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webrtc/voice_engine/channel.cc
Elad Alon 604c14df58 Reland "Remove deprecated functions from RtcEventLog" 
The unified Log() interface replaces the many old LogX() functions. This helps hide dependencies between the modules which log different events.

TBR=stefan@webrtc.org

Bug: webrtc:8111
Change-Id: I36c8b6c4cf03d738c9033af2e98db6dc200eede9
Reviewed-on: https://webrtc-review.googlesource.com/6940
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20170}
2017-10-05 15:07:53 +00:00

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "voice_engine/channel.h"
#include <algorithm>
#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "api/array_view.h"
#include "audio/utility/audio_frame_operations.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "logging/rtc_event_log/rtc_event_log.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/codecs/audio_format_conversion.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/include/module_common_types.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "modules/rtp_rtcp/include/rtp_receiver.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/checks.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/thread_checker.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#include "voice_engine/utility.h"
namespace webrtc {
namespace voe {
namespace {
constexpr double kAudioSampleDurationSeconds = 0.01;
constexpr int64_t kMaxRetransmissionWindowMs = 1000;
constexpr int64_t kMinRetransmissionWindowMs = 30;
} // namespace
const int kTelephoneEventAttenuationdB = 10;
class RtcEventLogProxy final : public webrtc::RtcEventLog {
public:
RtcEventLogProxy() : event_log_(nullptr) {}
bool StartLogging(std::unique_ptr<RtcEventLogOutput> output) override {
RTC_NOTREACHED();
return false;
}
void StopLogging() override { RTC_NOTREACHED(); }
void Log(std::unique_ptr<RtcEvent> event) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->Log(std::move(event));
}
}
void SetEventLog(RtcEventLog* event_log) {
rtc::CritScope lock(&crit_);
event_log_ = event_log;
}
private:
rtc::CriticalSection crit_;
RtcEventLog* event_log_ RTC_GUARDED_BY(crit_);
RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogProxy);
};
class RtcpRttStatsProxy final : public RtcpRttStats {
public:
RtcpRttStatsProxy() : rtcp_rtt_stats_(nullptr) {}
void OnRttUpdate(int64_t rtt) override {
rtc::CritScope lock(&crit_);
if (rtcp_rtt_stats_)
rtcp_rtt_stats_->OnRttUpdate(rtt);
}
int64_t LastProcessedRtt() const override {
rtc::CritScope lock(&crit_);
if (!rtcp_rtt_stats_)
return 0;
return rtcp_rtt_stats_->LastProcessedRtt();
}
void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
rtc::CritScope lock(&crit_);
rtcp_rtt_stats_ = rtcp_rtt_stats;
}
private:
rtc::CriticalSection crit_;
RtcpRttStats* rtcp_rtt_stats_ RTC_GUARDED_BY(crit_);
RTC_DISALLOW_COPY_AND_ASSIGN(RtcpRttStatsProxy);
};
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);
}
std::vector<PacketFeedback> GetTransportFeedbackVector() const override {
RTC_NOTREACHED();
return std::vector<PacketFeedback>();
}
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);
}
}
private:
rtc::ThreadChecker thread_checker_;
rtc::CriticalSection crit_;
RtpPacketSender* rtp_packet_sender_ RTC_GUARDED_BY(&crit_);
};
class VoERtcpObserver : public RtcpBandwidthObserver {
public:
explicit VoERtcpObserver(Channel* owner)
: owner_(owner), bandwidth_observer_(nullptr) {}
virtual ~VoERtcpObserver() {}
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:
Channel* 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_);
};
class Channel::ProcessAndEncodeAudioTask : public rtc::QueuedTask {
public:
ProcessAndEncodeAudioTask(std::unique_ptr<AudioFrame> audio_frame,
Channel* channel)
: audio_frame_(std::move(audio_frame)), channel_(channel) {
RTC_DCHECK(channel_);
}
private:
bool Run() override {
RTC_DCHECK_RUN_ON(channel_->encoder_queue_);
channel_->ProcessAndEncodeAudioOnTaskQueue(audio_frame_.get());
return true;
}
std::unique_ptr<AudioFrame> audio_frame_;
Channel* const channel_;
};
int32_t Channel::SendData(FrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) {
RTC_DCHECK_RUN_ON(encoder_queue_);
if (_includeAudioLevelIndication) {
// Store current audio level in the RTP/RTCP module.
// The level will be used in combination with voice-activity state
// (frameType) to add an RTP header extension
_rtpRtcpModule->SetAudioLevel(rms_level_.Average());
}
// Push data from ACM to RTP/RTCP-module to deliver audio frame for
// packetization.
// This call will trigger Transport::SendPacket() from the RTP/RTCP module.
if (!_rtpRtcpModule->SendOutgoingData(
(FrameType&)frameType, payloadType, timeStamp,
// Leaving the time when this frame was
// received from the capture device as
// undefined for voice for now.
