool 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 GetBitrate() const override; |
| |
| // Network |
| bool 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. |
| RtpRtcp* GetRtpRtcp() const override; |
| |
| // DTMF. |
| bool SendTelephoneEventOutband(int event, int duration_ms) override; |
| bool SetSendTelephoneEventPayloadType(int payload_type, |
| int payload_frequency) override; |
| |
| // RTP+RTCP |
| void SetLocalSSRC(uint32_t ssrc) override; |
| void SetRid(const std::string& rid, |
| int extension_id, |
| int repaired_extension_id) override; |
| void SetMid(const std::string& mid, int extension_id) override; |
| void SetExtmapAllowMixed(bool extmap_allow_mixed) override; |
| void SetSendAudioLevelIndicationStatus(bool enable, int id) override; |
| void EnableSendTransportSequenceNumber(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; |
| |
| // The existence of this function alongside OnUplinkPacketLossRate is |
| // a compromise. We want the encoder to be agnostic of the PLR source, but |
| // we also don't want it to receive conflicting information from TWCC and |
| // from RTCP-XR. |
| void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) override; |
| |
| void OnRecoverableUplinkPacketLossRate( |
| float recoverable_packet_loss_rate) override; |
| |
| int64_t GetRTT() const override; |
| |
| // E2EE Custom Audio Frame Encryption |
| void SetFrameEncryptor( |
| rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override; |
| |
| private: |
| class ProcessAndEncodeAudioTask; |
| |
| // From AudioPacketizationCallback in the ACM |
| int32_t SendData(FrameType frameType, |
| uint8_t payloadType, |
| uint32_t timeStamp, |
| const uint8_t* payloadData, |
| size_t payloadSize, |
| const RTPFragmentationHeader* fragmentation) override; |
| |
| void OnUplinkPacketLossRate(float packet_loss_rate); |
| bool InputMute() const; |
| |
| int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type, int id); |
| |
| int32_t SendRtpAudio(FrameType frameType, |
| uint8_t payloadType, |
| uint32_t timeStamp, |
| rtc::ArrayView<const uint8_t> payload, |
| const RTPFragmentationHeader* fragmentation); |
| |
| int32_t SendMediaTransportAudio(FrameType frameType, |
| uint8_t payloadType, |
| uint32_t timeStamp, |
| rtc::ArrayView<const uint8_t> payload, |
| const RTPFragmentationHeader* fragmentation); |
| |
| // Return media transport or nullptr if using RTP. |
| MediaTransportInterface* media_transport() { return media_transport_; } |
| |
| // Called on the encoder task queue when a new input audio frame is ready |
| // for encoding. |
| void ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input); |
| |
| void OnReceivedRtt(int64_t rtt_ms); |
| |
| void OnTargetTransferRate(TargetTransferRate) override; |
| |
| // 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::ThreadChecker worker_thread_checker_; |
| rtc::ThreadChecker module_process_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_; |
| |
| rtc::CriticalSection volume_settings_critsect_; |
| |
| bool sending_ RTC_GUARDED_BY(&worker_thread_checker_) = false; |
| |
| RtcEventLog* const event_log_; |
| |
| std::unique_ptr<RtpRtcp> _rtpRtcpModule; |
| |
| std::unique_ptr<AudioCodingModule> audio_coding_; |
| uint32_t _timeStamp RTC_GUARDED_BY(encoder_queue_); |
| |
| // uses |
| ProcessThread* const _moduleProcessThreadPtr; |
| RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_); |
| bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_); |
| bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_); |
| // VoeRTP_RTCP |
| // TODO(henrika): can today be accessed on the main thread and on the |
| // task queue; hence potential race. |
| bool _includeAudioLevelIndication; |
| |
| // RtcpBandwidthObserver |
| const std::unique_ptr<VoERtcpObserver> rtcp_observer_; |
| |
| PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) = |
| nullptr; |
| const std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
| const std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
| const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
| const std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
| |
| rtc::ThreadChecker construction_thread_; |
| |
