blob: 756e622acb549a4a50561b4e42699fdb947e93fb [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 "api/sequence_checker.h"
#include "audio/channel_send_frame_transformer_delegate.h"
#include "audio/utility/audio_frame_operations.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_processing/rms_level.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace voe {
namespace {
constexpr int64_t kMaxRetransmissionWindowMs = 1000;
constexpr int64_t kMinRetransmissionWindowMs = 30;
class RtpPacketSenderProxy;
class TransportSequenceNumberProxy;
class ChannelSend : public ChannelSendInterface,
public AudioPacketizationCallback, // receive encoded
// packets from the ACM
public RtcpPacketTypeCounterObserver,
public ReportBlockDataObserver {
public:
ChannelSend(Clock* clock,
TaskQueueFactory* task_queue_factory,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options,
bool extmap_allow_mixed,
int rtcp_report_interval_ms,
uint32_t ssrc,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
RtpTransportControllerSendInterface* transport_controller,
const FieldTrialsView& field_trials);
~ChannelSend() override;
// Send using this encoder, with this payload type.
void SetEncoder(int payload_type,
const SdpAudioFormat& encoder_format,
std::unique_ptr<AudioEncoder> encoder) override;
void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
modifier) override;
void CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) override;
// API methods
void StartSend() override;
void StopSend() override;
// Codecs
void OnBitrateAllocation(BitrateAllocationUpdate update) override;
int GetTargetBitrate() const override;
// Network
void ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
// Muting, Volume and Level.
void SetInputMute(bool enable) override;
// Stats.
ANAStats GetANAStatistics() const override;
// Used by AudioSendStream.
RtpRtcpInterface* GetRtpRtcp() const override;
void RegisterCngPayloadType(int payload_type, int payload_frequency) override;
// DTMF.
bool SendTelephoneEventOutband(int event, int duration_ms) override;
void SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) override;
// RTP+RTCP
void SetSendAudioLevelIndicationStatus(bool enable, int id) override;
void RegisterSenderCongestionControlObjects(
RtpTransportControllerSendInterface* transport) override;
void ResetSenderCongestionControlObjects() override;
void SetRTCP_CNAME(absl::string_view c_name) override;
std::vector<ReportBlockData> GetRemoteRTCPReportBlocks() const override;
CallSendStatistics GetRTCPStatistics() const override;
// ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
// which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
// the actual processing of the audio takes place. The processing mainly
// consists of encoding and preparing the result for sending by adding it to a
// send queue.
// The main reason for using a task queue here is to release the native,
// OS-specific, audio capture thread as soon as possible to ensure that it
// can go back to sleep and be prepared to deliver an new captured audio
// packet.
void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) override;
int64_t GetRTT() const override;
// E2EE Custom Audio Frame Encryption
void SetFrameEncryptor(
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override;
// Sets a frame transformer between encoder and packetizer, to transform
// encoded frames before sending them out the network.
void SetEncoderToPacketizerFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override;
// RtcpPacketTypeCounterObserver.
void RtcpPacketTypesCounterUpdated(
uint32_t ssrc,
const RtcpPacketTypeCounter& packet_counter) override;
// ReportBlockDataObserver.
void OnReportBlockDataUpdated(ReportBlockData report_block) override;
private:
// From AudioPacketizationCallback in the ACM
int32_t SendData(AudioFrameType frameType,
uint8_t payloadType,
uint32_t rtp_timestamp,
const uint8_t* payloadData,
size_t payloadSize,
int64_t absolute_capture_timestamp_ms) override;
bool InputMute() const;
int32_t SendRtpAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t rtp_timestamp_without_offset,
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs,
absl::optional<uint8_t> audio_level_dbov)
RTC_RUN_ON(encoder_queue_checker_);
void OnReceivedRtt(int64_t rtt_ms);
void InitFrameTransformerDelegate(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer);
// Thread checkers document and lock usage of some methods on voe::Channel to
// specific threads we know about. The goal is to eventually split up
// voe::Channel into parts with single-threaded semantics, and thereby reduce
// the need for locks.
RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
// Methods accessed from audio and video threads are checked for sequential-
// only access. We don't necessarily own and control these threads, so thread
// checkers cannot be used. E.g. Chromium may transfer "ownership" from one
// audio thread to another, but access is still sequential.
rtc::RaceChecker audio_thread_race_checker_;
mutable Mutex volume_settings_mutex_;
const uint32_t ssrc_;
bool sending_ RTC_GUARDED_BY(&worker_thread_checker_) = false;
RtcEventLog* const event_log_;
std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
std::unique_ptr<RTPSenderAudio> rtp_sender_audio_;
std::unique_ptr<AudioCodingModule> audio_coding_;
// This is just an offset, RTP module will add its own random offset.
uint32_t timestamp_ RTC_GUARDED_BY(audio_thread_race_checker_) = 0;
absl::optional<int64_t> last_capture_timestamp_ms_
RTC_GUARDED_BY(audio_thread_race_checker_);
RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_checker_);
bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_) = false;
bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_checker_) = false;
PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) =
nullptr;
const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_;
const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
RTC_NO_UNIQUE_ADDRESS SequenceChecker construction_thread_;
std::atomic<bool> include_audio_level_indication_ = false;
std::atomic<bool> encoder_queue_is_active_ = false;
std::atomic<bool> first_frame_ = true;
// E2EE Audio Frame Encryption
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_
RTC_GUARDED_BY(encoder_queue_checker_);
// E2EE Frame Encryption Options
const webrtc::CryptoOptions crypto_options_;
// Delegates calls to a frame transformer to transform audio, and
// receives callbacks with the transformed frames; delegates calls to
// ChannelSend::SendRtpAudio to send the transformed audio.
rtc::scoped_refptr<ChannelSendFrameTransformerDelegate>
frame_transformer_delegate_ RTC_GUARDED_BY(encoder_queue_checker_);
mutable Mutex rtcp_counter_mutex_;
RtcpPacketTypeCounter rtcp_packet_type_counter_
RTC_GUARDED_BY(rtcp_counter_mutex_);
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue_;
RTC_NO_UNIQUE_ADDRESS SequenceChecker encoder_queue_checker_;
SdpAudioFormat encoder_format_;
};
const int kTelephoneEventAttenuationdB = 10;
class RtpPacketSenderProxy : public RtpPacketSender {
public:
RtpPacketSenderProxy() : rtp_packet_pacer_(nullptr) {}
void SetPacketPacer(RtpPacketSender* rtp_packet_pacer) {
RTC_DCHECK(thread_checker_.IsCurrent());
MutexLock lock(&mutex_);
rtp_packet_pacer_ = rtp_packet_pacer;
}
void EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) override {
MutexLock lock(&mutex_);
rtp_packet_pacer_->EnqueuePackets(std::move(packets));
}
void RemovePacketsForSsrc(uint32_t ssrc) override {
MutexLock lock(&mutex_);
rtp_packet_pacer_->RemovePacketsForSsrc(ssrc);
}
private:
RTC_NO_UNIQUE_ADDRESS SequenceChecker thread_checker_;
Mutex mutex_;
RtpPacketSender* rtp_packet_pacer_ RTC_GUARDED_BY(&mutex_);
};
int32_t ChannelSend::SendData(AudioFrameType frameType,
uint8_t payloadType,
uint32_t rtp_timestamp,
const uint8_t* payloadData,
size_t payloadSize,
int64_t absolute_capture_timestamp_ms) {
RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
absl::optional<uint8_t> audio_level_dbov;
if (include_audio_level_indication_.load()) {
// Take the averaged audio levels from rms_level_ and reset it before
// invoking any async transformer.
audio_level_dbov = rms_level_.Average();
}
if (frame_transformer_delegate_) {
// Asynchronously transform the payload before sending it. After the payload
// is transformed, the delegate will call SendRtpAudio to send it.
char buf[1024];
rtc::SimpleStringBuilder mime_type(buf);
mime_type << MediaTypeToString(cricket::MEDIA_TYPE_AUDIO) << "/"
<< encoder_format_.name;
frame_transformer_delegate_->Transform(
frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(),
payloadData, payloadSize, absolute_capture_timestamp_ms,
rtp_rtcp_->SSRC(), mime_type.str(), audio_level_dbov);
return 0;
}
return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload,
absolute_capture_timestamp_ms, /*csrcs=*/{},
audio_level_dbov);
}
int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t rtp_timestamp_without_offset,
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs,
absl::optional<uint8_t> audio_level_dbov) {
// E2EE Custom Audio Frame Encryption (This is optional).