-1, payloadData, payloadSize, fragmentation, nullptr, nullptr)) {
LOG(LS_ERROR) <<
"Channel::SendData() failed to send data to RTP/RTCP module";
return -1;
}
return 0;
}
bool Channel::SendRtp(const uint8_t* data,
size_t len,
const PacketOptions& options) {
rtc::CritScope cs(&_callbackCritSect);
if (_transportPtr == NULL) {
LOG(LS_ERROR) << "Channel::SendPacket() failed to send RTP packet due to"
<< " invalid transport object";
return false;
}
uint8_t* bufferToSendPtr = (uint8_t*)data;
size_t bufferLength = len;
if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength, options)) {
LOG(LS_ERROR) << "Channel::SendPacket() RTP transmission failed";
return false;
}
return true;
}
bool Channel::SendRtcp(const uint8_t* data, size_t len) {
rtc::CritScope cs(&_callbackCritSect);
if (_transportPtr == NULL) {
LOG(LS_ERROR) << "Channel::SendRtcp() failed to send RTCP packet due to"
<< " invalid transport object";
return false;
}
uint8_t* bufferToSendPtr = (uint8_t*)data;
size_t bufferLength = len;
int n = _transportPtr->SendRtcp(bufferToSendPtr, bufferLength);
if (n < 0) {
LOG(LS_ERROR) << "Channel::SendRtcp() transmission failed";
return false;
}
return true;
}
void Channel::OnIncomingSSRCChanged(uint32_t ssrc) {
// Update ssrc so that NTP for AV sync can be updated.
_rtpRtcpModule->SetRemoteSSRC(ssrc);
}
void Channel::OnIncomingCSRCChanged(uint32_t CSRC, bool added) {
// TODO(saza): remove.
}
int32_t Channel::OnInitializeDecoder(int payload_type,
const SdpAudioFormat& audio_format,
uint32_t rate) {
if (!audio_coding_->RegisterReceiveCodec(payload_type, audio_format)) {
LOG(LS_WARNING) << "Channel::OnInitializeDecoder() invalid codec (pt="
<< payload_type << ", " << audio_format << ") received -1";
return -1;
}
return 0;
}
int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData,
size_t payloadSize,
const WebRtcRTPHeader* rtpHeader) {
if (!channel_state_.Get().playing) {
// Avoid inserting into NetEQ when we are not playing. Count the
// packet as discarded.
return 0;
}
// Push the incoming payload (parsed and ready for decoding) into the ACM
if (audio_coding_->IncomingPacket(payloadData, payloadSize, *rtpHeader) !=
0) {
LOG(LS_ERROR) <<
"Channel::OnReceivedPayloadData() unable to push data to the ACM";
return -1;
}
int64_t round_trip_time = 0;
_rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time, NULL, NULL,
NULL);
std::vector<uint16_t> nack_list = audio_coding_->GetNackList(round_trip_time);
if (!nack_list.empty()) {
// Can't use nack_list.data() since it's not supported by all
// compilers.
ResendPackets(&(nack_list[0]), static_cast<int>(nack_list.size()));
}
return 0;
}
bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
size_t rtp_packet_length) {
RTPHeader header;
if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
LOG(LS_WARNING) << "IncomingPacket invalid RTP header";
return false;
}
header.payload_type_frequency =
rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
if (header.payload_type_frequency < 0)
return false;
// TODO(nisse): Pass RtpPacketReceived with |recovered()| true.
return ReceivePacket(rtp_packet, rtp_packet_length, header);
}
AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame) {
audio_frame->sample_rate_hz_ = sample_rate_hz;
unsigned int ssrc;
RTC_CHECK_EQ(GetRemoteSSRC(ssrc), 0);
event_log_proxy_->Log(rtc::MakeUnique<RtcEventAudioPlayout>(ssrc));
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
bool muted;
if (audio_coding_->PlayoutData10Ms(audio_frame->sample_rate_hz_, audio_frame,
&muted) == -1) {
LOG(LS_ERROR) << "Channel::GetAudioFrame() PlayoutData10Ms() failed!";
// In all likelihood, the audio in this frame is garbage. We return an
// error so that the audio mixer module doesn't add it to the mix. As
// a result, it won't be played out and the actions skipped here are
// irrelevant.
return AudioMixer::Source::AudioFrameInfo::kError;
}
if (muted) {
// TODO(henrik.lundin): We should be able to do better than this. But we
// will have to go through all the cases below where the audio samples may
// be used, and handle the muted case in some way.
AudioFrameOperations::Mute(audio_frame);
}
// Store speech type for dead-or-alive detection
_outputSpeechType = audio_frame->speech_type_;
{
// Pass the audio buffers to an optional sink callback, before applying
// scaling/panning, as that applies to the mix operation.
// External recipients of the audio (e.g. via AudioTrack), will do their
// own mixing/dynamic processing.
rtc::CritScope cs(&_callbackCritSect);
if (audio_sink_) {
AudioSinkInterface::Data data(
audio_frame->data(), audio_frame->samples_per_channel_,
audio_frame->sample_rate_hz_, audio_frame->num_channels_,
audio_frame->timestamp_);
audio_sink_->OnData(data);
}
}
float output_gain = 1.0f;
{
rtc::CritScope cs(&volume_settings_critsect_);
output_gain = _outputGain;
}
// Output volume scaling
if (output_gain < 0.99f || output_gain > 1.01f) {
// TODO(solenberg): Combine with mute state - this can cause clicks!
AudioFrameOperations::ScaleWithSat(output_gain, audio_frame);
}
// Measure audio level (0-9)
// TODO(henrik.lundin) Use the |muted| information here too.
// TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| (see
// https://crbug.com/webrtc/7517).
_outputAudioLevel.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds);
if (capture_start_rtp_time_stamp_ < 0 && audio_frame->timestamp_ != 0) {
// The first frame with a valid rtp timestamp.
capture_start_rtp_time_stamp_ = audio_frame->timestamp_;
}
if (capture_start_rtp_time_stamp_ >= 0) {
// audio_frame.timestamp_ should be valid from now on.
