f25761d798
The 'Module' part of the implementation must not be called via the RtpRtcp interface, but is rather a part of the contract with ProcessThread. That in turn is an implementation detail for how timers are currently implemented in the default implementation. Along the way I'm deprecating away the factory function which was inside the interface and tied it to one specific implementation. Instead, I'm moving that to the implementation itself and down the line, we don't have to go through it if we just want to create an instance of the class. The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h header file (things moved from rtp_rtcp.h), the rest falls from that. Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce Bug: webrtc:11581 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31440}
128 lines
3.9 KiB
C++
128 lines
3.9 KiB
C++
/*
|
|
* 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/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);
|
|
|
|
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);
|
|
|
|
// 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
|