// Keep this buffer around for the lifetime of the send call.
rtc::Buffer encrypted_audio_payload;
// We don't invoke encryptor if payload is empty, which means we are to send
// DTMF, or the encoder entered DTX.
// TODO(minyue): see whether DTMF packets should be encrypted or not. In
// current implementation, they are not.
if (!payload.empty()) {
if (frame_encryptor_ != nullptr) {
// TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
// Allocate a buffer to hold the maximum possible encrypted payload.
size_t max_ciphertext_size = frame_encryptor_->GetMaxCiphertextByteSize(
cricket::MEDIA_TYPE_AUDIO, payload.size());
encrypted_audio_payload.SetSize(max_ciphertext_size);
// Encrypt the audio payload into the buffer.
size_t bytes_written = 0;
int encrypt_status = frame_encryptor_->Encrypt(
cricket::MEDIA_TYPE_AUDIO, rtp_rtcp_->SSRC(),
/*additional_data=*/nullptr, payload, encrypted_audio_payload,
&bytes_written);
if (encrypt_status != 0) {
RTC_DLOG(LS_ERROR)
<< "Channel::SendData() failed encrypt audio payload: "
<< encrypt_status;
return -1;
}
// Resize the buffer to the exact number of bytes actually used.
encrypted_audio_payload.SetSize(bytes_written);
// Rewrite the payloadData and size to the new encrypted payload.
payload = encrypted_audio_payload;
} else if (crypto_options_.sframe.require_frame_encryption) {
RTC_DLOG(LS_ERROR) << "Channel::SendData() failed sending audio payload: "
"A frame encryptor is required but one is not set.";
return -1;
}
}
// Push data from ACM to RTP/RTCP-module to deliver audio frame for
// packetization.
if (!rtp_rtcp_->OnSendingRtpFrame(rtp_timestamp_without_offset,
absolute_capture_timestamp_ms, payloadType,
/*force_sender_report=*/false)) {
return -1;
}
// RTCPSender has it's own copy of the timestamp offset, added in
// RTCPSender::BuildSR, hence we must not add the in the offset for the above
// call.
// TODO(nisse): Delete RTCPSender:timestamp_offset_, and see if we can confine
// knowledge of the offset to a single place.
// This call will trigger Transport::SendPacket() from the RTP/RTCP module.
RTPSenderAudio::RtpAudioFrame frame = {
.type = frameType,
.payload = payload,
.payload_id = payloadType,
.rtp_timestamp =
rtp_timestamp_without_offset + rtp_rtcp_->StartTimestamp(),
.csrcs = csrcs};
if (absolute_capture_timestamp_ms > 0) {
frame.capture_time = Timestamp::Millis(absolute_capture_timestamp_ms);
}
if (include_audio_level_indication_.load() && audio_level_dbov) {
frame.audio_level_dbov = *audio_level_dbov;
}
if (!rtp_sender_audio_->SendAudio(frame)) {
RTC_DLOG(LS_ERROR)
<< "ChannelSend::SendData() failed to send data to RTP/RTCP module";
return -1;
}
return 0;
}
ChannelSend::ChannelSend(
Clock* clock,
TaskQueueFactory* task_queue_factory,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options,
bool extmap_allow_mixed,
int rtcp_report_interval_ms,
uint32_t ssrc,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
RtpTransportControllerSendInterface* transport_controller,
const FieldTrialsView& field_trials)
: ssrc_(ssrc),
event_log_(rtc_event_log),
rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
retransmission_rate_limiter_(
new RateLimiter(clock, kMaxRetransmissionWindowMs)),
frame_encryptor_(frame_encryptor),
crypto_options_(crypto_options),
encoder_queue_(task_queue_factory->CreateTaskQueue(
"AudioEncoder",
TaskQueueFactory::Priority::NORMAL)),
encoder_queue_checker_(encoder_queue_.get()),
encoder_format_("x-unknown", 0, 0) {
audio_coding_ = AudioCodingModule::Create();
RtpRtcpInterface::Configuration configuration;
configuration.report_block_data_observer = this;
configuration.network_link_rtcp_observer =
transport_controller->GetRtcpObserver();
configuration.clock = (clock ? clock : Clock::GetRealTimeClock());
configuration.audio = true;