| const bool use_twcc_plr_for_ana_; |
| |
| rtc::CriticalSection encoder_queue_lock_; |
| bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false; |
| rtc::TaskQueue* const encoder_queue_ = nullptr; |
| |
| MediaTransportInterface* const media_transport_; |
| int media_transport_sequence_number_ RTC_GUARDED_BY(encoder_queue_) = 0; |
| |
| rtc::CriticalSection media_transport_lock_; |
| // Currently set by SetLocalSSRC. |
| uint64_t media_transport_channel_id_ RTC_GUARDED_BY(&media_transport_lock_) = |
| 0; |
| // Cache payload type and sampling frequency from most recent call to |
| // SetEncoder. Needed to set MediaTransportEncodedAudioFrame metadata, and |
| // invalidate on encoder change. |
| int media_transport_payload_type_ RTC_GUARDED_BY(&media_transport_lock_); |
| int media_transport_sampling_frequency_ |
| RTC_GUARDED_BY(&media_transport_lock_); |
| |
| // E2EE Audio Frame Encryption |
| rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_; |
| // E2EE Frame Encryption Options |
| const webrtc::CryptoOptions crypto_options_; |
| |
| rtc::CriticalSection bitrate_crit_section_; |
| int configured_bitrate_bps_ RTC_GUARDED_BY(bitrate_crit_section_) = 0; |
| }; |
| |
| const int kTelephoneEventAttenuationdB = 10; |
| |
| class TransportFeedbackProxy : public TransportFeedbackObserver { |
| public: |
| TransportFeedbackProxy() : feedback_observer_(nullptr) { |
| pacer_thread_.DetachFromThread(); |
| network_thread_.DetachFromThread(); |
| } |
| |
| void SetTransportFeedbackObserver( |
| TransportFeedbackObserver* feedback_observer) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| rtc::CritScope lock(&crit_); |
| feedback_observer_ = feedback_observer; |
| } |
| |
| // Implements TransportFeedbackObserver. |
| void AddPacket(uint32_t ssrc, |
| uint16_t sequence_number, |
| size_t length, |
| const PacedPacketInfo& pacing_info) override { |
| RTC_DCHECK(pacer_thread_.CalledOnValidThread()); |
| rtc::CritScope lock(&crit_); |
| if (feedback_observer_) |
| feedback_observer_->AddPacket(ssrc, sequence_number, length, pacing_info); |
| } |
| |
| void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override { |
| RTC_DCHECK(network_thread_.CalledOnValidThread()); |
| rtc::CritScope lock(&crit_); |
| if (feedback_observer_) |
| feedback_observer_->OnTransportFeedback(feedback); |
| } |
| |
| private: |
| rtc::CriticalSection crit_; |
| rtc::ThreadChecker thread_checker_; |
| rtc::ThreadChecker pacer_thread_; |
| rtc::ThreadChecker network_thread_; |
| TransportFeedbackObserver* feedback_observer_ RTC_GUARDED_BY(&crit_); |
| }; |
| |
| class TransportSequenceNumberProxy : public TransportSequenceNumberAllocator { |
| public: |
| TransportSequenceNumberProxy() : seq_num_allocator_(nullptr) { |
| pacer_thread_.DetachFromThread(); |
| } |
| |
| void SetSequenceNumberAllocator( |
| TransportSequenceNumberAllocator* seq_num_allocator) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| rtc::CritScope lock(&crit_); |
| seq_num_allocator_ = seq_num_allocator; |
| } |
| |
| // Implements TransportSequenceNumberAllocator. |
| uint16_t AllocateSequenceNumber() override { |
| RTC_DCHECK(pacer_thread_.CalledOnValidThread()); |
| rtc::CritScope lock(&crit_); |
| if (!seq_num_allocator_) |
| return 0; |
| return seq_num_allocator_->AllocateSequenceNumber(); |
| } |
| |
| private: |
| rtc::CriticalSection crit_; |
| rtc::ThreadChecker thread_checker_; |
| rtc::ThreadChecker pacer_thread_; |
| TransportSequenceNumberAllocator* seq_num_allocator_ RTC_GUARDED_BY(&crit_); |
| }; |
| |
| class RtpPacketSenderProxy : public RtpPacketSender { |
| public: |
| RtpPacketSenderProxy() : rtp_packet_sender_(nullptr) {} |
| |
| void SetPacketSender(RtpPacketSender* rtp_packet_sender) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| rtc::CritScope lock(&crit_); |
| rtp_packet_sender_ = rtp_packet_sender; |
| } |
| |
| // Implements RtpPacketSender. |
| void InsertPacket(Priority priority, |
| uint32_t ssrc, |
| uint16_t sequence_number, |
| int64_t capture_time_ms, |
| size_t bytes, |
| bool retransmission) override { |
| rtc::CritScope lock(&crit_); |
| if (rtp_packet_sender_) { |
| rtp_packet_sender_->InsertPacket(priority, ssrc, sequence_number, |