// Compute elapsed time.
int64_t unwrap_timestamp =
rtp_ts_wraparound_handler_->Unwrap(audio_frame->timestamp_);
audio_frame->elapsed_time_ms_ =
(unwrap_timestamp - capture_start_rtp_time_stamp_) /
(GetRtpTimestampRateHz() / 1000);
{
rtc::CritScope lock(&ts_stats_lock_);
// Compute ntp time.
audio_frame->ntp_time_ms_ =
ntp_estimator_.Estimate(audio_frame->timestamp_);
// |ntp_time_ms_| won't be valid until at least 2 RTCP SRs are received.
if (audio_frame->ntp_time_ms_ > 0) {
// Compute |capture_start_ntp_time_ms_| so that
// |capture_start_ntp_time_ms_| + |elapsed_time_ms_| == |ntp_time_ms_|
capture_start_ntp_time_ms_ =
audio_frame->ntp_time_ms_ - audio_frame->elapsed_time_ms_;
}
}
}
return muted ? AudioMixer::Source::AudioFrameInfo::kMuted
: AudioMixer::Source::AudioFrameInfo::kNormal;
}
int Channel::PreferredSampleRate() const {
// Return the bigger of playout and receive frequency in the ACM.
return std::max(audio_coding_->ReceiveFrequency(),
audio_coding_->PlayoutFrequency());
}
int32_t Channel::CreateChannel(Channel*& channel,
int32_t channelId,
uint32_t instanceId,
const VoEBase::ChannelConfig& config) {
channel = new Channel(channelId, instanceId, config);
if (channel == NULL) {
LOG(LS_ERROR) << "unable to allocate memory for new channel";
return -1;
}
return 0;
}
Channel::Channel(int32_t channelId,
uint32_t instanceId,
const VoEBase::ChannelConfig& config)
: _instanceId(instanceId),
_channelId(channelId),
event_log_proxy_(new RtcEventLogProxy()),
rtcp_rtt_stats_proxy_(new RtcpRttStatsProxy()),
rtp_header_parser_(RtpHeaderParser::Create()),
rtp_payload_registry_(new RTPPayloadRegistry()),
rtp_receive_statistics_(
ReceiveStatistics::Create(Clock::GetRealTimeClock())),
rtp_receiver_(
RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
this,
this,
rtp_payload_registry_.get())),
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
_outputAudioLevel(),
_timeStamp(0), // This is just an offset, RTP module will add it's own
// random offset
ntp_estimator_(Clock::GetRealTimeClock()),
playout_timestamp_rtp_(0),
playout_delay_ms_(0),
send_sequence_number_(0),
rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
capture_start_rtp_time_stamp_(-1),
capture_start_ntp_time_ms_(-1),
_moduleProcessThreadPtr(NULL),
_audioDeviceModulePtr(NULL),
_transportPtr(NULL),
input_mute_(false),
previous_frame_muted_(false),
_outputGain(1.0f),
_includeAudioLevelIndication(false),
transport_overhead_per_packet_(0),
rtp_overhead_per_packet_(0),
_outputSpeechType(AudioFrame::kNormalSpeech),
rtcp_observer_(new VoERtcpObserver(this)),
associate_send_channel_(ChannelOwner(nullptr)),
pacing_enabled_(config.enable_voice_pacing),
feedback_observer_proxy_(new TransportFeedbackProxy()),
seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
kMaxRetransmissionWindowMs)),
decoder_factory_(config.acm_config.decoder_factory),
use_twcc_plr_for_ana_(
webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled") {
AudioCodingModule::Config acm_config(config.acm_config);
acm_config.neteq_config.enable_muted_state = true;
audio_coding_.reset(AudioCodingModule::Create(acm_config));
_outputAudioLevel.Clear();
RtpRtcp::Configuration configuration;
configuration.audio = true;
configuration.outgoing_transport = this;
configuration.overhead_observer = this;
configuration.receive_statistics = rtp_receive_statistics_.get();
configuration.bandwidth_callback = rtcp_observer_.get();
if (pacing_enabled_) {
configuration.paced_sender = rtp_packet_sender_proxy_.get();
configuration.transport_sequence_number_allocator =
seq_num_allocator_proxy_.get();
configuration.transport_feedback_callback = feedback_observer_proxy_.get();
}
configuration.event_log = &(*event_log_proxy_);
configuration.rtt_stats = &(*rtcp_rtt_stats_proxy_);
configuration.retransmission_rate_limiter =
retransmission_rate_limiter_.get();
_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
_rtpRtcpModule->SetSendingMediaStatus(false);
}
Channel::~Channel() {
RTC_DCHECK(!channel_state_.Get().sending);
RTC_DCHECK(!channel_state_.Get().playing);
}
int32_t Channel::Init() {
RTC_DCHECK(construction_thread_.CalledOnValidThread());
channel_state_.Reset();
// --- Initial sanity
if (_moduleProcessThreadPtr == NULL) {
LOG(LS_ERROR) << "Channel::Init() must call SetEngineInformation() first";
return -1;
}
// --- Add modules to process thread (for periodic schedulation)
_moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE);
// --- ACM initialization
if (audio_coding_->InitializeReceiver() == -1) {
LOG(LS_ERROR) << "Channel::Init() unable to initialize the ACM - 1";
return -1;
}
// --- RTP/RTCP module initialization
// 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.
telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
// RTCP is enabled by default.
_rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
// --- Register all permanent callbacks
if (audio_coding_->RegisterTransportCallback(this) == -1) {
LOG(LS_ERROR) << "Channel::Init() callbacks not registered";
return -1;
}
return 0;
}
void Channel::Terminate() {
RTC_DCHECK(construction_thread_.CalledOnValidThread());
// Must be called on the same thread as Init().
rtp_receive_statistics_->RegisterRtcpStatisticsCallback(NULL);
StopSend();
StopPlayout();
// The order to safely shutdown modules in a channel is:
// 1. De-register callbacks in modules
// 2. De-register modules in process thread
// 3. Destroy modules
if (audio_coding_->RegisterTransportCallback(NULL) == -1) {
LOG(LS_WARNING) << "Terminate() failed to de-register transport callback"
<< " (Audio coding module)";
}
// De-register modules in process thread
if (_moduleProcessThreadPtr)
_moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get());
// End of modules shutdown
}
int32_t Channel::SetEngineInformation(ProcessThread& moduleProcessThread,
AudioDeviceModule& audioDeviceModule,
rtc::TaskQueue* encoder_queue) {
RTC_DCHECK(encoder_queue);
RTC_DCHECK(!encoder_queue_);
_moduleProcessThreadPtr = &moduleProcessThread;
_audioDeviceModulePtr = &audioDeviceModule;
encoder_queue_ = encoder_queue;
return 0;
}
void Channel::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
rtc::CritScope cs(&_callbackCritSect);
audio_sink_ = std::move(sink);
}
const rtc::scoped_refptr<AudioDecoderFactory>&
Channel::GetAudioDecoderFactory() const {
return decoder_factory_;
}
int32_t Channel::StartPlayout() {
if (channel_state_.Get().playing) {
return 0;
}
channel_state_.SetPlaying(true);
return 0;
}
int32_t Channel::StopPlayout() {
if (!channel_state_.Get().playing) {
return 0;
}
channel_state_.SetPlaying(false);
_outputAudioLevel.Clear();
return 0;
}
int32_t Channel::StartSend() {
if (channel_state_.Get().sending) {
return 0;
}
channel_state_.SetSending(true);
{
// It is now OK to start posting tasks to the encoder task queue.
rtc::CritScope cs(&encoder_queue_lock_);
encoder_queue_is_active_ = true;
}
// Resume the previous sequence number which was reset by StopSend(). This
// needs to be done before |sending| is set to true on the RTP/RTCP module.
if (send_sequence_number_) {
_rtpRtcpModule->SetSequenceNumber(send_sequence_number_);
}
_rtpRtcpModule->SetSendingMediaStatus(true);
if (_rtpRtcpModule->SetSendingStatus(true) != 0) {
LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending";
_rtpRtcpModule->SetSendingMediaStatus(false);
rtc::CritScope cs(&_callbackCritSect);
channel_state_.SetSending(false);
return -1;
}
return 0;
}
void Channel::StopSend() {
if (!channel_state_.Get().sending) {
return;
}
channel_state_.SetSending(false);
// Post a task to the encoder thread which sets an event when the task is
// executed. We know that no more encoding tasks will be added to the task
// queue for this channel since sending is now deactivated. It means that,
// if we wait for the event to bet set, we know that no more pending tasks
// exists and it is therfore guaranteed that the task queue will never try
// to acccess and invalid channel object.
RTC_DCHECK(encoder_queue_);
rtc::Event flush(false, false);
{
// Clear |encoder_queue_is_active_| under lock to prevent any other tasks
// than this final "flush task" to be posted on the queue.
rtc::CritScope cs(&encoder_queue_lock_);
encoder_queue_is_active_ = false;
encoder_queue_->PostTask([&flush]() { flush.Set(); });
}
flush.Wait(rtc::Event::kForever);
// Store the sequence number to be able to pick up the same sequence for
// the next StartSend(). This is needed for restarting device, otherwise
// it might cause libSRTP to complain about packets being replayed.
// TODO(xians): Remove this workaround after RtpRtcpModule's refactoring
// CL is landed. See issue
// https://code.google.com/p/webrtc/issues/detail?id=2111 .
send_sequence_number_ = _rtpRtcpModule->SequenceNumber();
// Reset sending SSRC and sequence number and triggers direct transmission
// of RTCP BYE
if (_rtpRtcpModule->SetSendingStatus(false) == -1) {
LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending";
}
_rtpRtcpModule->SetSendingMediaStatus(false);
}
bool Channel::SetEncoder(int payload_type,
std::unique_ptr<AudioEncoder> encoder) {
RTC_DCHECK_GE(payload_type, 0);
RTC_DCHECK_LE(payload_type, 127);
// TODO(ossu): Make CodecInsts up, for now: one for the RTP/RTCP module and
// one for for us to keep track of sample rate and number of channels, etc.
// 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.
CodecInst rtp_codec;
rtp_codec.pltype = payload_type;
strncpy(rtp_codec.plname, "audio", sizeof(rtp_codec.plname));
rtp_codec.plname[sizeof(rtp_codec.plname) - 1] = 0;
// Seems unclear if it should be clock rate or sample rate. CodecInst
// supposedly carries the sample rate, but only clock rate seems sensible to
// send to the RTP/RTCP module.
rtp_codec.plfreq = encoder->RtpTimestampRateHz();
rtp_codec.pacsize = rtc::CheckedDivExact(
static_cast<int>(encoder->Max10MsFramesInAPacket() * rtp_codec.plfreq),
100);
rtp_codec.channels = encoder->NumChannels();
rtp_codec.rate = 0;
// For audio encoding we need, instead, the actual sample rate of the codec.
