zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
dba16fd4236b71d9dc2219ce34a77ca2d87723a5
platform-external-webrtc/audio/channel_send.cc
Sebastian Jansson 44dd9f29c7 Adds ChannelSend specific encoder task queue. 
Before this change the encoder tasks runs on a shared worker queue.
That makes the destruction require synchronization to avoid races.
By keeping a separate encode queue to ChannelSend, we can safely
destruct the object without worrying for left over tasks, as they
will be stopped when the task queue is destroyed.

For TaskQueue implementations using one thread per TaskQueue this
will increase the thread count by the number of AudioSendStreams,
which typically is just one.

This is partly a reland of 9b9344742b186b14d87e827e71a1757f4c94b30e

Original change's description:
> Removes lock from ChannelSend.
>
> The lock isn't really needed as encoder_queue_is_active_ can be checked
> on the task queue to provide synchronization.
>
> There is one behavioral change due to this: We will not cancel any currently
> pending encoding tasks when we stop sending, they will be allowed to finish.
>
> Bug: webrtc:10365
> Change-Id: I2b4897dde8d49bc7ee5d2d69694616aee8aaea38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125096
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26963}

Bug: webrtc:10365
Change-Id: Iafb84e25d90ec8639359be80fad65763d08e5719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125740
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27038}
2019-03-08 15:53:12 +00:00

1257 lines
46 KiB
C++
Raw Blame History

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "audio/channel_send.h"
#include <algorithm>
#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "audio/utility/audio_frame_operations.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_processing/rms_level.h"
#include "modules/pacing/packet_router.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/checks.h"
#include "rtc_base/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 kAudioFrameSpeech:
return MediaTransportEncodedAudioFrame::FrameType::kSpeech;
break;
case kAudioFrameCN:
return MediaTransportEncodedAudioFrame::FrameType::
kDiscontinuousTransmission;
break;
default:
RTC_CHECK(false) << "Unexpected frame type=" << frame_type;
break;
}
}
class RtpPacketSenderProxy;
class TransportFeedbackProxy;
class TransportSequenceNumberProxy;
class VoERtcpObserver;
class ChannelSend
: public ChannelSendInterface,
public AudioPacketizationCallback, // receive encoded packets from the
// ACM
public TargetTransferRateObserver {
public:
// TODO(nisse): Make OnUplinkPacketLossRate public, and delete friend
// declaration.
friend class VoERtcpObserver;
ChannelSend(Clock* clock,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options,
bool extmap_allow_mixed,
int rtcp_report_interval_ms);
~ChannelSend() override;
// Send using this encoder, with this payload type.
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 SetLocalSSRC(uint32_t ssrc) override;
void SetRid(const std::string& rid,
int extension_id,
int repaired_extension_id) override;
void SetMid(const std::string& mid, int extension_id) override;
void SetExtmapAllowMixed(bool extmap_allow_mixed) override;
void SetSendAudioLevelIndicationStatus(bool enable, int id) override;
void EnableSendTransportSequenceNumber(int id) override;
void RegisterSenderCongestionControlObjects(
RtpTransportControllerSendInterface* transport,
RtcpBandwidthObserver* bandwidth_observer) override;
void ResetSenderCongestionControlObjects() override;
void SetRTCP_CNAME(absl::string_view c_name) override;
std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const override;
CallSendStatistics GetRTCPStatistics() const override;
// ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
// which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
// the actual processing of the audio takes place. The processing mainly
// consists of encoding and preparing the result for sending by adding it to a
// send queue.