configuration.outgoing_transport = rtp_transport;
configuration.paced_sender = rtp_packet_pacer_proxy_.get();
configuration.event_log = event_log_;
configuration.rtt_stats = rtcp_rtt_stats;
if (field_trials.IsDisabled("WebRTC-DisableRtxRateLimiter")) {
configuration.retransmission_rate_limiter =
retransmission_rate_limiter_.get();
}
configuration.extmap_allow_mixed = extmap_allow_mixed;
configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
configuration.rtcp_packet_type_counter_observer = this;
configuration.local_media_ssrc = ssrc;
configuration.field_trials = &field_trials;
rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration);
rtp_rtcp_->SetSendingMediaStatus(false);
rtp_sender_audio_ = std::make_unique<RTPSenderAudio>(configuration.clock,
rtp_rtcp_->RtpSender());
// Ensure that RTCP is enabled by default for the created channel.
rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
int error = audio_coding_->RegisterTransportCallback(this);
RTC_DCHECK_EQ(0, error);
if (frame_transformer)
InitFrameTransformerDelegate(std::move(frame_transformer));
}
ChannelSend::~ChannelSend() {
RTC_DCHECK(construction_thread_.IsCurrent());
// Resets the delegate's callback to ChannelSend::SendRtpAudio.
if (frame_transformer_delegate_)
frame_transformer_delegate_->Reset();
StopSend();
int error = audio_coding_->RegisterTransportCallback(NULL);
RTC_DCHECK_EQ(0, error);
// Delete the encoder task queue first to ensure that there are no running
// tasks when the other members are destroyed.
encoder_queue_ = nullptr;
}
void ChannelSend::StartSend() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK(!sending_);
sending_ = true;
RTC_DCHECK(packet_router_);
packet_router_->AddSendRtpModule(rtp_rtcp_.get(), /*remb_candidate=*/false);
rtp_rtcp_->SetSendingMediaStatus(true);
int ret = rtp_rtcp_->SetSendingStatus(true);
RTC_DCHECK_EQ(0, ret);
// It is now OK to start processing on the encoder task queue.
first_frame_.store(true);
encoder_queue_is_active_.store(true);
}
void ChannelSend::StopSend() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (!sending_) {
return;
}
sending_ = false;
encoder_queue_is_active_.store(false);
// Wait until all pending encode tasks are executed and clear any remaining
// buffers in the encoder.
rtc::Event flush;
encoder_queue_->PostTask([this, &flush]() {
RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
CallEncoder([](AudioEncoder* encoder) { encoder->Reset(); });
flush.Set();
});
flush.Wait(rtc::Event::kForever);
// Reset sending SSRC and sequence number and triggers direct transmission
// of RTCP BYE
if (rtp_rtcp_->SetSendingStatus(false) == -1) {
RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending";
}
rtp_rtcp_->SetSendingMediaStatus(false);
RTC_DCHECK(packet_router_);
packet_router_->RemoveSendRtpModule(rtp_rtcp_.get());
rtp_packet_pacer_proxy_->RemovePacketsForSsrc(rtp_rtcp_->SSRC());
}
void ChannelSend::SetEncoder(int payload_type,
const SdpAudioFormat& encoder_format,
std::unique_ptr<AudioEncoder> encoder) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK_GE(payload_type, 0);
RTC_DCHECK_LE(payload_type, 127);
// The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate)
// as well as some other things, so we collect this info and send it along.
rtp_rtcp_->RegisterSendPayloadFrequency(payload_type,
encoder->RtpTimestampRateHz());
rtp_sender_audio_->RegisterAudioPayload("audio", payload_type,
encoder->RtpTimestampRateHz(),
encoder->NumChannels(), 0);
encoder_format_ = encoder_format;
audio_coding_->SetEncoder(std::move(encoder));
}
void ChannelSend::ModifyEncoder(
rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
// This method can be called on the worker thread, module process thread
// or network thread. Audio coding is thread safe, so we do not need to
// enforce the calling thread.