// The rest of the information should be the same.
CodecInst send_codec = rtp_codec;
send_codec.plfreq = encoder->SampleRateHz();
cached_send_codec_.emplace(send_codec);
if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) {
_rtpRtcpModule->DeRegisterSendPayload(payload_type);
if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) {
LOG(LS_ERROR)
<< "SetEncoder() failed to register codec to RTP/RTCP module";
return false;
}
}
audio_coding_->SetEncoder(std::move(encoder));
codec_manager_.UnsetCodecInst();
return true;
}
void Channel::ModifyEncoder(
rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
audio_coding_->ModifyEncoder(modifier);
}
int32_t Channel::GetSendCodec(CodecInst& codec) {
if (cached_send_codec_) {
codec = *cached_send_codec_;
return 0;
} else {
const CodecInst* send_codec = codec_manager_.GetCodecInst();
if (send_codec) {
codec = *send_codec;
return 0;
}
}
return -1;
}
int32_t Channel::GetRecCodec(CodecInst& codec) {
return (audio_coding_->ReceiveCodec(&codec));
}
int32_t Channel::SetSendCodec(const CodecInst& codec) {
if (!codec_manager_.RegisterEncoder(codec) ||
!codec_manager_.MakeEncoder(&rent_a_codec_, audio_coding_.get())) {
LOG(LS_ERROR) << "SetSendCodec() failed to register codec to ACM";
return -1;
}
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
_rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
LOG(LS_ERROR)
<< "SetSendCodec() failed to register codec to RTP/RTCP module";
return -1;
}
}
cached_send_codec_.reset();
return 0;
}
void Channel::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) {
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder) {
(*encoder)->OnReceivedUplinkBandwidth(
bitrate_bps, rtc::Optional<int64_t>(probing_interval_ms));
}
});
retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
}
void Channel::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {
if (!use_twcc_plr_for_ana_)
return;
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder) {
(*encoder)->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
}
});
}
void Channel::OnRecoverableUplinkPacketLossRate(
float recoverable_packet_loss_rate) {
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder) {
(*encoder)->OnReceivedUplinkRecoverablePacketLossFraction(
recoverable_packet_loss_rate);
}
});
}
void Channel::OnUplinkPacketLossRate(float packet_loss_rate) {
if (use_twcc_plr_for_ana_)
return;
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder) {
(*encoder)->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
}
});
}
void Channel::SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
rtp_payload_registry_->SetAudioReceivePayloads(codecs);
audio_coding_->SetReceiveCodecs(codecs);
}
bool Channel::EnableAudioNetworkAdaptor(const std::string& config_string) {
bool success = false;
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder) {
success = (*encoder)->EnableAudioNetworkAdaptor(config_string,
event_log_proxy_.get());
}
});
return success;
}
void Channel::DisableAudioNetworkAdaptor() {
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder)
(*encoder)->DisableAudioNetworkAdaptor();
});
}
void Channel::SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) {
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder) {
(*encoder)->SetReceiverFrameLengthRange(min_frame_length_ms,
max_frame_length_ms);
}
});
}
void Channel::RegisterTransport(Transport* transport) {
rtc::CritScope cs(&_callbackCritSect);
_transportPtr = transport;
}
void Channel::OnRtpPacket(const RtpPacketReceived& packet) {
RTPHeader header;
packet.GetHeader(&header);
// Store playout timestamp for the received RTP packet
UpdatePlayoutTimestamp(false);
header.payload_type_frequency =
rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
if (header.payload_type_frequency >= 0) {
bool in_order = IsPacketInOrder(header);
rtp_receive_statistics_->IncomingPacket(
header, packet.size(), IsPacketRetransmitted(header, in_order));
rtp_payload_registry_->SetIncomingPayloadType(header);
ReceivePacket(packet.data(), packet.size(), header);
}
}
bool Channel::ReceivePacket(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header) {
const uint8_t* payload = packet + header.headerLength;
assert(packet_length >= header.headerLength);
size_t payload_length = packet_length - header.headerLength;
const auto pl =
rtp_payload_registry_->PayloadTypeToPayload(header.payloadType);
if (!pl) {
return false;
}
return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
pl->typeSpecific);
}
bool Channel::IsPacketInOrder(const RTPHeader& header) const {
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(header.ssrc);
if (!statistician)
return false;
return statistician->IsPacketInOrder(header.sequenceNumber);
}
bool Channel::IsPacketRetransmitted(const RTPHeader& header,
bool in_order) const {
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(header.ssrc);
if (!statistician)
return false;
// Check if this is a retransmission.
int64_t min_rtt = 0;
_rtpRtcpModule->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
return !in_order && statistician->IsRetransmitOfOldPacket(header, min_rtt);
}
int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
// Store playout timestamp for the received RTCP packet
UpdatePlayoutTimestamp(true);
// Deliver RTCP packet to RTP/RTCP module for parsing
_rtpRtcpModule->IncomingRtcpPacket(data, length);
int64_t rtt = GetRTT(true);
if (rtt == 0) {
// Waiting for valid RTT.
return 0;
}
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);
// Invoke audio encoders OnReceivedRtt().
