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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/audio/channel_send.cc
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/channel_send.cc')
-rw-r--r--third_party/libwebrtc/audio/channel_send.cc983
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
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+++ b/third_party/libwebrtc/audio/channel_send.cc
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+/*
+ * 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