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webrtc_m130/pc/channel_manager.cc
Tomas Gunnarsson 1933d3b677 Move network thread invokes for initialization for media channels, out. 
Remove Init_w and Deinit(), both of which were wrappers around Invoke()
calls from the worker thread to the network thread.

Instead, replace them with Init_n() and Deinit_n() that are currently*
required to be called by external code in order to associate/disassociate
the channels with the transport.

This CL mostly moves things around in order to prepare for upcoming
changes, but it does change channel destruction in the following way:
- When destroying channels, we don't block the worker thread anymore
  while uninitialization happens on the network thread. Previously
  both signal and worker threads were blocked during the
  uninitialization in the ChannelManager.

* In an upcoming CL, Init_n() and Deinit_n() will be called internally
  from a different method that's always called on the network thread
  when a channel is associated/disassociated with a transceiver. When
  we're there, we will have removed several invokes that currently are
  a part of constructing/destructing channel objects.

Bug: webrtc:11992
Change-Id: Ibc30447a40749ceb36d37834b0cfc5c5ea60e895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246502
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35707}
2022-01-17 14:06:42 +00:00

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/*
* Copyright 2004 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 "pc/channel_manager.h"
#include <algorithm>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/sequence_checker.h"
#include "media/base/media_constants.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
namespace cricket {
// static
std::unique_ptr<ChannelManager> ChannelManager::Create(
std::unique_ptr<MediaEngineInterface> media_engine,
bool enable_rtx,
rtc::Thread* worker_thread,
rtc::Thread* network_thread) {
RTC_DCHECK_RUN_ON(worker_thread);
RTC_DCHECK(network_thread);
RTC_DCHECK(worker_thread);
if (media_engine)
media_engine->Init();
return absl::WrapUnique(new ChannelManager(
std::move(media_engine), enable_rtx, worker_thread, network_thread));
}
ChannelManager::ChannelManager(
std::unique_ptr<MediaEngineInterface> media_engine,
bool enable_rtx,
rtc::Thread* worker_thread,
rtc::Thread* network_thread)
: media_engine_(std::move(media_engine)),
worker_thread_(worker_thread),
network_thread_(network_thread),
enable_rtx_(enable_rtx) {
RTC_DCHECK(worker_thread_);
RTC_DCHECK(network_thread_);
RTC_DCHECK_RUN_ON(worker_thread_);
}
ChannelManager::~ChannelManager() {
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(voice_channels_.empty());
RTC_DCHECK(video_channels_.empty());
}
void ChannelManager::GetSupportedAudioSendCodecs(
std::vector<AudioCodec>* codecs) const {
if (!media_engine_) {
return;
}
*codecs = media_engine_->voice().send_codecs();
}
void ChannelManager::GetSupportedAudioReceiveCodecs(
std::vector<AudioCodec>* codecs) const {
if (!media_engine_) {
return;
}
*codecs = media_engine_->voice().recv_codecs();
}
void ChannelManager::GetSupportedVideoSendCodecs(
std::vector<VideoCodec>* codecs) const {
if (!media_engine_) {
return;
}
codecs->clear();
std::vector<VideoCodec> video_codecs = media_engine_->video().send_codecs();
for (const auto& video_codec : video_codecs) {
if (!enable_rtx_ &&
absl::EqualsIgnoreCase(kRtxCodecName, video_codec.name)) {
continue;
}
codecs->push_back(video_codec);
}
}
void ChannelManager::GetSupportedVideoReceiveCodecs(
std::vector<VideoCodec>* codecs) const {
if (!media_engine_) {
return;
}
codecs->clear();
std::vector<VideoCodec> video_codecs = media_engine_->video().recv_codecs();
for (const auto& video_codec : video_codecs) {
if (!enable_rtx_ &&
absl::EqualsIgnoreCase(kRtxCodecName, video_codec.name)) {
continue;
}
codecs->push_back(video_codec);
}
}
RtpHeaderExtensions ChannelManager::GetDefaultEnabledAudioRtpHeaderExtensions()
const {
if (!media_engine_)
return {};
return GetDefaultEnabledRtpHeaderExtensions(media_engine_->voice());
}
std::vector<webrtc::RtpHeaderExtensionCapability>
ChannelManager::GetSupportedAudioRtpHeaderExtensions() const {
if (!media_engine_)
return {};
return media_engine_->voice().GetRtpHeaderExtensions();
}
RtpHeaderExtensions ChannelManager::GetDefaultEnabledVideoRtpHeaderExtensions()
const {
if (!media_engine_)
return {};