| capture_time_ms, bytes, retransmission); |
| } |
| } |
| |
| void SetAccountForAudioPackets(bool account_for_audio) override { |
| RTC_NOTREACHED(); |
| } |
| |
| private: |
| rtc::ThreadChecker thread_checker_; |
| rtc::CriticalSection crit_; |
| RtpPacketSender* rtp_packet_sender_ RTC_GUARDED_BY(&crit_); |
| }; |
| |
| class VoERtcpObserver : public RtcpBandwidthObserver { |
| public: |
| explicit VoERtcpObserver(ChannelSend* owner) |
| : owner_(owner), bandwidth_observer_(nullptr) {} |
| ~VoERtcpObserver() override {} |
| |
| void SetBandwidthObserver(RtcpBandwidthObserver* bandwidth_observer) { |
| rtc::CritScope lock(&crit_); |
| bandwidth_observer_ = bandwidth_observer; |
| } |
| |
| void OnReceivedEstimatedBitrate(uint32_t bitrate) override { |
| rtc::CritScope lock(&crit_); |
| if (bandwidth_observer_) { |
| bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate); |
| } |
| } |
| |
| void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks, |
| int64_t rtt, |
| int64_t now_ms) override { |
| { |
| rtc::CritScope lock(&crit_); |
| 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_; |
| rtc::CriticalSection crit_; |
| RtcpBandwidthObserver* bandwidth_observer_ RTC_GUARDED_BY(crit_); |
| }; |
| |
| class ChannelSend::ProcessAndEncodeAudioTask : public rtc::QueuedTask { |
| public: |
| ProcessAndEncodeAudioTask(std::unique_ptr<AudioFrame> audio_frame, |
| ChannelSend* channel) |
| : audio_frame_(std::move(audio_frame)), channel_(channel) { |
| RTC_DCHECK(channel_); |
| } |
| |
| private: |
| bool Run() override { |
| RTC_DCHECK_RUN_ON(channel_->encoder_queue_); |
| channel_->ProcessAndEncodeAudioOnTaskQueue(audio_frame_.get()); |
| return true; |
| } |
| |
| std::unique_ptr<AudioFrame> audio_frame_; |
| ChannelSend* const channel_; |
| }; |
| |
| int32_t ChannelSend::SendData(FrameType frameType, |
| uint8_t payloadType, |
| uint32_t timeStamp, |
| const uint8_t* payloadData, |
| size_t payloadSize, |
| const RTPFragmentationHeader* fragmentation) { |
| RTC_DCHECK_RUN_ON(encoder_queue_); |
| rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize); |
| |
| if (media_transport() != nullptr) { |
| if (frameType == kEmptyFrame) { |
| // TODO(bugs.webrtc.org/9719): Media transport Send doesn't support |
| // sending empty frames. |
| return 0; |
| } |
| |
| return SendMediaTransportAudio(frameType, payloadType, timeStamp, payload, |
| fragmentation); |
| } else { |
| return SendRtpAudio(frameType, payloadType, timeStamp, payload, |
| fragmentation); |
| } |
| } |
| |
| int32_t ChannelSend::SendRtpAudio(FrameType frameType, |
| uint8_t payloadType, |
| uint32_t timeStamp, |
| rtc::ArrayView<const uint8_t> payload, |
| const RTPFragmentationHeader* fragmentation) { |
| RTC_DCHECK_RUN_ON(encoder_queue_); |
| if (_includeAudioLevelIndication) { |
| // Store current audio level in the RTP/RTCP module. |
| // The level will be used in combination with voice-activity state |
| // (frameType) to add an RTP header extension |
| _rtpRtcpModule->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; |
| 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, _rtpRtcpModule->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. |
| // This call will trigger Transport::SendPacket() from the RTP/RTCP module. |
| if (!_rtpRtcpModule->SendOutgoingData((FrameType&)frameType, payloadType, |
| timeStamp, |
| // Leaving the time when this frame was |
| // received from the capture device as |
| // undefined for voice for now. |
| -1, payload.data(), payload.size(), |
| fragmentation, nullptr, nullptr)) { |
| RTC_DLOG(LS_ERROR) |
| << "ChannelSend::SendData() failed to send data to RTP/RTCP module"; |
| return -1; |
| } |
| |
| return 0; |
| } |
| |
| int32_t ChannelSend::SendMediaTransportAudio( |
| FrameType frameType, |
| uint8_t payloadType, |
| uint32_t timeStamp, |
| rtc::ArrayView<const uint8_t> payload, |
| const RTPFragmentationHeader* fragmentation) { |
| RTC_DCHECK_RUN_ON(encoder_queue_); |
| // TODO(nisse): Use null _transportPtr for MediaTransport. |
| // RTC_DCHECK(_transportPtr == nullptr); |
| uint64_t channel_id; |
| int sampling_rate_hz; |
| { |
| rtc::CritScope cs(&media_transport_lock_); |
| if (media_transport_payload_type_ != payloadType) { |
