Google Git
Sign in
webrtc / src.git / 2666cf7eba4bdd697d59d0451a8f74a05d4d207e / . / voice_engine / channel.cc
blob: 43088b696af4e454bf691f30e1f460603f9f4990 [file] [log] [blame]
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "voice_engine/channel.h"
#include <algorithm>
#include <utility>
#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 "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/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 "system_wrappers/include/trace.h"
#include "voice_engine/statistics.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(const std::string& file_name,
int64_t max_size_bytes) override {
RTC_NOTREACHED();
return false;
}
bool StartLogging(rtc::PlatformFile log_file,
int64_t max_size_bytes) override {
RTC_NOTREACHED();
return false;
}
void StopLogging() override { RTC_NOTREACHED(); }
void LogVideoReceiveStreamConfig(
const webrtc::rtclog::StreamConfig&) override {
RTC_NOTREACHED();
}
void LogVideoSendStreamConfig(const webrtc::rtclog::StreamConfig&) override {
RTC_NOTREACHED();
}
void LogAudioReceiveStreamConfig(
const webrtc::rtclog::StreamConfig& config) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogAudioReceiveStreamConfig(config);
}
}
void LogAudioSendStreamConfig(
const webrtc::rtclog::StreamConfig& config) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogAudioSendStreamConfig(config);
}
}
void LogRtpHeader(webrtc::PacketDirection direction,
const uint8_t* header,
size_t packet_length) override {
LogRtpHeader(direction, header, packet_length, PacedPacketInfo::kNotAProbe);
}
void LogRtpHeader(webrtc::PacketDirection direction,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogRtpHeader(direction, header, packet_length,
probe_cluster_id);
}
}
void LogRtcpPacket(webrtc::PacketDirection direction,
const uint8_t* packet,
size_t length) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogRtcpPacket(direction, packet, length);
}
}
void LogAudioPlayout(uint32_t ssrc) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogAudioPlayout(ssrc);
}
}
void LogLossBasedBweUpdate(int32_t bitrate_bps,
uint8_t fraction_loss,
int32_t total_packets) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogLossBasedBweUpdate(bitrate_bps, fraction_loss,
total_packets);
}
}
void LogDelayBasedBweUpdate(int32_t bitrate_bps,
BandwidthUsage detector_state) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogDelayBasedBweUpdate(bitrate_bps, detector_state);
}
}
void LogAudioNetworkAdaptation(
const AudioEncoderRuntimeConfig& config) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogAudioNetworkAdaptation(config);
}
}
void LogProbeClusterCreated(int id,
int bitrate_bps,
int min_probes,
int min_bytes) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogProbeClusterCreated(id, bitrate_bps, min_probes,
min_bytes);
}
};
void LogProbeResultSuccess(int id, int bitrate_bps) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogProbeResultSuccess(id, bitrate_bps);
}
};
void LogProbeResultFailure(int id,
ProbeFailureReason failure_reason) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogProbeResultFailure(id, failure_reason);
}
};
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_);
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendData(frameType=%u, payloadType=%u, timeStamp=%u,"
" payloadSize=%" PRIuS ", fragmentation=0x%x)",
frameType, payloadType, timeStamp, payloadSize, fragmentation);
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)) {
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
"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) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendPacket(channel=%d, len=%" PRIuS ")", len);
rtc::CritScope cs(&_callbackCritSect);
if (_transportPtr == NULL) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"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)) {
std::string transport_name =
_externalTransport ? "external transport" : "WebRtc sockets";
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendPacket() RTP transmission using %s failed",
transport_name.c_str());
return false;
}
return true;
}
bool Channel::SendRtcp(const uint8_t* data, size_t len) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendRtcp(len=%" PRIuS ")", len);
rtc::CritScope cs(&_callbackCritSect);
if (_transportPtr == NULL) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"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) {
std::string transport_name =
_externalTransport ? "external transport" : "WebRtc sockets";
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendRtcp() transmission using %s failed",
transport_name.c_str());
return false;
}
return true;
}
void Channel::OnIncomingSSRCChanged(uint32_t ssrc) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnIncomingSSRCChanged(SSRC=%d)", ssrc);
// Update ssrc so that NTP for AV sync can be updated.
