webrtc/pc/channelmanager.cc
Zhi Huang 2dfc42d7b6 Prepare to make BaseChannel depend on RtpTransportInternal only.
Eventually we want BaseChannel to depend on the RtpTransportInternal
instead of DtlsTransportInternal and share RtpTransport when bundling.
This CL is the first step.

Add SetRtpTransport and Init_w(RtptransportInternal*) to BaseChannel.
These two methods would replace the existing SetTransports and Init_w
methods.

Add new CreateVoice/VideoChannel methods to the ChannelManager which
 take RtpTransportInternal instead of Dtls/PacketTransportInternal.

|cotnent_name| is removed from the SrtpTransport to simplify to code
since it is only used for debugging.

InitNetwork_n is removed from BaseChannel in CL as well.

Bug: webrtc:7013
Change-Id: I35b1565958548bd4896854c49e61d3ee160b7634
Reviewed-on: https://webrtc-review.googlesource.com/27840
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21057}
2017-12-04 22:27:39 +00:00

526 lines
17 KiB
C++

/*
* 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/channelmanager.h"
#include <algorithm>
#include <utility>
#include "media/base/rtpdataengine.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/trace_event.h"
namespace cricket {
ChannelManager::ChannelManager(
std::unique_ptr<MediaEngineInterface> media_engine,
std::unique_ptr<DataEngineInterface> data_engine,
rtc::Thread* worker_thread,
rtc::Thread* network_thread)
: media_engine_(std::move(media_engine)),
data_engine_(std::move(data_engine)),
main_thread_(rtc::Thread::Current()),
worker_thread_(worker_thread),
network_thread_(network_thread) {
RTC_DCHECK(data_engine_);
RTC_DCHECK(worker_thread_);
RTC_DCHECK(network_thread_);
}
ChannelManager::~ChannelManager() {
if (initialized_) {
Terminate();
}
// The media engine needs to be deleted on the worker thread for thread safe
// destruction,
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] { media_engine_.reset(); });
}
bool ChannelManager::SetVideoRtxEnabled(bool enable) {
// To be safe, this call is only allowed before initialization. Apps like
// Flute only have a singleton ChannelManager and we don't want this flag to
// be toggled between calls or when there's concurrent calls. We expect apps
// to enable this at startup and retain that setting for the lifetime of the
// app.
if (!initialized_) {
enable_rtx_ = enable;
return true;
} else {
RTC_LOG(LS_WARNING) << "Cannot toggle rtx after initialization!";
return false;
}
}
void ChannelManager::GetSupportedAudioSendCodecs(
std::vector<AudioCodec>* codecs) const {
if (!media_engine_) {
return;
}
*codecs = media_engine_->audio_send_codecs();
}
void ChannelManager::GetSupportedAudioReceiveCodecs(
std::vector<AudioCodec>* codecs) const {
if (!media_engine_) {
return;
}
*codecs = media_engine_->audio_recv_codecs();
}
void ChannelManager::GetSupportedAudioRtpHeaderExtensions(
RtpHeaderExtensions* ext) const {
if (!media_engine_) {
return;
}
*ext = media_engine_->GetAudioCapabilities().header_extensions;
}
void ChannelManager::GetSupportedVideoCodecs(
std::vector<VideoCodec>* codecs) const {
if (!media_engine_) {
return;
}
codecs->clear();
std::vector<VideoCodec> video_codecs = media_engine_->video_codecs();
for (const auto& video_codec : video_codecs) {
if (!enable_rtx_ &&
_stricmp(kRtxCodecName, video_codec.name.c_str()) == 0) {
continue;
}
codecs->push_back(video_codec);
}
}
void ChannelManager::GetSupportedVideoRtpHeaderExtensions(
RtpHeaderExtensions* ext) const {
if (!media_engine_) {
return;
}
*ext = media_engine_->GetVideoCapabilities().header_extensions;
}
void ChannelManager::GetSupportedDataCodecs(
std::vector<DataCodec>* codecs) const {
*codecs = data_engine_->data_codecs();
}
bool ChannelManager::Init() {
RTC_DCHECK(!initialized_);
if (initialized_) {
return false;
}
RTC_DCHECK(network_thread_);
RTC_DCHECK(worker_thread_);
if (!network_thread_->IsCurrent()) {
// Do not allow invoking calls to other threads on the network thread.
network_thread_->Invoke<void>(
RTC_FROM_HERE, [&] { network_thread_->SetAllowBlockingCalls(false); });
}
if (media_engine_) {
initialized_ = worker_thread_->Invoke<bool>(
RTC_FROM_HERE, [&] { return media_engine_->Init(); });
RTC_DCHECK(initialized_);
} else {
initialized_ = true;
}
return initialized_;
}
void ChannelManager::Terminate() {
RTC_DCHECK(initialized_);
if (!initialized_) {
return;
}
// Need to destroy the channels on the worker thread.
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
video_channels_.clear();
voice_channels_.clear();
data_channels_.clear();
});
initialized_ = false;
}
VoiceChannel* ChannelManager::CreateVoiceChannel(
webrtc::Call* call,
const cricket::MediaConfig& media_config,
DtlsTransportInternal* rtp_transport,
DtlsTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const AudioOptions& options) {
return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
return CreateVoiceChannel_w(
call, media_config, rtp_transport, rtcp_transport, rtp_transport,
rtcp_transport, signaling_thread, content_name, srtp_required, options);
});
}
VoiceChannel* ChannelManager::CreateVoiceChannel(
webrtc::Call* call,
const cricket::MediaConfig& media_config,
rtc::PacketTransportInternal* rtp_transport,
rtc::PacketTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const AudioOptions& options) {
return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
return CreateVoiceChannel_w(call, media_config, nullptr, nullptr,
rtp_transport, rtcp_transport, signaling_thread,
content_name, srtp_required, options);
});
}
VoiceChannel* ChannelManager::CreateVoiceChannel(
webrtc::Call* call,
const cricket::MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const AudioOptions& options) {
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,
options);
});
}
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(initialized_);
RTC_DCHECK(call);
if (!media_engine_) {
return nullptr;
}
VoiceMediaChannel* media_channel =
media_engine_->CreateChannel(call, media_config, options);
if (!media_channel) {
return nullptr;
}
auto voice_channel = rtc::MakeUnique<VoiceChannel>(
worker_thread_, network_thread_, signaling_thread, media_engine_.get(),
rtc::WrapUnique(media_channel), content_name,
rtp_transport->rtcp_packet_transport() == nullptr, srtp_required);
voice_channel->Init_w(rtp_transport);
VoiceChannel* voice_channel_ptr = voice_channel.get();
voice_channels_.push_back(std::move(voice_channel));
return voice_channel_ptr;
}
VoiceChannel* ChannelManager::CreateVoiceChannel_w(
webrtc::Call* call,
const cricket::MediaConfig& media_config,
DtlsTransportInternal* rtp_dtls_transport,
DtlsTransportInternal* rtcp_dtls_transport,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const AudioOptions& options) {
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(initialized_);
RTC_DCHECK(call);
if (!media_engine_) {
return nullptr;
}
VoiceMediaChannel* media_channel = media_engine_->CreateChannel(
call, media_config, options);
if (!media_channel) {
return nullptr;
}
auto voice_channel = rtc::MakeUnique<VoiceChannel>(
worker_thread_, network_thread_, signaling_thread, media_engine_.get(),
rtc::WrapUnique(media_channel), content_name,
rtcp_packet_transport == nullptr, srtp_required);
voice_channel->Init_w(rtp_dtls_transport, rtcp_dtls_transport,
rtp_packet_transport, rtcp_packet_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");
if (!voice_channel) {
return;
}
if (!worker_thread_->IsCurrent()) {
worker_thread_->Invoke<void>(RTC_FROM_HERE,
[&] { DestroyVoiceChannel(voice_channel); });
return;
}
RTC_DCHECK(initialized_);
auto it = std::find_if(voice_channels_.begin(), voice_channels_.end(),
[&](const std::unique_ptr<VoiceChannel>& p) {
return p.get() == voice_channel;
});
RTC_DCHECK(it != voice_channels_.end());
if (it == voice_channels_.end()) {
return;
}
voice_channels_.erase(it);
}
VideoChannel* ChannelManager::CreateVideoChannel(
webrtc::Call* call,
const cricket::MediaConfig& media_config,
DtlsTransportInternal* rtp_transport,
DtlsTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const VideoOptions& options) {
return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
return CreateVideoChannel_w(
call, media_config, rtp_transport, rtcp_transport, rtp_transport,
rtcp_transport, signaling_thread, content_name, srtp_required, options);
});
}
VideoChannel* ChannelManager::CreateVideoChannel(
webrtc::Call* call,
const cricket::MediaConfig& media_config,
rtc::PacketTransportInternal* rtp_transport,
rtc::PacketTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const VideoOptions& options) {
return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
return CreateVideoChannel_w(call, media_config, nullptr, nullptr,
rtp_transport, rtcp_transport, signaling_thread,
content_name, srtp_required, options);
});
}
VideoChannel* ChannelManager::CreateVideoChannel(
webrtc::Call* call,
const cricket::MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const VideoOptions& options) {
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,
options);
});
}
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(initialized_);
RTC_DCHECK(call);
if (!media_engine_) {
return nullptr;
}
VideoMediaChannel* media_channel =
media_engine_->CreateVideoChannel(call, media_config, options);
if (!media_channel) {
return nullptr;
}
auto video_channel = rtc::MakeUnique<VideoChannel>(
worker_thread_, network_thread_, signaling_thread,