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder)
(*encoder)->OnReceivedRtt(rtt);
});
uint32_t ntp_secs = 0;
uint32_t ntp_frac = 0;
uint32_t rtp_timestamp = 0;
if (0 !=
_rtpRtcpModule->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL,
&rtp_timestamp)) {
// Waiting for RTCP.
return 0;
}
{
rtc::CritScope lock(&ts_stats_lock_);
ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
}
return 0;
}
int Channel::GetSpeechOutputLevel() const {
return _outputAudioLevel.Level();
}
int Channel::GetSpeechOutputLevelFullRange() const {
return _outputAudioLevel.LevelFullRange();
}
double Channel::GetTotalOutputEnergy() const {
return _outputAudioLevel.TotalEnergy();
}
double Channel::GetTotalOutputDuration() const {
return _outputAudioLevel.TotalDuration();
}
void Channel::SetInputMute(bool enable) {
rtc::CritScope cs(&volume_settings_critsect_);
input_mute_ = enable;
}
bool Channel::InputMute() const {
rtc::CritScope cs(&volume_settings_critsect_);
return input_mute_;
}
void Channel::SetChannelOutputVolumeScaling(float scaling) {
rtc::CritScope cs(&volume_settings_critsect_);
_outputGain = scaling;
}
int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
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 -1;
}
if (_rtpRtcpModule->SendTelephoneEventOutband(
event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
LOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event";
return -1;
}
return 0;
}
int Channel::SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) {
RTC_DCHECK_LE(0, payload_type);
RTC_DCHECK_GE(127, payload_type);
CodecInst codec = {0};
codec.pltype = payload_type;
codec.plfreq = payload_frequency;
memcpy(codec.plname, "telephone-event", 16);
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
_rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
LOG(LS_ERROR) << "SetSendTelephoneEventPayloadType() failed to register "
"send payload type";
return -1;
}
}
return 0;
}
int Channel::SetLocalSSRC(unsigned int ssrc) {
if (channel_state_.Get().sending) {
LOG(LS_ERROR) << "SetLocalSSRC() already sending";
return -1;
}
_rtpRtcpModule->SetSSRC(ssrc);
return 0;
}
int Channel::GetRemoteSSRC(unsigned int& ssrc) {
ssrc = rtp_receiver_->SSRC();
return 0;
}
int Channel::SetSendAudioLevelIndicationStatus(bool enable, unsigned char id) {
_includeAudioLevelIndication = enable;
return SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
}
int Channel::SetReceiveAudioLevelIndicationStatus(bool enable,
unsigned char id) {
rtp_header_parser_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel);
if (enable &&
!rtp_header_parser_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
id)) {
return -1;
}
return 0;
}
void Channel::EnableSendTransportSequenceNumber(int id) {
int ret =
SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id);
RTC_DCHECK_EQ(0, ret);
}
void Channel::EnableReceiveTransportSequenceNumber(int id) {
rtp_header_parser_->DeregisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber);
bool ret = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber, id);
RTC_DCHECK(ret);
}
void Channel::RegisterSenderCongestionControlObjects(
RtpTransportControllerSendInterface* transport,
RtcpBandwidthObserver* bandwidth_observer) {
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 Channel::RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) {
RTC_DCHECK(packet_router);
RTC_DCHECK(!packet_router_);
constexpr bool remb_candidate = false;
packet_router->AddReceiveRtpModule(_rtpRtcpModule.get(), remb_candidate);
packet_router_ = packet_router;
}
void Channel::ResetSenderCongestionControlObjects() {
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 Channel::ResetReceiverCongestionControlObjects() {
RTC_DCHECK(packet_router_);
packet_router_->RemoveReceiveRtpModule(_rtpRtcpModule.get());
packet_router_ = nullptr;
}
void Channel::SetRTCPStatus(bool enable) {
_rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff);
}
int Channel::SetRTCP_CNAME(const char cName[256]) {
if (_rtpRtcpModule->SetCNAME(cName) != 0) {
LOG(LS_ERROR) << "SetRTCP_CNAME() failed to set RTCP CNAME";
return -1;
}
return 0;
}
int Channel::GetRemoteRTCPReportBlocks(
std::vector<ReportBlock>* report_blocks) {
if (report_blocks == NULL) {
LOG(LS_ERROR) << "GetRemoteRTCPReportBlock()s invalid report_blocks.";
return -1;
}
// 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;
if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks) != 0) {
return -1;
}
if (rtcp_report_blocks.empty())
return 0;
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 0;
}
int Channel::GetRTPStatistics(CallStatistics& stats) {
// --- RtcpStatistics
// The jitter statistics is updated for each received RTP packet and is
// based on received packets.