// The main reason for using a task queue here is to release the native,
// OS-specific, audio capture thread as soon as possible to ensure that it
// can go back to sleep and be prepared to deliver an new captured audio
// packet.
void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) override;
// The existence of this function alongside OnUplinkPacketLossRate is
// a compromise. We want the encoder to be agnostic of the PLR source, but
// we also don't want it to receive conflicting information from TWCC and
// from RTCP-XR.
void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) override;
void OnRecoverableUplinkPacketLossRate(
float recoverable_packet_loss_rate) override;
int64_t GetRTT() const override;
// E2EE Custom Audio Frame Encryption
void SetFrameEncryptor(
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override;
private:
// From AudioPacketizationCallback in the ACM
int32_t SendData(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) override;
void OnUplinkPacketLossRate(float packet_loss_rate);
bool InputMute() const;
int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type, int id);
int32_t SendRtpAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
rtc::ArrayView<const uint8_t> payload,
const RTPFragmentationHeader* fragmentation)
RTC_RUN_ON(encoder_queue_);
int32_t SendMediaTransportAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
rtc::ArrayView<const uint8_t> payload,
const RTPFragmentationHeader* fragmentation)
RTC_RUN_ON(encoder_queue_);
// Return media transport or nullptr if using RTP.
MediaTransportInterface* media_transport() { return media_transport_; }
// Called on the encoder task queue when a new input audio frame is ready
// for encoding.
void ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input)
RTC_RUN_ON(encoder_queue_);
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<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
rtc::ThreadChecker construction_thread_;
const bool use_twcc_plr_for_ana_;
bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_) = false;
MediaTransportInterface* const media_transport_;
int media_transport_sequence_number_ RTC_GUARDED_BY(encoder_queue_) = 0;
rtc::CriticalSection media_transport_lock_;
// Currently set by SetLocalSSRC.
uint64_t media_transport_channel_id_ RTC_GUARDED_BY(&media_transport_lock_) =
0;
// Cache payload type and sampling frequency from most recent call to
// SetEncoder. Needed to set MediaTransportEncodedAudioFrame metadata, and
// invalidate on encoder change.
int media_transport_payload_type_ RTC_GUARDED_BY(&media_transport_lock_);
int media_transport_sampling_frequency_
RTC_GUARDED_BY(&media_transport_lock_);
// E2EE Audio Frame Encryption
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_
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_.DetachFromThread();
network_thread_.DetachFromThread();
}
void SetTransportFeedbackObserver(
TransportFeedbackObserver* feedback_observer) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
rtc::CritScope lock(&crit_);
feedback_observer_ = feedback_observer;
}
// Implements TransportFeedbackObserver.
void AddPacket(uint32_t ssrc,
uint16_t sequence_number,
size_t length,
const PacedPacketInfo& pacing_info) override {
RTC_DCHECK(pacer_thread_.CalledOnValidThread());
rtc::CritScope lock(&crit_);
if (feedback_observer_)
feedback_observer_->AddPacket(ssrc, sequence_number, length, pacing_info);
}
void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override {
RTC_DCHECK(network_thread_.CalledOnValidThread());
rtc::CritScope lock(&crit_);
if (feedback_observer_)
feedback_observer_->OnTransportFeedback(feedback);
}
private:
rtc::CriticalSection crit_;
rtc::ThreadChecker thread_checker_;
rtc::ThreadChecker pacer_thread_;
rtc::ThreadChecker network_thread_;
TransportFeedbackObserver* feedback_observer_ RTC_GUARDED_BY(&crit_);
};
class TransportSequenceNumberProxy : public TransportSequenceNumberAllocator {
public:
TransportSequenceNumberProxy() : seq_num_allocator_(nullptr) {
pacer_thread_.DetachFromThread();
}
void SetSequenceNumberAllocator(
TransportSequenceNumberAllocator* seq_num_allocator) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
rtc::CritScope lock(&crit_);
seq_num_allocator_ = seq_num_allocator;
}
// Implements TransportSequenceNumberAllocator.
uint16_t AllocateSequenceNumber() override {
RTC_DCHECK(pacer_thread_.CalledOnValidThread());
rtc::CritScope lock(&crit_);
if (!seq_num_allocator_)
return 0;
return seq_num_allocator_->AllocateSequenceNumber();
}
private:
rtc::CriticalSection crit_;
rtc::ThreadChecker thread_checker_;
rtc::ThreadChecker pacer_thread_;
TransportSequenceNumberAllocator* seq_num_allocator_ RTC_GUARDED_BY(&crit_);
};
class RtpPacketSenderProxy : public RtpPacketSender {
public:
RtpPacketSenderProxy() : rtp_packet_sender_(nullptr) {}
void SetPacketSender(RtpPacketSender* rtp_packet_sender) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
rtc::CritScope lock(&crit_);
rtp_packet_sender_ = rtp_packet_sender;
}
// Implements RtpPacketSender.
