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Nagram/TMessagesProj/jni/voip/webrtc/audio/channel_send.cc
DrKLO e8d88e56e4 Update to 7.8.0 (2359)
2021-06-25 03:43:10 +03:00

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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 "modules/utility/include/process_thread.h"
#include "rtc_base/checks.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/synchronization/mutex.h"
#include "rtc_base/task_queue.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;
class RtpPacketSenderProxy;
class TransportSequenceNumberProxy;
class VoERtcpObserver;
class ChannelSend : public ChannelSendInterface,
public AudioPacketizationCallback { // receive encoded
// packets from the ACM
public:
// TODO(nisse): Make OnUplinkPacketLossRate public, and delete friend
// declaration.
friend class VoERtcpObserver;
ChannelSend(Clock* clock,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
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);
~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 GetBitrate() 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;
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;
void OnUplinkPacketLossRate(float packet_loss_rate);
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_;
SequenceChecker 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_;
mutable Mutex volume_settings_mutex_;
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_;
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_mutex_);
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;
TransportFeedbackObserver* const feedback_observer_;
const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_;
const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
SequenceChecker construction_thread_;
bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_) = 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 bitrate_mutex_;
int configured_bitrate_bps_ RTC_GUARDED_BY(bitrate_mutex_) = 0;
// Defined last to ensure that there are no running tasks when the other
// members are destroyed.
rtc::TaskQueue encoder_queue_;
const bool fixing_timestamp_stall_;
};
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));
}
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 (_includeAudioLevelIndication) {
// 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,
ProcessThread* module_process_thread,
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)
: 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_(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)),
fixing_timestamp_stall_(
!field_trial::IsDisabled("WebRTC-Audio-FixTimestampStall")) {
RTC_DCHECK(module_process_thread);
module_process_thread_checker_.Detach();
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 ? 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;
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.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());
_moduleProcessThreadPtr->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
// 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);
if (_moduleProcessThreadPtr)
_moduleProcessThreadPtr->DeRegisterModule(rtp_rtcp_.get());
}
void ChannelSend::StartSend() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK(!sending_);
sending_ = true;
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_.PostTask([this] {
RTC_DCHECK_RUN_ON(&encoder_queue_);
encoder_queue_is_active_ = true;
});
}
void ChannelSend::StopSend() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (!sending_) {
return;
}
sending_ = false;
rtc::Event flush;
encoder_queue_.PostTask([this, &flush]() {
RTC_DCHECK_RUN_ON(&encoder_queue_);
encoder_queue_is_active_ = false;
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);
}
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());
MutexLock lock(&bitrate_mutex_);
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 {
MutexLock lock(&bitrate_mutex_);
return configured_bitrate_bps_;
}
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_);
_includeAudioLevelIndication = enable;
if (enable) {
rtp_rtcp_->RegisterRtpHeaderExtension(AudioLevel::kUri, id);
} else {
rtp_rtcp_->DeregisterSendRtpHeaderExtension(AudioLevel::kUri);
}
}
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);
constexpr bool remb_candidate = false;
packet_router->AddSendRtpModule(rtp_rtcp_.get(), remb_candidate);
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_->RemoveSendRtpModule(rtp_rtcp_.get());
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();
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.retransmitted_packets_sent = rtp_stats.retransmitted.packets;
stats.report_block_datas = rtp_rtcp_->GetLatestReportBlockData();
return stats;
}
void ChannelSend::ProcessAndEncodeAudio(
std::unique_ptr<AudioFrame> audio_frame) {
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);
// 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_) {
if (fixing_timestamp_stall_) {
_timeStamp +=
static_cast<uint32_t>(audio_frame->samples_per_channel_);
}
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 (_includeAudioLevelIndication) {
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;
// Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.
// The ACM resamples internally.
audio_frame->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_frame) < 0) {
RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed.";
return;
}
_timeStamp += static_cast<uint32_t>(audio_frame->samples_per_channel_);
});
}
ANAStats ChannelSend::GetANAStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return audio_coding_->GetANAStats();
}
RtpRtcpInterface* ChannelSend::GetRtpRtcp() const {
RTC_DCHECK(module_process_thread_checker_.IsCurrent());
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,
ProcessThread* module_process_thread,
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) {
return std::make_unique<ChannelSend>(
clock, task_queue_factory, module_process_thread, 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);
}
} // namespace voe
} // namespace webrtc
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