RtcpStatistics statistics;
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(rtp_receiver_->SSRC());
if (statistician) {
statistician->GetStatistics(&statistics,
_rtpRtcpModule->RTCP() == RtcpMode::kOff);
}
stats.fractionLost = statistics.fraction_lost;
stats.cumulativeLost = statistics.packets_lost;
stats.extendedMax = statistics.extended_highest_sequence_number;
stats.jitterSamples = statistics.jitter;
// --- RTT
stats.rttMs = GetRTT(true);
// --- Data counters
size_t bytesSent(0);
uint32_t packetsSent(0);
size_t bytesReceived(0);
uint32_t packetsReceived(0);
if (statistician) {
statistician->GetDataCounters(&bytesReceived, &packetsReceived);
}
if (_rtpRtcpModule->DataCountersRTP(&bytesSent, &packetsSent) != 0) {
LOG(LS_WARNING) << "GetRTPStatistics() failed to retrieve RTP datacounters"
<< " => output will not be complete";
}
stats.bytesSent = bytesSent;
stats.packetsSent = packetsSent;
stats.bytesReceived = bytesReceived;
stats.packetsReceived = packetsReceived;
// --- Timestamps
{
rtc::CritScope lock(&ts_stats_lock_);
stats.capture_start_ntp_time_ms_ = capture_start_ntp_time_ms_;
}
return 0;
}
void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) {
// None of these functions can fail.
// If pacing is enabled we always store packets.
if (!pacing_enabled_)
_rtpRtcpModule->SetStorePacketsStatus(enable, maxNumberOfPackets);
rtp_receive_statistics_->SetMaxReorderingThreshold(maxNumberOfPackets);
if (enable)
audio_coding_->EnableNack(maxNumberOfPackets);
else
audio_coding_->DisableNack();
}
// Called when we are missing one or more packets.
int Channel::ResendPackets(const uint16_t* sequence_numbers, int length) {
return _rtpRtcpModule->SendNACK(sequence_numbers, length);
}
void Channel::ProcessAndEncodeAudio(const AudioFrame& audio_input) {
// Avoid posting any new tasks if sending was already stopped in StopSend().
rtc::CritScope cs(&encoder_queue_lock_);
if (!encoder_queue_is_active_) {
return;
}
std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
// TODO(henrika): try to avoid copying by moving ownership of audio frame
// either into pool of frames or into the task itself.
audio_frame->CopyFrom(audio_input);
// 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(std::unique_ptr<rtc::QueuedTask>(
new ProcessAndEncodeAudioTask(std::move(audio_frame), this)));
}
void Channel::ProcessAndEncodeAudio(const int16_t* audio_data,
int sample_rate,
size_t number_of_frames,
size_t number_of_channels) {
// Avoid posting as new task if sending was already stopped in StopSend().
rtc::CritScope cs(&encoder_queue_lock_);
if (!encoder_queue_is_active_) {
return;
}
CodecInst codec;
const int result = GetSendCodec(codec);
std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
// TODO(ossu): Investigate how this could happen. b/62909493
if (result == 0) {
audio_frame->sample_rate_hz_ = std::min(codec.plfreq, sample_rate);
audio_frame->num_channels_ = std::min(number_of_channels, codec.channels);
} else {
audio_frame->sample_rate_hz_ = sample_rate;
audio_frame->num_channels_ = number_of_channels;
LOG(LS_WARNING) << "Unable to get send codec for channel " << ChannelId();
RTC_NOTREACHED();
}
RemixAndResample(audio_data, number_of_frames, number_of_channels,
sample_rate, &input_resampler_, audio_frame.get());
encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
new ProcessAndEncodeAudioTask(std::move(audio_frame), this)));
}
void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
RTC_DCHECK_RUN_ON(encoder_queue_);
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) {
LOG(LS_ERROR) << "ACM::Add10MsData() failed for channel " << _channelId;
return;
}
_timeStamp += static_cast<uint32_t>(audio_input->samples_per_channel_);
}
void Channel::set_associate_send_channel(const ChannelOwner& channel) {
RTC_DCHECK(!channel.channel() ||
channel.channel()->ChannelId() != _channelId);
rtc::CritScope lock(&assoc_send_channel_lock_);
associate_send_channel_ = channel;
}
void Channel::DisassociateSendChannel(int channel_id) {
rtc::CritScope lock(&assoc_send_channel_lock_);
Channel* channel = associate_send_channel_.channel();
if (channel && channel->ChannelId() == channel_id) {
// If this channel is associated with a send channel of the specified
// Channel ID, disassociate with it.
ChannelOwner ref(NULL);
associate_send_channel_ = ref;
}
}
void Channel::SetRtcEventLog(RtcEventLog* event_log) {
event_log_proxy_->SetEventLog(event_log);
}
void Channel::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
rtcp_rtt_stats_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
}
void Channel::UpdateOverheadForEncoder() {
size_t overhead_per_packet =
transport_overhead_per_packet_ + rtp_overhead_per_packet_;
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder) {
(*encoder)->OnReceivedOverhead(overhead_per_packet);
}
});
}
void Channel::SetTransportOverhead(size_t transport_overhead_per_packet) {
rtc::CritScope cs(&overhead_per_packet_lock_);
transport_overhead_per_packet_ = transport_overhead_per_packet;
UpdateOverheadForEncoder();
}
// TODO(solenberg): Make AudioSendStream an OverheadObserver instead.