audio_coding_->ModifyEncoder(modifier);
}
void ChannelSend::CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) {
ModifyEncoder([modifier](std::unique_ptr<AudioEncoder>* encoder_ptr) {
if (*encoder_ptr) {
modifier(encoder_ptr->get());
} else {
RTC_DLOG(LS_WARNING) << "Trying to call unset encoder.";
}
});
}
void ChannelSend::OnBitrateAllocation(BitrateAllocationUpdate update) {
// This method can be called on the worker thread, module process thread
// or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged.
// TODO(solenberg): Figure out a good way to check this or enforce calling
// rules.
// RTC_DCHECK(worker_thread_checker_.IsCurrent() ||
// module_process_thread_checker_.IsCurrent());
CallEncoder([&](AudioEncoder* encoder) {
encoder->OnReceivedUplinkAllocation(update);
});
retransmission_rate_limiter_->SetMaxRate(update.target_bitrate.bps());
}
int ChannelSend::GetTargetBitrate() const {
return audio_coding_->GetTargetBitrate();
}
void ChannelSend::OnReportBlockDataUpdated(ReportBlockData report_block) {
float packet_loss_rate = report_block.fraction_lost();
CallEncoder([&](AudioEncoder* encoder) {
encoder->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
});
}
void ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// Deliver RTCP packet to RTP/RTCP module for parsing
rtp_rtcp_->IncomingRtcpPacket(rtc::MakeArrayView(data, length));
int64_t rtt = GetRTT();
if (rtt == 0) {
// Waiting for valid RTT.
return;
}
int64_t nack_window_ms = rtt;
if (nack_window_ms < kMinRetransmissionWindowMs) {
nack_window_ms = kMinRetransmissionWindowMs;
} else if (nack_window_ms > kMaxRetransmissionWindowMs) {
nack_window_ms = kMaxRetransmissionWindowMs;
}
retransmission_rate_limiter_->SetWindowSize(nack_window_ms);
OnReceivedRtt(rtt);
}
void ChannelSend::SetInputMute(bool enable) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
MutexLock lock(&volume_settings_mutex_);
input_mute_ = enable;
}
bool ChannelSend::InputMute() const {
MutexLock lock(&volume_settings_mutex_);
return input_mute_;
}
bool ChannelSend::SendTelephoneEventOutband(int event, int duration_ms) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK_LE(0, event);
RTC_DCHECK_GE(255, event);
RTC_DCHECK_LE(0, duration_ms);
RTC_DCHECK_GE(65535, duration_ms);
if (!sending_) {
return false;
}
if (rtp_sender_audio_->SendTelephoneEvent(
event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
RTC_DLOG(LS_ERROR) << "SendTelephoneEvent() failed to send event";
return false;
}
return true;
}
void ChannelSend::RegisterCngPayloadType(int payload_type,
int payload_frequency) {
rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, payload_frequency);
rtp_sender_audio_->RegisterAudioPayload("CN", payload_type, payload_frequency,
1, 0);
}
void ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK_LE(0, payload_type);
RTC_DCHECK_GE(127, payload_type);
rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, payload_frequency);
rtp_sender_audio_->RegisterAudioPayload("telephone-event", payload_type,
payload_frequency, 0, 0);
}
void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
include_audio_level_indication_.store(enable);
if (enable) {
rtp_rtcp_->RegisterRtpHeaderExtension(AudioLevelExtension::Uri(), id);
} else {
rtp_rtcp_->DeregisterSendRtpHeaderExtension(AudioLevelExtension::Uri());
}
}
void ChannelSend::RegisterSenderCongestionControlObjects(
RtpTransportControllerSendInterface* transport) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RtpPacketSender* rtp_packet_pacer = transport->packet_sender();
PacketRouter* packet_router = transport->packet_router();
RTC_DCHECK(rtp_packet_pacer);
RTC_DCHECK(packet_router);
RTC_DCHECK(!packet_router_);
rtp_packet_pacer_proxy_->SetPacketPacer(rtp_packet_pacer);
rtp_rtcp_->SetStorePacketsStatus(true, 600);
packet_router_ = packet_router;
}
void ChannelSend::ResetSenderCongestionControlObjects() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK(packet_router_);
rtp_rtcp_->SetStorePacketsStatus(false, 600);
packet_router_ = nullptr;
rtp_packet_pacer_proxy_->SetPacketPacer(nullptr);
}
void ChannelSend::SetRTCP_CNAME(absl::string_view c_name) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// Note: SetCNAME() accepts a c string of length at most 255.
