c0df5fc25b
This is one last CL that includes the rest of VoIP API implementation. Bug: webrtc:11251 Change-Id: I3f1b0bf2fd48be864ffc73482105f9514f75f9e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173860 Commit-Queue: Tim Na <natim@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31168}
127 lines
3.9 KiB
C++
127 lines
3.9 KiB
C++
/*
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* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "audio/voip/audio_channel.h"
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#include <utility>
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#include <vector>
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#include "api/audio_codecs/audio_format.h"
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#include "api/task_queue/task_queue_factory.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/location.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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namespace {
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constexpr int kRtcpReportIntervalMs = 5000;
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} // namespace
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AudioChannel::AudioChannel(
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Transport* transport,
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uint32_t local_ssrc,
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TaskQueueFactory* task_queue_factory,
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ProcessThread* process_thread,
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AudioMixer* audio_mixer,
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
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: audio_mixer_(audio_mixer), process_thread_(process_thread) {
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RTC_DCHECK(task_queue_factory);
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RTC_DCHECK(process_thread);
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RTC_DCHECK(audio_mixer);
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Clock* clock = Clock::GetRealTimeClock();
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receive_statistics_ = ReceiveStatistics::Create(clock);
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RtpRtcp::Configuration rtp_config;
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rtp_config.clock = clock;
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rtp_config.audio = true;
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rtp_config.receive_statistics = receive_statistics_.get();
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rtp_config.rtcp_report_interval_ms = kRtcpReportIntervalMs;
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rtp_config.outgoing_transport = transport;
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rtp_config.local_media_ssrc = local_ssrc;
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rtp_rtcp_ = RtpRtcp::Create(rtp_config);
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rtp_rtcp_->SetSendingMediaStatus(false);
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rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
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// ProcessThread periodically services RTP stack for RTCP.
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process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
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ingress_ = std::make_unique<AudioIngress>(rtp_rtcp_.get(), clock,
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receive_statistics_.get(),
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std::move(decoder_factory));
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egress_ =
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std::make_unique<AudioEgress>(rtp_rtcp_.get(), clock, task_queue_factory);
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// Set the instance of audio ingress to be part of audio mixer for ADM to
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// fetch audio samples to play.
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audio_mixer_->AddSource(ingress_.get());
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}
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AudioChannel::~AudioChannel() {
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if (egress_->IsSending()) {
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StopSend();
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}
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if (ingress_->IsPlaying()) {
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StopPlay();
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}
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audio_mixer_->RemoveSource(ingress_.get());
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process_thread_->DeRegisterModule(rtp_rtcp_.get());
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}
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void AudioChannel::StartSend() {
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egress_->StartSend();
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// Start sending with RTP stack if it has not been sending yet.
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if (!rtp_rtcp_->Sending() && rtp_rtcp_->SetSendingStatus(true) != 0) {
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RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending";
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}
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}
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void AudioChannel::StopSend() {
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egress_->StopSend();
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// If the channel is not playing and RTP stack is active then deactivate RTP
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// stack. SetSendingStatus(false) triggers the transmission of RTCP BYE
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// message to remote endpoint.
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if (!IsPlaying() && rtp_rtcp_->Sending() &&
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rtp_rtcp_->SetSendingStatus(false) != 0) {
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RTC_DLOG(LS_ERROR) << "StopSend() RTP/RTCP failed to stop sending";
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}
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}
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void AudioChannel::StartPlay() {
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ingress_->StartPlay();
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// If RTP stack is not sending then start sending as in recv-only mode, RTCP
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// receiver report is expected.
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if (!rtp_rtcp_->Sending() && rtp_rtcp_->SetSendingStatus(true) != 0) {
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RTC_DLOG(LS_ERROR) << "StartPlay() RTP/RTCP failed to start sending";
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}
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}
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void AudioChannel::StopPlay() {
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ingress_->StopPlay();
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// Deactivate RTP stack only when both sending and receiving are stopped.
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if (!IsSendingMedia() && rtp_rtcp_->Sending() &&
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rtp_rtcp_->SetSendingStatus(false) != 0) {
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RTC_DLOG(LS_ERROR) << "StopPlay() RTP/RTCP failed to stop sending";
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}
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}
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} // namespace webrtc
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