return GetDefaultEnabledRtpHeaderExtensions(media_engine_->video());
}
std::vector<webrtc::RtpHeaderExtensionCapability>
ChannelManager::GetSupportedVideoRtpHeaderExtensions() const {
if (!media_engine_)
return {};
return media_engine_->video().GetRtpHeaderExtensions();
}
VoiceChannel* ChannelManager::CreateVoiceChannel(
webrtc::Call* call,
const MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator,
const AudioOptions& options) {
RTC_DCHECK(call);
RTC_DCHECK(media_engine_);
// TODO(bugs.webrtc.org/11992): Remove this workaround after updates in
// PeerConnection and add the expectation that we're already on the right
// thread.
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
return CreateVoiceChannel(call, media_config, rtp_transport,
signaling_thread, content_name, srtp_required,
crypto_options, ssrc_generator, options);
});
}
RTC_DCHECK_RUN_ON(worker_thread_);
VoiceMediaChannel* media_channel = media_engine_->voice().CreateMediaChannel(
call, media_config, options, crypto_options);
if (!media_channel) {
return nullptr;
}
auto voice_channel = std::make_unique<VoiceChannel>(
worker_thread_, network_thread_, signaling_thread,
absl::WrapUnique(media_channel), content_name, srtp_required,
crypto_options, ssrc_generator);
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(voice_channel->network_thread());
voice_channel->Init_n(rtp_transport);
});
VoiceChannel* voice_channel_ptr = voice_channel.get();
voice_channels_.push_back(std::move(voice_channel));
return voice_channel_ptr;
}
void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) {
TRACE_EVENT0("webrtc", "ChannelManager::DestroyVoiceChannel");
RTC_DCHECK(voice_channel);
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(voice_channel->network_thread());
voice_channel->Deinit_n();
});
if (!worker_thread_->IsCurrent()) {
worker_thread_->Invoke<void>(RTC_FROM_HERE,
[&] { DestroyVoiceChannel(voice_channel); });
return;
}
RTC_DCHECK_RUN_ON(worker_thread_);
voice_channels_.erase(absl::c_find_if(
voice_channels_, [&](const std::unique_ptr<VoiceChannel>& p) {
return p.get() == voice_channel;
}));
}
VideoChannel* ChannelManager::CreateVideoChannel(
webrtc::Call* call,
const MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator,
const VideoOptions& options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
RTC_DCHECK(call);
RTC_DCHECK(media_engine_);
// TODO(bugs.webrtc.org/11992): Remove this workaround after updates in
// PeerConnection and add the expectation that we're already on the right
// thread.
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
return CreateVideoChannel(call, media_config, rtp_transport,
signaling_thread, content_name, srtp_required,
crypto_options, ssrc_generator, options,
video_bitrate_allocator_factory);
});
}
RTC_DCHECK_RUN_ON(worker_thread_);
VideoMediaChannel* media_channel = media_engine_->video().CreateMediaChannel(
call, media_config, options, crypto_options,
video_bitrate_allocator_factory);
if (!media_channel) {
return nullptr;
}
auto video_channel = std::make_unique<VideoChannel>(
worker_thread_, network_thread_, signaling_thread,
absl::WrapUnique(media_channel), content_name, srtp_required,
crypto_options, ssrc_generator);
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(video_channel->network_thread());
video_channel->Init_n(rtp_transport);
});
VideoChannel* video_channel_ptr = video_channel.get();
video_channels_.push_back(std::move(video_channel));
return video_channel_ptr;
}
void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) {
TRACE_EVENT0("webrtc", "ChannelManager::DestroyVideoChannel");
RTC_DCHECK(video_channel);
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(video_channel->network_thread());
video_channel->Deinit_n();
});
if (!worker_thread_->IsCurrent()) {
worker_thread_->Invoke<void>(RTC_FROM_HERE,
[&] { DestroyVideoChannel(video_channel); });
return;
}
RTC_DCHECK_RUN_ON(worker_thread_);
video_channels_.erase(absl::c_find_if(
video_channels_, [&](const std::unique_ptr<VideoChannel>& p) {
return p.get() == video_channel;
}));
}
bool ChannelManager::StartAecDump(webrtc::FileWrapper file,
int64_t max_size_bytes) {
RTC_DCHECK_RUN_ON(worker_thread_);
return media_engine_->voice().StartAecDump(std::move(file), max_size_bytes);
}
void ChannelManager::StopAecDump() {
RTC_DCHECK_RUN_ON(worker_thread_);
media_engine_->voice().StopAecDump();
}
} // namespace cricket
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