const std::string c_name_limited(c_name.substr(0, 255));
int ret = rtp_rtcp_->SetCNAME(c_name_limited.c_str()) != 0;
RTC_DCHECK_EQ(0, ret) << "SetRTCP_CNAME() failed to set RTCP CNAME";
}
std::vector<ReportBlockData> ChannelSend::GetRemoteRTCPReportBlocks() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// Get the report blocks from the latest received RTCP Sender or Receiver
// Report. Each element in the vector contains the sender's SSRC and a
// report block according to RFC 3550.
return rtp_rtcp_->GetLatestReportBlockData();
}
CallSendStatistics ChannelSend::GetRTCPStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
CallSendStatistics stats = {0};
stats.rttMs = GetRTT();
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
rtp_rtcp_->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
stats.payload_bytes_sent =
rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
stats.header_and_padding_bytes_sent =
rtp_stats.transmitted.padding_bytes + rtp_stats.transmitted.header_bytes +
rtx_stats.transmitted.padding_bytes + rtx_stats.transmitted.header_bytes;
// TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up in
// separate outbound-rtp stream objects.
stats.retransmitted_bytes_sent = rtp_stats.retransmitted.payload_bytes;
stats.packetsSent =
rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
stats.total_packet_send_delay = rtp_stats.transmitted.total_packet_delay;
stats.retransmitted_packets_sent = rtp_stats.retransmitted.packets;
stats.report_block_datas = rtp_rtcp_->GetLatestReportBlockData();
{
MutexLock lock(&rtcp_counter_mutex_);
stats.nacks_received = rtcp_packet_type_counter_.nack_packets;
}
return stats;
}
void ChannelSend::RtcpPacketTypesCounterUpdated(
uint32_t ssrc,
const RtcpPacketTypeCounter& packet_counter) {
if (ssrc != ssrc_) {
return;
}
MutexLock lock(&rtcp_counter_mutex_);
rtcp_packet_type_counter_ = packet_counter;
}
void ChannelSend::ProcessAndEncodeAudio(
std::unique_ptr<AudioFrame> audio_frame) {
TRACE_EVENT0("webrtc", "ChannelSend::ProcessAndEncodeAudio");
RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
RTC_DCHECK_LE(audio_frame->num_channels_, 8);
if (!encoder_queue_is_active_.load()) {
return;
}
// Update `timestamp_` based on the capture timestamp for the first frame
// after sending is resumed.
if (first_frame_.load()) {
first_frame_.store(false);
if (last_capture_timestamp_ms_ &&
audio_frame->absolute_capture_timestamp_ms()) {
int64_t diff_ms = *audio_frame->absolute_capture_timestamp_ms() -
*last_capture_timestamp_ms_;
// Truncate to whole frames and subtract one since `timestamp_` was
// incremented after the last frame.
int64_t diff_frames = diff_ms * audio_frame->sample_rate_hz() / 1000 /
audio_frame->samples_per_channel() -
1;
timestamp_ += std::max<int64_t>(
diff_frames * audio_frame->samples_per_channel(), 0);
}
}
audio_frame->timestamp_ = timestamp_;
timestamp_ += audio_frame->samples_per_channel_;
last_capture_timestamp_ms_ = audio_frame->absolute_capture_timestamp_ms();
// Profile time between when the audio frame is added to the task queue and
// when the task is actually executed.
audio_frame->UpdateProfileTimeStamp();
encoder_queue_->PostTask(
[this, audio_frame = std::move(audio_frame)]() mutable {
RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
if (!encoder_queue_is_active_.load()) {
return;
}
// Measure time between when the audio frame is added to the task queue
// and when the task is actually executed. Goal is to keep track of
// unwanted extra latency added by the task queue.