void InsertPacket(Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) override {
rtc::CritScope lock(&crit_);
if (rtp_packet_sender_) {
rtp_packet_sender_->InsertPacket(priority, ssrc, sequence_number,
capture_time_ms, bytes, retransmission);
}
}
void SetAccountForAudioPackets(bool account_for_audio) override {
RTC_NOTREACHED();
}
private:
rtc::ThreadChecker thread_checker_;
rtc::CriticalSection crit_;
RtpPacketSender* rtp_packet_sender_ RTC_GUARDED_BY(&crit_);
};
class VoERtcpObserver : public RtcpBandwidthObserver {
public:
explicit VoERtcpObserver(ChannelSend* owner)
: owner_(owner), bandwidth_observer_(nullptr) {}
~VoERtcpObserver() override {}
void SetBandwidthObserver(RtcpBandwidthObserver* bandwidth_observer) {
rtc::CritScope lock(&crit_);
bandwidth_observer_ = bandwidth_observer;
}
void OnReceivedEstimatedBitrate(uint32_t bitrate) override {
rtc::CritScope lock(&crit_);
if (bandwidth_observer_) {
bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate);
}
}
void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
int64_t rtt,
int64_t now_ms) override {
{
rtc::CritScope lock(&crit_);
if (bandwidth_observer_) {
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, rtt,
now_ms);
}
}
// TODO(mflodman): Do we need to aggregate reports here or can we jut send
// what we get? I.e. do we ever get multiple reports bundled into one RTCP
// report for VoiceEngine?
if (report_blocks.empty())
return;
int fraction_lost_aggregate = 0;
int total_number_of_packets = 0;
// If receiving multiple report blocks, calculate the weighted average based
// on the number of packets a report refers to.
for (ReportBlockList::const_iterator block_it = report_blocks.begin();
block_it != report_blocks.end(); ++block_it) {
// Find the previous extended high sequence number for this remote SSRC,
// to calculate the number of RTP packets this report refers to. Ignore if
// we haven't seen this SSRC before.
std::map<uint32_t, uint32_t>::iterator seq_num_it =
extended_max_sequence_number_.find(block_it->source_ssrc);
int number_of_packets = 0;
if (seq_num_it != extended_max_sequence_number_.end()) {
number_of_packets =
block_it->extended_highest_sequence_number - seq_num_it->second;
}
fraction_lost_aggregate += number_of_packets * block_it->fraction_lost;
total_number_of_packets += number_of_packets;
extended_max_sequence_number_[block_it->source_ssrc] =
block_it->extended_highest_sequence_number;
}
int weighted_fraction_lost = 0;
if (total_number_of_packets > 0) {
weighted_fraction_lost =
(fraction_lost_aggregate + total_number_of_packets / 2) /
total_number_of_packets;
}
owner_->OnUplinkPacketLossRate(weighted_fraction_lost / 255.0f);
}
private:
ChannelSend* owner_;
// Maps remote side ssrc to extended highest sequence number received.
std::map<uint32_t, uint32_t> extended_max_sequence_number_;
rtc::CriticalSection crit_;
RtcpBandwidthObserver* bandwidth_observer_ RTC_GUARDED_BY(crit_);
};
int32_t ChannelSend::SendData(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) {
RTC_DCHECK_RUN_ON(&encoder_queue_);
rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
if (media_transport() != nullptr) {
if (frameType == kEmptyFrame) {
// TODO(bugs.webrtc.org/9719): Media transport Send doesn't support
// sending empty frames.
return 0;
}
return SendMediaTransportAudio(frameType, payloadType, timeStamp, payload,
fragmentation);
} else {
return SendRtpAudio(frameType, payloadType, timeStamp, payload,
fragmentation);
}
}
int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
rtc::ArrayView<const uint8_t> payload,
const RTPFragmentationHeader* fragmentation) {
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;
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,
const RTPFragmentationHeader* fragmentation) {
// TODO(nisse): Use null _transportPtr for MediaTransport.