| // Payload type is being changed, media_transport_sampling_frequency_, |
| // no longer current. |
| return -1; |
| } |
| sampling_rate_hz = media_transport_sampling_frequency_; |
| channel_id = media_transport_channel_id_; |
| } |
| MediaTransportEncodedAudioFrame frame( |
| /*sampling_rate_hz=*/sampling_rate_hz, |
| |
| // TODO(nisse): Timestamp and sample index are the same for all supported |
| // audio codecs except G722. Refactor audio coding module to only use |
| // sample index, and leave translation to RTP time, when needed, for |
| // RTP-specific code. |
| /*starting_sample_index=*/timeStamp, |
| |
| // Sample count isn't conveniently available from the AudioCodingModule, |
| // and needs some refactoring to wire up in a good way. For now, left as |
| // zero. |
| /*sample_count=*/0, |
| |
| /*sequence_number=*/media_transport_sequence_number_, |
| MediaTransportFrameTypeForWebrtcFrameType(frameType), payloadType, |
| std::vector<uint8_t>(payload.begin(), payload.end())); |
| |
| // TODO(nisse): Introduce a MediaTransportSender object bound to a specific |
| // channel id. |
| RTCError rtc_error = |
| media_transport()->SendAudioFrame(channel_id, std::move(frame)); |
| |
| if (!rtc_error.ok()) { |
| RTC_LOG(LS_ERROR) << "Failed to send frame, rtc_error=" |
| << ToString(rtc_error.type()) << ", " |
| << rtc_error.message(); |
| return -1; |
| } |
| |
| ++media_transport_sequence_number_; |
| |
| return 0; |
| } |
| |
| ChannelSend::ChannelSend(Clock* clock, |
| rtc::TaskQueue* encoder_queue, |
| ProcessThread* module_process_thread, |
| MediaTransportInterface* media_transport, |
| OverheadObserver* overhead_observer, |
| 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) |
| : event_log_(rtc_event_log), |
| _timeStamp(0), // This is just an offset, RTP module will add it's own |
| // random offset |
| _moduleProcessThreadPtr(module_process_thread), |
| input_mute_(false), |
| previous_frame_muted_(false), |
| _includeAudioLevelIndication(false), |
| rtcp_observer_(new VoERtcpObserver(this)), |
| feedback_observer_proxy_(new TransportFeedbackProxy()), |
| seq_num_allocator_proxy_(new TransportSequenceNumberProxy()), |
| rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), |
| retransmission_rate_limiter_( |
| new RateLimiter(clock, kMaxRetransmissionWindowMs)), |
| use_twcc_plr_for_ana_( |
| webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled"), |
| encoder_queue_(encoder_queue), |
| media_transport_(media_transport), |
| frame_encryptor_(frame_encryptor), |
| crypto_options_(crypto_options) { |
| RTC_DCHECK(module_process_thread); |
| RTC_DCHECK(encoder_queue); |
| module_process_thread_checker_.DetachFromThread(); |
| |
| audio_coding_.reset(AudioCodingModule::Create(AudioCodingModule::Config())); |
| |
| RtpRtcp::Configuration configuration; |
| |
| // We gradually remove codepaths that depend on RTP when using media |
| // transport. All of this logic should be moved to the future |
| // RTPMediaTransport. In this case it means that overhead and bandwidth |
| // observers should not be called when using media transport. |
| if (!media_transport_) { |
| // TODO(sukhanov): Overhead observer is only needed for RTP path, because in |
| // media transport audio overhead is currently considered constant (see |
| // getter MediaTransportInterface::GetAudioPacketOverhead). In the future |
| // when we introduce RTP media transport we should make audio overhead |
| // interface consistent and work for both RTP and non-RTP implementations. |
| configuration.overhead_observer = overhead_observer; |
| configuration.bandwidth_callback = rtcp_observer_.get(); |
| configuration.transport_feedback_callback = feedback_observer_proxy_.get(); |
| } |
| |
| configuration.clock = clock; |
| configuration.audio = true; |
| configuration.outgoing_transport = rtp_transport; |
| |
| configuration.paced_sender = rtp_packet_sender_proxy_.get(); |
| configuration.transport_sequence_number_allocator = |
| seq_num_allocator_proxy_.get(); |
| |
| configuration.event_log = event_log_; |
| configuration.rtt_stats = rtcp_rtt_stats; |
| configuration.retransmission_rate_limiter = |
| retransmission_rate_limiter_.get(); |