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Audio.EncodingTaskQueueLatencyMs",
audio_frame->ElapsedProfileTimeMs());
bool is_muted = InputMute();
AudioFrameOperations::Mute(audio_frame.get(), previous_frame_muted_,
is_muted);
if (include_audio_level_indication_.load()) {
size_t length =
audio_frame->samples_per_channel_ * audio_frame->num_channels_;
RTC_CHECK_LE(length, AudioFrame::kMaxDataSizeBytes);
if (is_muted && previous_frame_muted_) {
rms_level_.AnalyzeMuted(length);
} else {
rms_level_.Analyze(
rtc::ArrayView<const int16_t>(audio_frame->data(), length));
}
}
previous_frame_muted_ = is_muted;
// This call will trigger AudioPacketizationCallback::SendData if
// encoding is done and payload is ready for packetization and
// transmission. Otherwise, it will return without invoking the
// callback.
if (audio_coding_->Add10MsData(*audio_frame) < 0) {
RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed.";
return;
}
});
}
ANAStats ChannelSend::GetANAStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return audio_coding_->GetANAStats();
}
RtpRtcpInterface* ChannelSend::GetRtpRtcp() const {
return rtp_rtcp_.get();
}
int64_t ChannelSend::GetRTT() const {
std::vector<ReportBlockData> report_blocks =
rtp_rtcp_->GetLatestReportBlockData();
if (report_blocks.empty()) {
return 0;
}
// We don't know in advance the remote ssrc used by the other end's receiver
// reports, so use the first report block for the RTT.
return report_blocks.front().last_rtt().ms();
}
void ChannelSend::SetFrameEncryptor(
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
encoder_queue_->PostTask([this, frame_encryptor]() mutable {
RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
frame_encryptor_ = std::move(frame_encryptor);
});
}
void ChannelSend::SetEncoderToPacketizerFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (!frame_transformer)
return;
encoder_queue_->PostTask(
[this, frame_transformer = std::move(frame_transformer)]() mutable {
RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
InitFrameTransformerDelegate(std::move(frame_transformer));
});
}
void ChannelSend::OnReceivedRtt(int64_t rtt_ms) {
// Invoke audio encoders OnReceivedRtt().
CallEncoder(
[rtt_ms](AudioEncoder* encoder) { encoder->OnReceivedRtt(rtt_ms); });
}
void ChannelSend::InitFrameTransformerDelegate(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
RTC_DCHECK(frame_transformer);
RTC_DCHECK(!frame_transformer_delegate_);
// Pass a callback to ChannelSend::SendRtpAudio, to be called by the delegate
// to send the transformed audio.
ChannelSendFrameTransformerDelegate::SendFrameCallback send_audio_callback =
[this](AudioFrameType frameType, uint8_t payloadType,
uint32_t rtp_timestamp_with_offset,
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs,
absl::optional<uint8_t> audio_level_dbov) {
RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
return SendRtpAudio(
frameType, payloadType,
rtp_timestamp_with_offset - rtp_rtcp_->StartTimestamp(), payload,
absolute_capture_timestamp_ms, csrcs, audio_level_dbov);
};
frame_transformer_delegate_ =
rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
std::move(send_audio_callback), std::move(frame_transformer),
encoder_queue_.get());
frame_transformer_delegate_->Init();
}
} // namespace
std::unique_ptr<ChannelSendInterface> CreateChannelSend(
Clock* clock,
TaskQueueFactory* task_queue_factory,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options,
bool extmap_allow_mixed,
int rtcp_report_interval_ms,
uint32_t ssrc,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
RtpTransportControllerSendInterface* transport_controller,
const FieldTrialsView& field_trials) {
return std::make_unique<ChannelSend>(
clock, task_queue_factory, rtp_transport, rtcp_rtt_stats, rtc_event_log,
frame_encryptor, crypto_options, extmap_allow_mixed,
rtcp_report_interval_ms, ssrc, std::move(frame_transformer),
transport_controller, field_trials);
}
} // namespace voe
} // namespace webrtc