// RTC_DCHECK(_transportPtr == nullptr);
uint64_t channel_id;
int sampling_rate_hz;
{
rtc::CritScope cs(&media_transport_lock_);
if (media_transport_payload_type_ != payloadType) {
// Payload type is being changed, media_transport_sampling_frequency_,
// no longer current.
return -1;
}
sampling_rate_hz = media_transport_sampling_frequency_;
channel_id = media_transport_channel_id_;
}
MediaTransportEncodedAudioFrame frame(
/*sampling_rate_hz=*/sampling_rate_hz,
// TODO(nisse): Timestamp and sample index are the same for all supported
// audio codecs except G722. Refactor audio coding module to only use
// sample index, and leave translation to RTP time, when needed, for
// RTP-specific code.
/*starting_sample_index=*/timeStamp,
// Sample count isn't conveniently available from the AudioCodingModule,
// and needs some refactoring to wire up in a good way. For now, left as
// zero.
/*sample_count=*/0,
/*sequence_number=*/media_transport_sequence_number_,
MediaTransportFrameTypeForWebrtcFrameType(frameType), payloadType,
std::vector<uint8_t>(payload.begin(), payload.end()));
// TODO(nisse): Introduce a MediaTransportSender object bound to a specific
// channel id.
RTCError rtc_error =
media_transport()->SendAudioFrame(channel_id, std::move(frame));
if (!rtc_error.ok()) {
RTC_LOG(LS_ERROR) << "Failed to send frame, rtc_error="
<< ToString(rtc_error.type()) << ", "
<< rtc_error.message();
return -1;
}
++media_transport_sequence_number_;
return 0;
}
ChannelSend::ChannelSend(Clock* clock,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options,
bool extmap_allow_mixed,
int rtcp_report_interval_ms)
: event_log_(rtc_event_log),
_timeStamp(0), // This is just an offset, RTP module will add it's own
// random offset
_moduleProcessThreadPtr(module_process_thread),
input_mute_(false),
previous_frame_muted_(false),
_includeAudioLevelIndication(false),
rtcp_observer_(new VoERtcpObserver(this)),
feedback_observer_proxy_(new TransportFeedbackProxy()),
seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
retransmission_rate_limiter_(
new RateLimiter(clock, kMaxRetransmissionWindowMs)),
use_twcc_plr_for_ana_(
webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled"),
media_transport_(media_transport),
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_.DetachFromThread();
audio_coding_.reset(AudioCodingModule::Create(AudioCodingModule::Config()));
RtpRtcp::Configuration configuration;
// We gradually remove codepaths that depend on RTP when using media
// transport. All of this logic should be moved to the future
// RTPMediaTransport. In this case it means that overhead and bandwidth
// observers should not be called when using media transport.
if (!media_transport_) {
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_sender_proxy_.get();
configuration.transport_sequence_number_allocator =
seq_num_allocator_proxy_.get();
configuration.event_log = event_log_;
configuration.rtt_stats = rtcp_rtt_stats;
configuration.retransmission_rate_limiter =
retransmission_rate_limiter_.get();
configuration.extmap_allow_mixed = extmap_allow_mixed;
configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
_rtpRtcpModule = RtpRtcp::Create(configuration);
_rtpRtcpModule->SetSendingMediaStatus(false);
rtp_sender_audio_ = absl::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_) {
RTC_DLOG(LS_INFO) << "Setting media_transport_ rate observers.";
media_transport_->AddTargetTransferRateObserver(this);
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.
// Note that, the module will keep generating RTCP until it is explicitly
// disabled by the user.