_rtpRtcpModule->SetRemoteSSRC(ssrc);
}
void Channel::OnIncomingCSRCChanged(uint32_t CSRC, bool added) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnIncomingCSRCChanged(CSRC=%d, added=%d)", CSRC,
added);
}
int32_t Channel::OnInitializeDecoder(
int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
int frequency,
size_t channels,
uint32_t rate) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnInitializeDecoder(payloadType=%d, "
"payloadName=%s, frequency=%u, channels=%" PRIuS ", rate=%u)",
payloadType, payloadName, frequency, channels, rate);
CodecInst receiveCodec = {0};
CodecInst dummyCodec = {0};
receiveCodec.pltype = payloadType;
receiveCodec.plfreq = frequency;
receiveCodec.channels = channels;
receiveCodec.rate = rate;
strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
audio_coding_->Codec(payloadName, &dummyCodec, frequency, channels);
receiveCodec.pacsize = dummyCodec.pacsize;
// Register the new codec to the ACM
if (!audio_coding_->RegisterReceiveCodec(receiveCodec.pltype,
CodecInstToSdp(receiveCodec))) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnInitializeDecoder() invalid codec ("
"pt=%d, name=%s) received - 1",
payloadType, payloadName);
_engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR);
return -1;
}
return 0;
}
int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData,
size_t payloadSize,
const WebRtcRTPHeader* rtpHeader) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnReceivedPayloadData(payloadSize=%" PRIuS
","
" payloadType=%u, audioChannel=%" PRIuS ")",
payloadSize, rtpHeader->header.payloadType,
rtpHeader->type.Audio.channel);
if (!channel_state_.Get().playing) {
// Avoid inserting into NetEQ when we are not playing. Count the
// packet as discarded.
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"received packet is discarded since playing is not"
" activated");
return 0;
}
// Push the incoming payload (parsed and ready for decoding) into the ACM
if (audio_coding_->IncomingPacket(payloadData, payloadSize, *rtpHeader) !=
0) {
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
"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)) {
WEBRTC_TRACE(kTraceDebug, webrtc::kTraceVoice, _channelId,
"IncomingPacket invalid RTP header");
return false;
}
header.payload_type_frequency =
rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
if (header.payload_type_frequency < 0)
return false;
return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
}
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_->LogAudioPlayout(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) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"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) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
instanceId);
channel = new Channel(channelId, instanceId, config);
if (channel == NULL) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel() unable to allocate memory for"
" 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(),
_externalTransport(false),
_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),
_engineStatisticsPtr(NULL),
_moduleProcessThreadPtr(NULL),
_audioDeviceModulePtr(NULL),
_callbackCritSectPtr(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") {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::Channel() - ctor");
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());
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::Init()");
channel_state_.Reset();
// --- Initial sanity
if ((_engineStatisticsPtr == NULL) || (_moduleProcessThreadPtr == NULL)) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"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) {
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"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) {
_engineStatisticsPtr->SetLastError(
VE_CANNOT_INIT_CHANNEL, kTraceError,
"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().