rtc::WrapUnique(media_channel), content_name,
rtp_transport->rtcp_packet_transport() == nullptr, srtp_required);
video_channel->Init_w(rtp_transport);
VideoChannel* video_channel_ptr = video_channel.get();
video_channels_.push_back(std::move(video_channel));
return video_channel_ptr;
}
VideoChannel* ChannelManager::CreateVideoChannel_w(
webrtc::Call* call,
const cricket::MediaConfig& media_config,
DtlsTransportInternal* rtp_dtls_transport,
DtlsTransportInternal* rtcp_dtls_transport,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const VideoOptions& options) {
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(initialized_);
RTC_DCHECK(call);
if (!media_engine_) {
return nullptr;
}
VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel(
call, media_config, options);
if (!media_channel) {
return nullptr;
}
auto video_channel = rtc::MakeUnique<VideoChannel>(
worker_thread_, network_thread_, signaling_thread,
rtc::WrapUnique(media_channel), content_name,
rtcp_packet_transport == nullptr, srtp_required);
video_channel->Init_w(rtp_dtls_transport, rtcp_dtls_transport,
rtp_packet_transport, rtcp_packet_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");
if (!video_channel) {
return;
}
if (!worker_thread_->IsCurrent()) {
worker_thread_->Invoke<void>(RTC_FROM_HERE,
[&] { DestroyVideoChannel(video_channel); });
return;
}
RTC_DCHECK(initialized_);
auto it = std::find_if(video_channels_.begin(), video_channels_.end(),
[&](const std::unique_ptr<VideoChannel>& p) {
return p.get() == video_channel;
});
RTC_DCHECK(it != video_channels_.end());
if (it == video_channels_.end()) {
return;
}
video_channels_.erase(it);
}
RtpDataChannel* ChannelManager::CreateRtpDataChannel(
const cricket::MediaConfig& media_config,
DtlsTransportInternal* rtp_transport,
DtlsTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required) {
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<RtpDataChannel*>(RTC_FROM_HERE, [&] {
return CreateRtpDataChannel(media_config, rtp_transport, rtcp_transport,
signaling_thread, content_name,
srtp_required);
});
}
// This is ok to alloc from a thread other than the worker thread.
RTC_DCHECK(initialized_);
DataMediaChannel* media_channel = data_engine_->CreateChannel(media_config);
if (!media_channel) {
RTC_LOG(LS_WARNING) << "Failed to create RTP data channel.";
return nullptr;
}
auto data_channel = rtc::MakeUnique<RtpDataChannel>(
worker_thread_, network_thread_, signaling_thread,
rtc::WrapUnique(media_channel), content_name, rtcp_transport == nullptr,
srtp_required);
data_channel->Init_w(rtp_transport, rtcp_transport, rtp_transport,
rtcp_transport);
RtpDataChannel* data_channel_ptr = data_channel.get();
data_channels_.push_back(std::move(data_channel));
return data_channel_ptr;
}
RtpDataChannel* ChannelManager::CreateRtpDataChannel(
const cricket::MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required) {
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<RtpDataChannel*>(RTC_FROM_HERE, [&] {
return CreateRtpDataChannel(media_config, rtp_transport, signaling_thread,
content_name, srtp_required);
});
}
// This is ok to alloc from a thread other than the worker thread.
RTC_DCHECK(initialized_);
DataMediaChannel* media_channel = data_engine_->CreateChannel(media_config);
if (!media_channel) {
RTC_LOG(LS_WARNING) << "Failed to create RTP data channel.";
return nullptr;
}
auto data_channel = rtc::MakeUnique<RtpDataChannel>(
worker_thread_, network_thread_, signaling_thread,
rtc::WrapUnique(media_channel), content_name,
rtp_transport->rtcp_packet_transport() == nullptr, srtp_required);
data_channel->Init_w(rtp_transport);
RtpDataChannel* data_channel_ptr = data_channel.get();
data_channels_.push_back(std::move(data_channel));
return data_channel_ptr;
}
void ChannelManager::DestroyRtpDataChannel(RtpDataChannel* data_channel) {
TRACE_EVENT0("webrtc", "ChannelManager::DestroyRtpDataChannel");
if (!data_channel) {
return;
}
if (!worker_thread_->IsCurrent()) {
worker_thread_->Invoke<void>(
RTC_FROM_HERE, [&] { return DestroyRtpDataChannel(data_channel); });
return;
}
RTC_DCHECK(initialized_);
auto it = std::find_if(data_channels_.begin(), data_channels_.end(),
[&](const std::unique_ptr<RtpDataChannel>& p) {
return p.get() == data_channel;
});
RTC_DCHECK(it != data_channels_.end());
if (it == data_channels_.end()) {
return;
}
data_channels_.erase(it);
}
bool ChannelManager::StartAecDump(rtc::PlatformFile file,
int64_t max_size_bytes) {
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_engine_->StartAecDump(file, max_size_bytes);
});
}
void ChannelManager::StopAecDump() {
worker_thread_->Invoke<void>(RTC_FROM_HERE,
[&] { media_engine_->StopAecDump(); });
}
} // namespace cricket