// After StopListen (when no sockets exists), RTCP packets will no longer
// be transmitted since the Transport object will then be invalid.
// RTCP is enabled by default.
_rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
int error = audio_coding_->RegisterTransportCallback(this);
RTC_DCHECK_EQ(0, error);
}
ChannelSend::~ChannelSend() {
RTC_DCHECK(construction_thread_.CalledOnValidThread());
if (media_transport_) {
media_transport_->RemoveTargetTransferRateObserver(this);
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_) {
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_.CalledOnValidThread() ||
// module_process_thread_checker_.CalledOnValidThread());
rtc::CritScope lock(&bitrate_crit_section_);
CallEncoder([&](AudioEncoder* encoder) {
encoder->OnReceivedUplinkAllocation(update);
});
retransmission_rate_limiter_->SetMaxRate(update.target_bitrate.bps());
configured_bitrate_bps_ = update.target_bitrate.bps();
}
int ChannelSend::GetBitrate() const {
rtc::CritScope lock(&bitrate_crit_section_);
return configured_bitrate_bps_;
}
void ChannelSend::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (!use_twcc_plr_for_ana_)
return;
CallEncoder([&](AudioEncoder* encoder) {
encoder->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
});
}
void ChannelSend::OnRecoverableUplinkPacketLossRate(
float recoverable_packet_loss_rate) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
CallEncoder([&](AudioEncoder* encoder) {
encoder->OnReceivedUplinkRecoverablePacketLossFraction(
recoverable_packet_loss_rate);
});
}
void ChannelSend::OnUplinkPacketLossRate(float packet_loss_rate) {
if (use_twcc_plr_for_ana_)
return;
CallEncoder([&](AudioEncoder* encoder) {
encoder->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
});
}
void ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
// May be called on either worker thread or network thread.
if (media_transport_) {
// Ignore RTCP packets while media transport is used.
// Those packets should not arrive, but we are seeing occasional packets.
return;
}
// 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::SetLocalSSRC(uint32_t ssrc) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK(!sending_);
if (media_transport_) {
rtc::CritScope cs(&media_transport_lock_);
media_transport_channel_id_ = ssrc;
}
_rtpRtcpModule->SetSSRC(ssrc);
}
void ChannelSend::SetRid(const std::string& rid,
int extension_id,
int repaired_extension_id) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (extension_id != 0) {
int ret = SetSendRtpHeaderExtension(!rid.empty(), kRtpExtensionRtpStreamId,
extension_id);
RTC_DCHECK_EQ(0, ret);
}
if (repaired_extension_id != 0) {
int ret = SetSendRtpHeaderExtension(!rid.empty(), kRtpExtensionRtpStreamId,
repaired_extension_id);
RTC_DCHECK_EQ(0, ret);
}
_rtpRtcpModule->SetRid(rid);
}
void ChannelSend::SetMid(const std::string& mid, int extension_id) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
int ret = SetSendRtpHeaderExtension(true, kRtpExtensionMid, extension_id);
RTC_DCHECK_EQ(0, ret);
_rtpRtcpModule->SetMid(mid);
}
void ChannelSend::SetExtmapAllowMixed(bool extmap_allow_mixed) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
_rtpRtcpModule->SetExtmapAllowMixed(extmap_allow_mixed);
}
void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
_includeAudioLevelIndication = enable;
int ret = SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
RTC_DCHECK_EQ(0, ret);
}
void ChannelSend::EnableSendTransportSequenceNumber(int id) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
int ret =
SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id);
RTC_DCHECK_EQ(0, ret);
}
void ChannelSend::RegisterSenderCongestionControlObjects(
RtpTransportControllerSendInterface* transport,
RtcpBandwidthObserver* bandwidth_observer) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RtpPacketSender* rtp_packet_sender = transport->packet_sender();
TransportFeedbackObserver* transport_feedback_observer =
transport->transport_feedback_observer();
