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webrtc / src / 33678af88db375287b565804be3751f377645aa2 / . / audio / channel_send.cc
blob: 569615bad6f7739e585962061493ec23fb3df446 [file] [log] [blame]
/*
* 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 "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/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/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::AudioFrameType frame_type) {
switch (frame_type) {
case AudioFrameType::kAudioFrameSpeech:
return MediaTransportEncodedAudioFrame::FrameType::kSpeech;
break;
case AudioFrameType::kAudioFrameCN:
return MediaTransportEncodedAudioFrame::FrameType::
kDiscontinuousTransmission;
break;
default:
RTC_CHECK(false) << "Unexpected frame type="
<< static_cast<int>(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,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
const MediaTransportConfig& media_transport_config,
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,
uint32_t ssrc);
~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.
RtpRtcp* 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 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:
// From AudioPacketizationCallback in the ACM
int32_t SendData(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
size_t payloadSize) override;
void OnUplinkPacketLossRate(float packet_loss_rate);
bool InputMute() const;
void SetSendRtpHeaderExtension(bool enable, absl::string_view uri, int id);
int32_t SendRtpAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
rtc::ArrayView<const uint8_t> payload)
RTC_RUN_ON(encoder_queue_);
int32_t SendMediaTransportAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
rtc::ArrayView<const uint8_t> payload)
RTC_RUN_ON(encoder_queue_);
// Return media transport or nullptr if using RTP.
MediaTransportInterface* media_transport() {
return media_transport_config_.media_transport;
}
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<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_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<RtpPacketSenderProxy> rtp_packet_pacer_proxy_;
const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
rtc::ThreadChecker construction_thread_;
const bool use_twcc_plr_for_ana_;
bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_) = false;
MediaTransportConfig media_transport_config_;
int media_transport_sequence_number_ RTC_GUARDED_BY(encoder_queue_) = 0;
rtc::CriticalSection media_transport_lock_;
// Currently set to local SSRC at construction.
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_
RTC_GUARDED_BY(encoder_queue_);
// 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;
// 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 TransportFeedbackProxy : public TransportFeedbackObserver {
public:
TransportFeedbackProxy() : feedback_observer_(nullptr) {
pacer_thread_.Detach();
network_thread_.Detach();
}
void SetTransportFeedbackObserver(
TransportFeedbackObserver* feedback_observer) {
RTC_DCHECK(thread_checker_.IsCurrent());
rtc::CritScope lock(&crit_);
feedback_observer_ = feedback_observer;
}
// Implements TransportFeedbackObserver.
void OnAddPacket(const RtpPacketSendInfo& packet_info) override {
RTC_DCHECK(pacer_thread_.IsCurrent());
rtc::CritScope lock(&crit_);
if (feedback_observer_)
feedback_observer_->OnAddPacket(packet_info);
}
void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override {
RTC_DCHECK(network_thread_.IsCurrent());
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 RtpPacketSenderProxy : public RtpPacketSender {
public:
RtpPacketSenderProxy() : rtp_packet_pacer_(nullptr) {}
void SetPacketPacer(RtpPacketSender* rtp_packet_pacer) {
RTC_DCHECK(thread_checker_.IsCurrent());
rtc::CritScope lock(&crit_);
rtp_packet_pacer_ = rtp_packet_pacer;
}
void EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) override {
rtc::CritScope lock(&crit_);
rtp_packet_pacer_->EnqueuePackets(std::move(packets));
}
private:
rtc::ThreadChecker thread_checker_;
rtc::CriticalSection crit_;
RtpPacketSender* rtp_packet_pacer_ 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_);
};
int32_t ChannelSend::SendData(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
size_t payloadSize) {
RTC_DCHECK_RUN_ON(&encoder_queue_);
rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
if (media_transport() != nullptr) {
if (frameType == AudioFrameType::kEmptyFrame) {
// TODO(bugs.webrtc.org/9719): Media transport Send doesn't support
// sending empty frames.
return 0;
}
return SendMediaTransportAudio(frameType, payloadType, timeStamp, payload);
} else {
return SendRtpAudio(frameType, payloadType, timeStamp, payload);
}
}
int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
rtc::ArrayView<const uint8_t> payload) {
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, _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.
if (!_rtpRtcpModule->OnSendingRtpFrame(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 false;
}
// 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.
const uint32_t rtp_timestamp = timeStamp + _rtpRtcpModule->StartTimestamp();
// This call will trigger Transport::SendPacket() from the RTP/RTCP module.
if (!rtp_sender_audio_->SendAudio(frameType, payloadType, rtp_timestamp,
payload.data(), payload.size())) {
RTC_DLOG(LS_ERROR)
<< "ChannelSend::SendData() failed to send data to RTP/RTCP module";
return -1;
}
return 0;
}
int32_t ChannelSend::SendMediaTransportAudio(
AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
rtc::ArrayView<const uint8_t> payload) {
// 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.