| configuration.extmap_allow_mixed = extmap_allow_mixed; |
| configuration.rtcp_report_interval_ms = rtcp_report_interval_ms; |
| |
| _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration)); |
| _rtpRtcpModule->SetSendingMediaStatus(false); |
| |
| // We want to invoke the 'TargetRateObserver' and |OnOverheadChanged| |
| // callbacks after the audio_coding_ is fully initialized. |
| if (media_transport_) { |
| RTC_DLOG(LS_INFO) << "Setting media_transport_ rate observers."; |
| media_transport_->AddTargetTransferRateObserver(this); |
| } else { |
| RTC_DLOG(LS_INFO) << "Not setting media_transport_ rate observers."; |
| } |
| |
| _moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE); |
| |
| // Ensure that RTCP is enabled by default for the created channel. |
| // Note that, the module will keep generating RTCP until it is explicitly |
| // disabled by the user. |
| // After StopListen (when no sockets exists), RTCP packets will no longer |
| // be transmitted since the Transport object will then be invalid. |
| // RTCP is enabled by default. |
| _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound); |
| |
| int error = audio_coding_->RegisterTransportCallback(this); |
| RTC_DCHECK_EQ(0, error); |
| } |
| |
| ChannelSend::~ChannelSend() { |
| RTC_DCHECK(construction_thread_.CalledOnValidThread()); |
| |
| if (media_transport_) { |
| media_transport_->RemoveTargetTransferRateObserver(this); |
| } |
| |
| StopSend(); |
| int error = audio_coding_->RegisterTransportCallback(NULL); |
| RTC_DCHECK_EQ(0, error); |
| |
| if (_moduleProcessThreadPtr) |
| _moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get()); |
| } |
| |
| void ChannelSend::StartSend() { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| RTC_DCHECK(!sending_); |
| sending_ = true; |
| |
| _rtpRtcpModule->SetSendingMediaStatus(true); |
| int ret = _rtpRtcpModule->SetSendingStatus(true); |
| RTC_DCHECK_EQ(0, ret); |
| { |
| // It is now OK to start posting tasks to the encoder task queue. |
| rtc::CritScope cs(&encoder_queue_lock_); |
| encoder_queue_is_active_ = true; |
| } |
| } |
| |
| void ChannelSend::StopSend() { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| if (!sending_) { |
| return; |
| } |
| sending_ = false; |
| |
| // Post a task to the encoder thread which sets an event when the task is |
| // executed. We know that no more encoding tasks will be added to the task |
| // queue for this channel since sending is now deactivated. It means that, |
| // if we wait for the event to bet set, we know that no more pending tasks |
| // exists and it is therfore guaranteed that the task queue will never try |
| // to acccess and invalid channel object. |
| RTC_DCHECK(encoder_queue_); |
| |
| rtc::Event flush; |
| { |
| // Clear |encoder_queue_is_active_| under lock to prevent any other tasks |
| // than this final "flush task" to be posted on the queue. |
| rtc::CritScope cs(&encoder_queue_lock_); |
| encoder_queue_is_active_ = false; |
| encoder_queue_->PostTask([&flush]() { flush.Set(); }); |
| } |
| flush.Wait(rtc::Event::kForever); |
| |
| // Reset sending SSRC and sequence number and triggers direct transmission |
| // of RTCP BYE |
| if (_rtpRtcpModule->SetSendingStatus(false) == -1) { |
| RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending"; |
| } |
| _rtpRtcpModule->SetSendingMediaStatus(false); |
| } |
| |
| bool 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. |
| _rtpRtcpModule->RegisterAudioSendPayload(payload_type, |
| "audio", |
| encoder->RtpTimestampRateHz(), |
| encoder->NumChannels(), |
| 0); |
| |
| if (media_transport_) { |
| rtc::CritScope cs(&media_transport_lock_); |
| media_transport_payload_type_ = payload_type; |
| // TODO(nisse): Currently broken for G722, since timestamps passed through |
| // encoder use RTP clock rather than sample count, and they differ for G722. |
| media_transport_sampling_frequency_ = encoder->RtpTimestampRateHz(); |
| } |
| audio_coding_->SetEncoder(std::move(encoder)); |
| return true; |
| } |
| |
| 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_.CalledOnValidThread() || |
| // module_process_thread_checker_.CalledOnValidThread()); |
| rtc::CritScope lock(&bitrate_crit_section_); |
| |