PacketRouter* packet_router = transport->packet_router();
RTC_DCHECK(rtp_packet_sender);
RTC_DCHECK(transport_feedback_observer);
RTC_DCHECK(packet_router);
RTC_DCHECK(!packet_router_);
rtcp_observer_->SetBandwidthObserver(bandwidth_observer);
feedback_observer_proxy_->SetTransportFeedbackObserver(
transport_feedback_observer);
seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router);
rtp_packet_sender_proxy_->SetPacketSender(rtp_packet_sender);
_rtpRtcpModule->SetStorePacketsStatus(true, 600);
constexpr bool remb_candidate = false;
packet_router->AddSendRtpModule(_rtpRtcpModule.get(), remb_candidate);
packet_router_ = packet_router;
}
void ChannelSend::ResetSenderCongestionControlObjects() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK(packet_router_);
_rtpRtcpModule->SetStorePacketsStatus(false, 600);
rtcp_observer_->SetBandwidthObserver(nullptr);
feedback_observer_proxy_->SetTransportFeedbackObserver(nullptr);
seq_num_allocator_proxy_->SetSequenceNumberAllocator(nullptr);
packet_router_->RemoveSendRtpModule(_rtpRtcpModule.get());
packet_router_ = nullptr;
rtp_packet_sender_proxy_->SetPacketSender(nullptr);
}
void ChannelSend::SetRTCP_CNAME(absl::string_view c_name) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// Note: SetCNAME() accepts a c string of length at most 255.
const std::string c_name_limited(c_name.substr(0, 255));
int ret = _rtpRtcpModule->SetCNAME(c_name_limited.c_str()) != 0;
RTC_DCHECK_EQ(0, ret) << "SetRTCP_CNAME() failed to set RTCP CNAME";
}
std::vector<ReportBlock> ChannelSend::GetRemoteRTCPReportBlocks() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// Get the report blocks from the latest received RTCP Sender or Receiver
// Report. Each element in the vector contains the sender's SSRC and a
// report block according to RFC 3550.
std::vector<RTCPReportBlock> rtcp_report_blocks;
int ret = _rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks);
RTC_DCHECK_EQ(0, ret);
std::vector<ReportBlock> report_blocks;
std::vector<RTCPReportBlock>::const_iterator it = rtcp_report_blocks.begin();
for (; it != rtcp_report_blocks.end(); ++it) {
ReportBlock report_block;
report_block.sender_SSRC = it->sender_ssrc;
report_block.source_SSRC = it->source_ssrc;
report_block.fraction_lost = it->fraction_lost;
report_block.cumulative_num_packets_lost = it->packets_lost;
report_block.extended_highest_sequence_number =
it->extended_highest_sequence_number;
report_block.interarrival_jitter = it->jitter;
report_block.last_SR_timestamp = it->last_sender_report_timestamp;
report_block.delay_since_last_SR = it->delay_since_last_sender_report;
report_blocks.push_back(report_block);
}
return report_blocks;
}
CallSendStatistics ChannelSend::GetRTCPStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
CallSendStatistics stats = {0};
stats.rttMs = GetRTT();
size_t bytesSent(0);
uint32_t packetsSent(0);
if (_rtpRtcpModule->DataCountersRTP(&bytesSent, &packetsSent) != 0) {
RTC_DLOG(LS_WARNING)
<< "GetRTPStatistics() failed to retrieve RTP datacounters"
<< " => output will not be complete";
}
stats.bytesSent = bytesSent;
stats.packetsSent = packetsSent;
return stats;
}
void ChannelSend::ProcessAndEncodeAudio(
std::unique_ptr<AudioFrame> audio_frame) {
RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
struct ProcessAndEncodeAudio {
void operator()() {
RTC_DCHECK_RUN_ON(&channel->encoder_queue_);
if (!channel->encoder_queue_is_active_) {
return;
}
channel->ProcessAndEncodeAudioOnTaskQueue(audio_frame.get());
}
std::unique_ptr<AudioFrame> audio_frame;
ChannelSend* const channel;
};
// 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(ProcessAndEncodeAudio{std::move(audio_frame), this});
}
void ChannelSend::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
RTC_DCHECK_GT(audio_input->samples_per_channel_, 0);
RTC_DCHECK_LE(audio_input->num_channels_, 2);
// Measure time between when the audio frame is added to the task queue and
// when the task is actually executed. Goal is to keep track of unwanted
// extra latency added by the task queue.