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::Terminate");
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) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"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(Statistics& engineStatistics,
ProcessThread& moduleProcessThread,
AudioDeviceModule& audioDeviceModule,
rtc::CriticalSection* callbackCritSect,
rtc::TaskQueue* encoder_queue) {
RTC_DCHECK(encoder_queue);
RTC_DCHECK(!encoder_queue_);
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetEngineInformation()");
_engineStatisticsPtr = &engineStatistics;
_moduleProcessThreadPtr = &moduleProcessThread;
_audioDeviceModulePtr = &audioDeviceModule;
_callbackCritSectPtr = callbackCritSect;
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() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StartPlayout()");
if (channel_state_.Get().playing) {
return 0;
}
channel_state_.SetPlaying(true);
return 0;
}
int32_t Channel::StopPlayout() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StopPlayout()");
if (!channel_state_.Get().playing) {
return 0;
}
channel_state_.SetPlaying(false);
_outputAudioLevel.Clear();
return 0;
}
int32_t Channel::StartSend() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"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) {
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"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() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"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) {
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
"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) {
WEBRTC_TRACE(
kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"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) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetSendCodec()");
if (!codec_manager_.RegisterEncoder(codec) ||
!codec_manager_.MakeEncoder(&rent_a_codec_, audio_coding_.get())) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"SetSendCodec() failed to register codec to ACM");
return -1;
}
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
_rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"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) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps);
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);
}
});
}
int32_t Channel::RegisterExternalTransport(Transport* transport) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterExternalTransport()");
rtc::CritScope cs(&_callbackCritSect);
if (_externalTransport) {
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"RegisterExternalTransport() external transport already enabled");
return -1;
}
_externalTransport = true;
_transportPtr = transport;
return 0;
}
int32_t Channel::DeRegisterExternalTransport() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::DeRegisterExternalTransport()");
rtc::CritScope cs(&_callbackCritSect);
if (_transportPtr) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"DeRegisterExternalTransport() all transport is disabled");
} else {
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"DeRegisterExternalTransport() external transport already "
"disabled");
}
_externalTransport = false;
_transportPtr = NULL;
return 0;
}
void Channel::OnRtpPacket(const RtpPacketReceived& packet) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnRtpPacket()");
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, in_order);
}
}
bool Channel::ReceivePacket(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header,
bool in_order) {
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, in_order);
}
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) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::ReceivedRTCPPacket()");
// 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) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendTelephoneEventOutband(...)");
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) {
_engineStatisticsPtr->SetLastError(
VE_SEND_DTMF_FAILED, kTraceWarning,
"SendTelephoneEventOutband() failed to send event");
return -1;
}
return 0;
}
int Channel::SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetSendTelephoneEventPayloadType()");
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) {
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetSendTelephoneEventPayloadType() failed to register send"
"payload type");
return -1;
}
}
return 0;
}
int Channel::SetLocalSSRC(unsigned int ssrc) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetLocalSSRC()");
if (channel_state_.Get().sending) {
_engineStatisticsPtr->SetLastError(VE_ALREADY_SENDING, kTraceError,
"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) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetRTCPStatus()");
_rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff);
}
int Channel::SetRTCP_CNAME(const char cName[256]) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetRTCP_CNAME()");
if (_rtpRtcpModule->SetCNAME(cName) != 0) {
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetRTCP_CNAME() failed to set RTCP CNAME");
return -1;
}
return 0;
}
int Channel::GetRemoteRTCPReportBlocks(
std::vector<ReportBlock>* report_blocks) {
if (report_blocks == NULL) {
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"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) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"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) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetMinimumPlayoutDelay()");
if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
(delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) {
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetMinimumPlayoutDelay() invalid min delay");
return -1;
}
if (audio_coding_->SetMinimumPlayoutDelay(delayMs) != 0) {
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"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) {
_engineStatisticsPtr->SetLastError(
VE_CANNOT_RETRIEVE_VALUE, kTraceStateInfo,
"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) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::UpdatePlayoutTimestamp() failed to read playout"
" delay from the ADM");
_engineStatisticsPtr->SetLastError(
VE_CANNOT_RETRIEVE_VALUE, kTraceError,
"UpdatePlayoutTimestamp() failed to retrieve playout delay");
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));
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu",
playout_timestamp);
{
rtc::CritScope lock(&video_sync_lock_);
if (!rtcp) {
playout_timestamp_rtp_ = playout_timestamp;
}
playout_delay_ms_ = delay_ms;
}
}
void Channel::RegisterReceiveCodecsToRTPModule() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterReceiveCodecsToRTPModule()");
CodecInst codec;
const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
for (int idx = 0; idx < nSupportedCodecs; idx++) {
// Open up the RTP/RTCP receiver for all supported codecs
if ((audio_coding_->Codec(idx, &codec) == -1) ||
(rtp_receiver_->RegisterReceivePayload(codec) == -1)) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterReceiveCodecsToRTPModule() unable"
" to register %s (%d/%d/%" PRIuS
"/%d) to RTP/RTCP "
"receiver",
codec.plname, codec.pltype, codec.plfreq, codec.channels,
codec.rate);
} else {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterReceiveCodecsToRTPModule() %s "
"(%d/%d/%" PRIuS
"/%d) has been added to the RTP/RTCP "
"receiver",
codec.plname, codec.pltype, codec.plfreq, codec.channels,
codec.rate);
}
}
}
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