| CallEncoder([&](AudioEncoder* encoder) { |
| encoder->OnReceivedUplinkAllocation(update); |
| }); |
| retransmission_rate_limiter_->SetMaxRate(update.target_bitrate.bps()); |
| configured_bitrate_bps_ = update.target_bitrate.bps(); |
| } |
| |
| int ChannelSend::GetBitrate() const { |
| rtc::CritScope lock(&bitrate_crit_section_); |
| return configured_bitrate_bps_; |
| } |
| |
| void ChannelSend::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| if (!use_twcc_plr_for_ana_) |
| return; |
| CallEncoder([&](AudioEncoder* encoder) { |
| encoder->OnReceivedUplinkPacketLossFraction(packet_loss_rate); |
| }); |
| } |
| |
| void ChannelSend::OnRecoverableUplinkPacketLossRate( |
| float recoverable_packet_loss_rate) { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| CallEncoder([&](AudioEncoder* encoder) { |
| encoder->OnReceivedUplinkRecoverablePacketLossFraction( |
| recoverable_packet_loss_rate); |
| }); |
| } |
| |
| void ChannelSend::OnUplinkPacketLossRate(float packet_loss_rate) { |
| if (use_twcc_plr_for_ana_) |
| return; |
| CallEncoder([&](AudioEncoder* encoder) { |
| encoder->OnReceivedUplinkPacketLossFraction(packet_loss_rate); |
| }); |
| } |
| |
| // TODO(nisse): Delete always-true return value. |
| bool ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) { |
| // May be called on either worker thread or network thread. |
| if (media_transport_) { |
| // Ignore RTCP packets while media transport is used. |
| // Those packets should not arrive, but we are seeing occasional packets. |
| return 0; |
| } |
| |
| // Deliver RTCP packet to RTP/RTCP module for parsing |
| _rtpRtcpModule->IncomingRtcpPacket(data, length); |
| |
| int64_t rtt = GetRTT(); |
| if (rtt == 0) { |
| // Waiting for valid RTT. |
| return true; |
| } |
| |
| 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); |
| return true; |
| } |
| |
| void ChannelSend::SetInputMute(bool enable) { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| rtc::CritScope cs(&volume_settings_critsect_); |
| input_mute_ = enable; |
| } |
| |
| bool ChannelSend::InputMute() const { |
| rtc::CritScope cs(&volume_settings_critsect_); |
| 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 (_rtpRtcpModule->SendTelephoneEventOutband( |
| event, duration_ms, kTelephoneEventAttenuationdB) != 0) { |
| RTC_DLOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event"; |
| return false; |
| } |
| return true; |
| } |
| |
| bool 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); |
| _rtpRtcpModule->RegisterAudioSendPayload(payload_type, "telephone-event", |
| payload_frequency, 0, 0); |
| return true; |
| } |
| |
| void ChannelSend::SetLocalSSRC(uint32_t ssrc) { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| RTC_DCHECK(!sending_); |
| |
| if (media_transport_) { |
| rtc::CritScope cs(&media_transport_lock_); |
| media_transport_channel_id_ = ssrc; |
| } |
| _rtpRtcpModule->SetSSRC(ssrc); |
| } |
| |
| void ChannelSend::SetRid(const std::string& rid, |
| int extension_id, |
| int repaired_extension_id) { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| if (extension_id != 0) { |
| int ret = SetSendRtpHeaderExtension(!rid.empty(), kRtpExtensionRtpStreamId, |
| extension_id); |
| RTC_DCHECK_EQ(0, ret); |
| } |
| if (repaired_extension_id != 0) { |
| int ret = SetSendRtpHeaderExtension(!rid.empty(), kRtpExtensionRtpStreamId, |
| repaired_extension_id); |
| RTC_DCHECK_EQ(0, ret); |
| } |
| _rtpRtcpModule->SetRid(rid); |
| } |
| |
| void ChannelSend::SetMid(const std::string& mid, int extension_id) { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| int ret = SetSendRtpHeaderExtension(true, kRtpExtensionMid, extension_id); |
| RTC_DCHECK_EQ(0, ret); |
| _rtpRtcpModule->SetMid(mid); |
| } |
| |
| void ChannelSend::SetExtmapAllowMixed(bool extmap_allow_mixed) { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| _rtpRtcpModule->SetExtmapAllowMixed(extmap_allow_mixed); |
| } |
| |
| void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| _includeAudioLevelIndication = enable; |
| int ret = SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id); |
| RTC_DCHECK_EQ(0, ret); |
| } |
| |
| void ChannelSend::EnableSendTransportSequenceNumber(int id) { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| int ret = |
| SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id); |
| RTC_DCHECK_EQ(0, ret); |
| } |
| |
| void ChannelSend::RegisterSenderCongestionControlObjects( |
| RtpTransportControllerSendInterface* transport, |
| RtcpBandwidthObserver* bandwidth_observer) { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| RtpPacketSender* rtp_packet_sender = transport->packet_sender(); |
| TransportFeedbackObserver* transport_feedback_observer = |
| transport->transport_feedback_observer(); |
| PacketRouter* packet_router = transport->packet_router(); |
| |
| RTC_DCHECK(rtp_packet_sender); |
| RTC_DCHECK(transport_feedback_observer); |
| RTC_DCHECK(packet_router); |
| RTC_DCHECK(!packet_router_); |
| rtcp_observer_->SetBandwidthObserver(bandwidth_observer); |
| feedback_observer_proxy_->SetTransportFeedbackObserver( |
| transport_feedback_observer); |
| seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router); |
| rtp_packet_sender_proxy_->SetPacketSender(rtp_packet_sender); |
| _rtpRtcpModule->SetStorePacketsStatus(true, 600); |
| constexpr bool remb_candidate = false; |
| packet_router->AddSendRtpModule(_rtpRtcpModule.get(), remb_candidate); |
| packet_router_ = packet_router; |
| } |
| |
| void ChannelSend::ResetSenderCongestionControlObjects() { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| RTC_DCHECK(packet_router_); |
| _rtpRtcpModule->SetStorePacketsStatus(false, 600); |
| rtcp_observer_->SetBandwidthObserver(nullptr); |
| feedback_observer_proxy_->SetTransportFeedbackObserver(nullptr); |
| seq_num_allocator_proxy_->SetSequenceNumberAllocator(nullptr); |
| packet_router_->RemoveSendRtpModule(_rtpRtcpModule.get()); |
| packet_router_ = nullptr; |
| rtp_packet_sender_proxy_->SetPacketSender(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 = _rtpRtcpModule->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<RTCPReportBlock> rtcp_report_blocks; |
| |
| int ret = _rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks); |
| RTC_DCHECK_EQ(0, ret); |
| |
| std::vector<ReportBlock> report_blocks; |
| |
| std::vector<RTCPReportBlock>::const_iterator it = rtcp_report_blocks.begin(); |
| for (; it != rtcp_report_blocks.end(); ++it) { |
| ReportBlock report_block; |
| report_block.sender_SSRC = it->sender_ssrc; |
| report_block.source_SSRC = it->source_ssrc; |
| report_block.fraction_lost = it->fraction_lost; |
| report_block.cumulative_num_packets_lost = it->packets_lost; |
| report_block.extended_highest_sequence_number = |
| it->extended_highest_sequence_number; |
| report_block.interarrival_jitter = it->jitter; |
| report_block.last_SR_timestamp = it->last_sender_report_timestamp; |
| report_block.delay_since_last_SR = it->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(); |
| |
| size_t bytesSent(0); |
| uint32_t packetsSent(0); |
| |
| if (_rtpRtcpModule->DataCountersRTP(&bytesSent, &packetsSent) != 0) { |
| RTC_DLOG(LS_WARNING) |
| << "GetRTPStatistics() failed to retrieve RTP datacounters" |
| << " => output will not be complete"; |
| } |
| |
| stats.bytesSent = bytesSent; |
| stats.packetsSent = packetsSent; |
| |
| return stats; |
| } |
| |
| void ChannelSend::ProcessAndEncodeAudio( |
| std::unique_ptr<AudioFrame> audio_frame) { |
| RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_); |
| // Avoid posting any new tasks if sending was already stopped in StopSend(). |
| rtc::CritScope cs(&encoder_queue_lock_); |
| if (!encoder_queue_is_active_) { |
| 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(std::unique_ptr<rtc::QueuedTask>( |
| new ProcessAndEncodeAudioTask(std::move(audio_frame), this))); |
| } |
| |
| void ChannelSend::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) { |
| RTC_DCHECK_RUN_ON(encoder_queue_); |
| RTC_DCHECK_GT(audio_input->samples_per_channel_, 0); |
| RTC_DCHECK_LE(audio_input->num_channels_, 2); |
| |
| // 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_input->ElapsedProfileTimeMs()); |
| |
| bool is_muted = InputMute(); |
| AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted); |
| |
| if (_includeAudioLevelIndication) { |
| size_t length = |
| audio_input->samples_per_channel_ * audio_input->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_input->data(), length)); |