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Audio.EncodingTaskQueueLatencyMs",
audio_input->ElapsedProfileTimeMs());
bool is_muted = InputMute();
AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted);
if (_includeAudioLevelIndication) {
size_t length =
audio_input->samples_per_channel_ * audio_input->num_channels_;
RTC_CHECK_LE(length, AudioFrame::kMaxDataSizeBytes);
if (is_muted && previous_frame_muted_) {
rms_level_.AnalyzeMuted(length);
} else {
rms_level_.Analyze(
rtc::ArrayView<const int16_t>(audio_input->data(), length));
}
}
previous_frame_muted_ = is_muted;
// Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.
// The ACM resamples internally.
audio_input->timestamp_ = _timeStamp;
// This call will trigger AudioPacketizationCallback::SendData if encoding
// is done and payload is ready for packetization and transmission.
// Otherwise, it will return without invoking the callback.
if (audio_coding_->Add10MsData(*audio_input) < 0) {
RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed.";
return;
}
_timeStamp += static_cast<uint32_t>(audio_input->samples_per_channel_);
}
ANAStats ChannelSend::GetANAStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return audio_coding_->GetANAStats();
}
RtpRtcp* ChannelSend::GetRtpRtcp() const {
RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
return _rtpRtcpModule.get();
}
int ChannelSend::SetSendRtpHeaderExtension(bool enable,
RTPExtensionType type,
int id) {
int error = 0;
_rtpRtcpModule->DeregisterSendRtpHeaderExtension(type);
if (enable) {
// TODO(nisse): RtpRtcp::RegisterSendRtpHeaderExtension to take an int
// argument. Currently it wants an uint8_t.
error = _rtpRtcpModule->RegisterSendRtpHeaderExtension(
type, rtc::dchecked_cast<uint8_t>(id));
}
return error;
}
int64_t ChannelSend::GetRTT() const {
if (media_transport_) {
// GetRTT is generally used in the RTCP codepath, where media transport is
// not present and so it shouldn't be needed. But it's also invoked in
// 'GetStats' method, and for now returning media transport RTT here gives
// us "free" rtt stats for media transport.
auto target_rate = media_transport_->GetLatestTargetTransferRate();
if (target_rate.has_value()) {
return target_rate.value().network_estimate.round_trip_time.ms();
}
return 0;
}
RtcpMode method = _rtpRtcpModule->RTCP();
if (method == RtcpMode::kOff) {
return 0;
}
std::vector<RTCPReportBlock> report_blocks;
_rtpRtcpModule->RemoteRTCPStat(&report_blocks);
if (report_blocks.empty()) {
return 0;
}
int64_t rtt = 0;
int64_t avg_rtt = 0;
int64_t max_rtt = 0;
int64_t min_rtt = 0;
// We don't know in advance the remote ssrc used by the other end's receiver
// reports, so use the SSRC of the first report block for calculating the RTT.
if (_rtpRtcpModule->RTT(report_blocks[0].sender_ssrc, &rtt, &avg_rtt,
&min_rtt, &max_rtt) != 0) {
return 0;
}
return rtt;
}
void ChannelSend::SetFrameEncryptor(
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
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_);
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,
MediaTransportInterface* media_transport,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options,
bool extmap_allow_mixed,
int rtcp_report_interval_ms) {
return absl::make_unique<ChannelSend>(
clock, task_queue_factory, module_process_thread, media_transport,
overhead_observer, rtp_transport, rtcp_rtt_stats, rtc_event_log,
frame_encryptor, crypto_options, extmap_allow_mixed,
rtcp_report_interval_ms);
}
} // namespace voe
} // namespace webrtc
Reference in New Issue View Git Blame Copy Permalink