/*samples_per_channel=*/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,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
const MediaTransportConfig& media_transport_config,
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,
uint32_t ssrc)
: 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()),
rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
retransmission_rate_limiter_(
new RateLimiter(clock, kMaxRetransmissionWindowMs)),
use_twcc_plr_for_ana_(
webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled"),
media_transport_config_(media_transport_config),
frame_encryptor_(frame_encryptor),
crypto_options_(crypto_options),
encoder_queue_(task_queue_factory->CreateTaskQueue(
"AudioEncoder",
TaskQueueFactory::Priority::NORMAL)) {
RTC_DCHECK(module_process_thread);
module_process_thread_checker_.Detach();
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_config.media_transport) {
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.clock = Clock::GetRealTimeClock();
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;
if (media_transport_config_.media_transport) {
rtc::CritScope cs(&media_transport_lock_);
media_transport_channel_id_ = ssrc;
}
_rtpRtcpModule = RtpRtcp::Create(configuration);
_rtpRtcpModule->SetSendingMediaStatus(false);
rtp_sender_audio_ = std::make_unique<RTPSenderAudio>(
configuration.clock, _rtpRtcpModule->RtpSender());
// We want to invoke the 'TargetRateObserver' and |OnOverheadChanged|
// callbacks after the audio_coding_ is fully initialized.
if (media_transport_config.media_transport) {
RTC_DLOG(LS_INFO) << "Setting media_transport_ rate observers.";
media_transport_config.media_transport->AddTargetTransferRateObserver(this);
media_transport_config.media_transport->SetAudioOverheadObserver(
overhead_observer);
} 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.
_rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
int error = audio_coding_->RegisterTransportCallback(this);
RTC_DCHECK_EQ(0, error);
}
ChannelSend::~ChannelSend() {
RTC_DCHECK(construction_thread_.IsCurrent());
if (media_transport_config_.media_transport) {
media_transport_config_.media_transport->RemoveTargetTransferRateObserver(
this);
media_transport_config_.media_transport->SetAudioOverheadObserver(nullptr);
}
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 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 (_rtpRtcpModule->SetSendingStatus(false) == -1) {
RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending";
}
_rtpRtcpModule->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.
_rtpRtcpModule->RegisterSendPayloadFrequency(payload_type,
encoder->RtpTimestampRateHz());
rtp_sender_audio_->RegisterAudioPayload("audio", payload_type,
encoder->RtpTimestampRateHz(),
encoder->NumChannels(), 0);
if (media_transport_config_.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));
}
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());
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);
});
}
void ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
// May be called on either worker thread or network thread.
if (media_transport_config_.media_transport) {
// Ignore RTCP packets while media transport is used.
// Those packets should not arrive, but we are seeing occasional packets.
return;
}
// 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;
}
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_);
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 (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) {
_rtpRtcpModule->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);
_rtpRtcpModule->RegisterSendPayloadFrequency(payload_type, payload_frequency);
rtp_sender_audio_->RegisterAudioPayload("telephone-event", payload_type,
payload_frequency, 0, 0);
}
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) {
SetSendRtpHeaderExtension(!rid.empty(), RtpStreamId::kUri, extension_id);
}
if (repaired_extension_id != 0) {
SetSendRtpHeaderExtension(!rid.empty(), RtpStreamId::kUri,
repaired_extension_id);
}
_rtpRtcpModule->SetRid(rid);
}
void ChannelSend::SetMid(const std::string& mid, int extension_id) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
SetSendRtpHeaderExtension(true, RtpMid::kUri, extension_id);
_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;
SetSendRtpHeaderExtension(enable, AudioLevel::kUri, id);
}
void ChannelSend::EnableSendTransportSequenceNumber(int id) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
SetSendRtpHeaderExtension(true, TransportSequenceNumber::kUri, id);
}
void ChannelSend::RegisterSenderCongestionControlObjects(
RtpTransportControllerSendInterface* transport,
RtcpBandwidthObserver* bandwidth_observer) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RtpPacketSender* rtp_packet_pacer = transport->packet_sender();
TransportFeedbackObserver* transport_feedback_observer =
transport->transport_feedback_observer();
PacketRouter* packet_router = transport->packet_router();
RTC_DCHECK(rtp_packet_pacer);
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);
rtp_packet_pacer_proxy_->SetPacketPacer(rtp_packet_pacer);
_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);
packet_router_->RemoveSendRtpModule(_rtpRtcpModule.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 = _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();
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
_rtpRtcpModule->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 = _rtpRtcpModule->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_) {
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();
}
RtpRtcp* ChannelSend::GetRtpRtcp() const {
RTC_DCHECK(module_process_thread_checker_.IsCurrent());
return _rtpRtcpModule.get();
}
void ChannelSend::SetSendRtpHeaderExtension(bool enable,
absl::string_view uri,
int id) {
_rtpRtcpModule->DeregisterSendRtpHeaderExtension(uri);
if (enable) {
_rtpRtcpModule->RegisterRtpHeaderExtension(uri, id);
}
}
int64_t ChannelSend::GetRTT() const {
if (media_transport_config_.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_config_.media_transport->GetLatestTargetTransferRate();
if (target_rate.has_value()) {
return target_rate.value().network_estimate.round_trip_time.ms();
}
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_);
encoder_queue_.PostTask([this, frame_encryptor]() mutable {
RTC_DCHECK_RUN_ON(&encoder_queue_);
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_config_.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,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
const MediaTransportConfig& media_transport_config,
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,
uint32_t ssrc) {
return std::make_unique<ChannelSend>(
clock, task_queue_factory, module_process_thread, media_transport_config,
overhead_observer, rtp_transport, rtcp_rtt_stats, rtc_event_log,
frame_encryptor, crypto_options, extmap_allow_mixed,
rtcp_report_interval_ms, ssrc);
}
} // namespace voe
} // namespace webrtc