| } |
| } |
| previous_frame_muted_ = is_muted; |
| |
| // Add 10ms of raw (PCM) audio data to the encoder @ 32kHz. |
| |
| // The ACM resamples internally. |
| audio_input->timestamp_ = _timeStamp; |
| // 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_input) < 0) { |
| RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed."; |
| return; |
| } |
| |
| _timeStamp += static_cast<uint32_t>(audio_input->samples_per_channel_); |
| } |
| |
| ANAStats ChannelSend::GetANAStatistics() const { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| return audio_coding_->GetANAStats(); |
| } |
| |
| RtpRtcp* ChannelSend::GetRtpRtcp() const { |
| RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread()); |
| return _rtpRtcpModule.get(); |
| } |
| |
| int ChannelSend::SetSendRtpHeaderExtension(bool enable, |
| RTPExtensionType type, |
| int id) { |
| int error = 0; |
| _rtpRtcpModule->DeregisterSendRtpHeaderExtension(type); |
| if (enable) { |
| // TODO(nisse): RtpRtcp::RegisterSendRtpHeaderExtension to take an int |
| // argument. Currently it wants an uint8_t. |
| error = _rtpRtcpModule->RegisterSendRtpHeaderExtension( |
| type, rtc::dchecked_cast<uint8_t>(id)); |
| } |
| return error; |
| } |
| |
| int64_t ChannelSend::GetRTT() const { |
| if (media_transport_) { |
| // GetRTT is generally used in the RTCP codepath, where media transport is |
| // not present and so it shouldn't be needed. But it's also invoked in |
| // 'GetStats' method, and for now returning media transport RTT here gives |
| // us "free" rtt stats for media transport. |
| auto target_rate = media_transport_->GetLatestTargetTransferRate(); |
| if (target_rate.has_value()) { |
| return target_rate.value().network_estimate.round_trip_time.ms(); |
| } |
| |
| return 0; |
| } |
| RtcpMode method = _rtpRtcpModule->RTCP(); |
| if (method == RtcpMode::kOff) { |
| return 0; |
| } |
| std::vector<RTCPReportBlock> report_blocks; |
| _rtpRtcpModule->RemoteRTCPStat(&report_blocks); |
| |
| if (report_blocks.empty()) { |
| return 0; |
| } |
| |
| int64_t rtt = 0; |
| int64_t avg_rtt = 0; |
| int64_t max_rtt = 0; |
| int64_t min_rtt = 0; |
| // We don't know in advance the remote ssrc used by the other end's receiver |
| // reports, so use the SSRC of the first report block for calculating the RTT. |
| if (_rtpRtcpModule->RTT(report_blocks[0].sender_ssrc, &rtt, &avg_rtt, |
| &min_rtt, &max_rtt) != 0) { |
| return 0; |
| } |
| return rtt; |
| } |
| |
| void ChannelSend::SetFrameEncryptor( |
| rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) { |
| RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| rtc::CritScope cs(&encoder_queue_lock_); |
| if (encoder_queue_is_active_) { |
| encoder_queue_->PostTask([this, frame_encryptor]() mutable { |
| this->frame_encryptor_ = std::move(frame_encryptor); |
| }); |
| } else { |
| frame_encryptor_ = std::move(frame_encryptor); |
| } |
| } |
| |
| // TODO(sukhanov): Consider moving TargetTransferRate observer to |
| // AudioSendStream. Since AudioSendStream owns encoder and configures ANA, it |
| // makes sense to consolidate all rate (and overhead) calculation there. |
| void ChannelSend::OnTargetTransferRate(TargetTransferRate rate) { |
| RTC_DCHECK(media_transport_); |
| OnReceivedRtt(rate.network_estimate.round_trip_time.ms()); |
| } |
| |
| void ChannelSend::OnReceivedRtt(int64_t rtt_ms) { |
| // Invoke audio encoders OnReceivedRtt(). |
| CallEncoder( |
| [rtt_ms](AudioEncoder* encoder) { encoder->OnReceivedRtt(rtt_ms); }); |
| } |
| |
| } // namespace |
| |
| std::unique_ptr<ChannelSendInterface> CreateChannelSend( |
| Clock* clock, |
| rtc::TaskQueue* encoder_queue, |
| ProcessThread* module_process_thread, |
| MediaTransportInterface* media_transport, |
| OverheadObserver* overhead_observer, |
| 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) { |
| return absl::make_unique<ChannelSend>( |
| clock, encoder_queue, module_process_thread, media_transport, |
| overhead_observer, rtp_transport, rtcp_rtt_stats, rtc_event_log, |
| frame_encryptor, crypto_options, extmap_allow_mixed, |
| rtcp_report_interval_ms); |
| } |
| |
| } // namespace voe |
| } // namespace webrtc |