void Channel::OnOverheadChanged(size_t overhead_bytes_per_packet) {
rtc::CritScope cs(&overhead_per_packet_lock_);
rtp_overhead_per_packet_ = overhead_bytes_per_packet;
UpdateOverheadForEncoder();
}
int Channel::GetNetworkStatistics(NetworkStatistics& stats) {
return audio_coding_->GetNetworkStatistics(&stats);
}
void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
audio_coding_->GetDecodingCallStatistics(stats);
}
ANAStats Channel::GetANAStatistics() const {
return audio_coding_->GetANAStats();
}
uint32_t Channel::GetDelayEstimate() const {
rtc::CritScope lock(&video_sync_lock_);
return audio_coding_->FilteredCurrentDelayMs() + playout_delay_ms_;
}
int Channel::SetMinimumPlayoutDelay(int delayMs) {
if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
(delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) {
LOG(LS_ERROR) << "SetMinimumPlayoutDelay() invalid min delay";
return -1;
}
if (audio_coding_->SetMinimumPlayoutDelay(delayMs) != 0) {
LOG(LS_ERROR) << "SetMinimumPlayoutDelay() failed to set min playout delay";
return -1;
}
return 0;
}
int Channel::GetPlayoutTimestamp(unsigned int& timestamp) {
uint32_t playout_timestamp_rtp = 0;
{
rtc::CritScope lock(&video_sync_lock_);
playout_timestamp_rtp = playout_timestamp_rtp_;
}
if (playout_timestamp_rtp == 0) {
LOG(LS_ERROR) << "GetPlayoutTimestamp() failed to retrieve timestamp";
return -1;
}
timestamp = playout_timestamp_rtp;
return 0;
}
int Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule,
RtpReceiver** rtp_receiver) const {
*rtpRtcpModule = _rtpRtcpModule.get();
*rtp_receiver = rtp_receiver_.get();
return 0;
}
void Channel::UpdatePlayoutTimestamp(bool rtcp) {
jitter_buffer_playout_timestamp_ = audio_coding_->PlayoutTimestamp();
if (!jitter_buffer_playout_timestamp_) {
// This can happen if this channel has not received any RTP packets. In
// this case, NetEq is not capable of computing a playout timestamp.
return;
}
uint16_t delay_ms = 0;
if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) {
LOG(LS_WARNING) << "Channel::UpdatePlayoutTimestamp() failed to read"
<< " playout delay from the ADM";
return;
}
RTC_DCHECK(jitter_buffer_playout_timestamp_);
uint32_t playout_timestamp = *jitter_buffer_playout_timestamp_;
// Remove the playout delay.
playout_timestamp -= (delay_ms * (GetRtpTimestampRateHz() / 1000));
{
rtc::CritScope lock(&video_sync_lock_);
if (!rtcp) {
playout_timestamp_rtp_ = playout_timestamp;
}
playout_delay_ms_ = delay_ms;
}
}
void Channel::RegisterReceiveCodecsToRTPModule() {
// TODO(kwiberg): Iterate over the factory's supported codecs instead?
const int nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
for (int idx = 0; idx < nSupportedCodecs; idx++) {
CodecInst codec;
if (audio_coding_->Codec(idx, &codec) == -1) {
LOG(LS_WARNING) << "Unable to register codec #" << idx
<< " for RTP/RTCP receiver.";
continue;
}
const SdpAudioFormat format = CodecInstToSdp(codec);
if (!decoder_factory_->IsSupportedDecoder(format) ||
rtp_receiver_->RegisterReceivePayload(codec.pltype, format) == -1) {
LOG(LS_WARNING) << "Unable to register " << format
<< " for RTP/RTCP receiver.";
}
}
}
int Channel::SetSendRtpHeaderExtension(bool enable,
RTPExtensionType type,
unsigned char id) {
int error = 0;
_rtpRtcpModule->DeregisterSendRtpHeaderExtension(type);
if (enable) {
error = _rtpRtcpModule->RegisterSendRtpHeaderExtension(type, id);
}
return error;
}
int Channel::GetRtpTimestampRateHz() const {
const auto format = audio_coding_->ReceiveFormat();
// Default to the playout frequency if we've not gotten any packets yet.
// TODO(ossu): Zero clockrate can only happen if we've added an external
// decoder for a format we don't support internally. Remove once that way of
// adding decoders is gone!
return (format && format->clockrate_hz != 0)
? format->clockrate_hz
: audio_coding_->PlayoutFrequency();
}
int64_t Channel::GetRTT(bool allow_associate_channel) const {
RtcpMode method = _rtpRtcpModule->RTCP();
if (method == RtcpMode::kOff) {
return 0;
}
std::vector<RTCPReportBlock> report_blocks;
_rtpRtcpModule->RemoteRTCPStat(&report_blocks);
int64_t rtt = 0;
if (report_blocks.empty()) {
if (allow_associate_channel) {
rtc::CritScope lock(&assoc_send_channel_lock_);
Channel* channel = associate_send_channel_.channel();
// Tries to get RTT from an associated channel. This is important for
// receive-only channels.
if (channel) {
// To prevent infinite recursion and deadlock, calling GetRTT of
// associate channel should always use "false" for argument:
// |allow_associate_channel|.
rtt = channel->GetRTT(false);
}
}
return rtt;
}
uint32_t remoteSSRC = rtp_receiver_->SSRC();
std::vector<RTCPReportBlock>::const_iterator it = report_blocks.begin();
for (; it != report_blocks.end(); ++it) {
if (it->sender_ssrc == remoteSSRC)
break;
}
if (it == report_blocks.end()) {
// We have not received packets with SSRC matching the report blocks.
// To calculate RTT we try with the SSRC of the first report block.
// This is very important for send-only channels where we don't know
// the SSRC of the other end.
remoteSSRC = report_blocks[0].sender_ssrc;
}
int64_t avg_rtt = 0;
int64_t max_rtt = 0;
int64_t min_rtt = 0;
if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
0) {
return 0;
}
return rtt;
}
} // namespace voe
} // namespace webrtc
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