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/*
* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "audio/voip/audio_channel.h"
#include <utility>
#include <vector>
#include "api/audio_codecs/audio_format.h"
#include "api/task_queue/task_queue_factory.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace {
constexpr int kRtcpReportIntervalMs = 5000;
} // namespace
AudioChannel::AudioChannel(
Transport* transport,
uint32_t local_ssrc,
TaskQueueFactory* task_queue_factory,
AudioMixer* audio_mixer,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
: audio_mixer_(audio_mixer) {
RTC_DCHECK(task_queue_factory);
RTC_DCHECK(audio_mixer);
Clock* clock = Clock::GetRealTimeClock();
receive_statistics_ = ReceiveStatistics::Create(clock);
RtpRtcpInterface::Configuration rtp_config;
rtp_config.clock = clock;
rtp_config.audio = true;
rtp_config.receive_statistics = receive_statistics_.get();
rtp_config.rtcp_report_interval_ms = kRtcpReportIntervalMs;
rtp_config.outgoing_transport = transport;
rtp_config.local_media_ssrc = local_ssrc;
rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config);
rtp_rtcp_->SetSendingMediaStatus(false);
rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
ingress_ = std::make_unique<AudioIngress>(rtp_rtcp_.get(), clock,
receive_statistics_.get(),
std::move(decoder_factory));
egress_ =
std::make_unique<AudioEgress>(rtp_rtcp_.get(), clock, task_queue_factory);
// Set the instance of audio ingress to be part of audio mixer for ADM to
// fetch audio samples to play.
audio_mixer_->AddSource(ingress_.get());
}
AudioChannel::~AudioChannel() {
if (egress_->IsSending()) {
StopSend();
}
if (ingress_->IsPlaying()) {
StopPlay();
}
audio_mixer_->RemoveSource(ingress_.get());
// TODO(bugs.webrtc.org/11581): unclear if we still need to clear `egress_`
// here.
egress_.reset();
ingress_.reset();
}
bool AudioChannel::StartSend() {
// If encoder has not been set, return false.
if (!egress_->StartSend()) {
return false;
}
// Start sending with RTP stack if it has not been sending yet.
if (!rtp_rtcp_->Sending()) {
rtp_rtcp_->SetSendingStatus(true);
}
return true;
}
void AudioChannel::StopSend() {
egress_->StopSend();
// Deactivate RTP stack when both sending and receiving are stopped.
// SetSendingStatus(false) triggers the transmission of RTCP BYE
// message to remote endpoint.
if (!ingress_->IsPlaying() && rtp_rtcp_->Sending()) {
rtp_rtcp_->SetSendingStatus(false);
}
}
bool AudioChannel::StartPlay() {
// If decoders have not been set, return false.
if (!ingress_->StartPlay()) {
return false;
}
// If RTP stack is not sending then start sending as in recv-only mode, RTCP
// receiver report is expected.
if (!rtp_rtcp_->Sending()) {
rtp_rtcp_->SetSendingStatus(true);
}
return true;
}
void AudioChannel::StopPlay() {
ingress_->StopPlay();
// Deactivate RTP stack only when both sending and receiving are stopped.
if (!rtp_rtcp_->SendingMedia() && rtp_rtcp_->Sending()) {
rtp_rtcp_->SetSendingStatus(false);
}
}
IngressStatistics AudioChannel::GetIngressStatistics() {
IngressStatistics ingress_stats;
NetworkStatistics stats = ingress_->GetNetworkStatistics();
ingress_stats.neteq_stats.total_samples_received = stats.totalSamplesReceived;
ingress_stats.neteq_stats.concealed_samples = stats.concealedSamples;
ingress_stats.neteq_stats.concealment_events = stats.concealmentEvents;
ingress_stats.neteq_stats.jitter_buffer_delay_ms = stats.jitterBufferDelayMs;
ingress_stats.neteq_stats.jitter_buffer_emitted_count =
stats.jitterBufferEmittedCount;
ingress_stats.neteq_stats.jitter_buffer_target_delay_ms =
stats.jitterBufferTargetDelayMs;
ingress_stats.neteq_stats.inserted_samples_for_deceleration =
stats.insertedSamplesForDeceleration;
ingress_stats.neteq_stats.removed_samples_for_acceleration =
stats.removedSamplesForAcceleration;
ingress_stats.neteq_stats.silent_concealed_samples =
stats.silentConcealedSamples;
ingress_stats.neteq_stats.fec_packets_received = stats.fecPacketsReceived;
ingress_stats.neteq_stats.fec_packets_discarded = stats.fecPacketsDiscarded;
ingress_stats.neteq_stats.delayed_packet_outage_samples =
stats.delayedPacketOutageSamples;
ingress_stats.neteq_stats.relative_packet_arrival_delay_ms =
stats.relativePacketArrivalDelayMs;
ingress_stats.neteq_stats.interruption_count = stats.interruptionCount;
ingress_stats.neteq_stats.total_interruption_duration_ms =
stats.totalInterruptionDurationMs;
ingress_stats.total_duration = ingress_->GetOutputTotalDuration();
return ingress_stats;
}
ChannelStatistics AudioChannel::GetChannelStatistics() {
ChannelStatistics channel_stat = ingress_->GetChannelStatistics();
StreamDataCounters rtp_stats, rtx_stats;
rtp_rtcp_->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
channel_stat.bytes_sent =
rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
channel_stat.packets_sent =
rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
return channel_stat;
}
} // namespace webrtc
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