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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 "rtc_base/critical_section.h"
#include "rtc_base/location.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,
ProcessThread* process_thread,
AudioMixer* audio_mixer,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
: audio_mixer_(audio_mixer), process_thread_(process_thread) {
RTC_DCHECK(task_queue_factory);
RTC_DCHECK(process_thread);
RTC_DCHECK(audio_mixer);
Clock* clock = Clock::GetRealTimeClock();
receive_statistics_ = ReceiveStatistics::Create(clock);
RtpRtcp::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_ = RtpRtcp::Create(rtp_config);
rtp_rtcp_->SetSendingMediaStatus(false);
rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
// ProcessThread periodically services RTP stack for RTCP.
process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
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());
process_thread_->DeRegisterModule(rtp_rtcp_.get());
}
void AudioChannel::StartSend() {
egress_->StartSend();
// Start sending with RTP stack if it has not been sending yet.
if (!rtp_rtcp_->Sending() && rtp_rtcp_->SetSendingStatus(true) != 0) {
RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending";
}
}
void AudioChannel::StopSend() {
egress_->StopSend();
// If the channel is not playing and RTP stack is active then deactivate RTP
// stack. SetSendingStatus(false) triggers the transmission of RTCP BYE
// message to remote endpoint.
if (!IsPlaying() && rtp_rtcp_->Sending() &&
rtp_rtcp_->SetSendingStatus(false) != 0) {
RTC_DLOG(LS_ERROR) << "StopSend() RTP/RTCP failed to stop sending";
}
}
void AudioChannel::StartPlay() {
ingress_->StartPlay();
// 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) != 0) {
RTC_DLOG(LS_ERROR) << "StartPlay() RTP/RTCP failed to start sending";
}
}
void AudioChannel::StopPlay() {
ingress_->StopPlay();
// Deactivate RTP stack only when both sending and receiving are stopped.
if (!IsSendingMedia() && rtp_rtcp_->Sending() &&
rtp_rtcp_->SetSendingStatus(false) != 0) {
RTC_DLOG(LS_ERROR) << "StopPlay() RTP/RTCP failed to stop sending";
}
}
} // namespace webrtc
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