webrtc_m130/pc/channelmanager.cc
Taylor Brandstetter b140b9fd69 Keep count of libsrtp clients, and only deinitialize when it goes to 0.
This is the same thing we're doing for usrsctp. Before this CL, the
first SrtpSession to call SetKey would initialize libsrtp, and
ChannelManager's destructor would deinitialize it. This works for an
application that only uses one instance of ChannelManager simultaneously
(or one instance of PeerConnectionFactory), but not one that uses
multiple.

Now, libsrtp is effectively reference-counted, with the first
SrtpSession to take a reference initializing it, and the last to remove
its reference deinitializing it.

This issue was discovered recently due to a change that resulted in
using srtp_update. Without using that method, the issue went unnoticed;
maybe srtp_protect/srtp_unprotect don't require initialization?

Bug: webrtc:8388
Change-Id: If1329360f0b469e454810e62e9b5acfbd4cba100
Reviewed-on: https://webrtc-review.googlesource.com/9000
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20262}
2017-10-13 05:11:27 +00:00

453 lines
15 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 "media/base/device.h"
#include "media/base/rtpdataengine.h"
#include "pc/srtpfilter.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/stringencode.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/trace_event.h"
namespace cricket {
using rtc::Bind;
ChannelManager::ChannelManager(std::unique_ptr<MediaEngineInterface> me,
std::unique_ptr<DataEngineInterface> dme,
rtc::Thread* thread) {
Construct(std::move(me), std::move(dme), thread, thread);
}
ChannelManager::ChannelManager(std::unique_ptr<MediaEngineInterface> me,
rtc::Thread* worker_thread,
rtc::Thread* network_thread) {
Construct(std::move(me),
std::unique_ptr<DataEngineInterface>(new RtpDataEngine()),
worker_thread, network_thread);
}
void ChannelManager::Construct(std::unique_ptr<MediaEngineInterface> me,
std::unique_ptr<DataEngineInterface> dme,
rtc::Thread* worker_thread,
rtc::Thread* network_thread) {
media_engine_ = std::move(me);
data_media_engine_ = std::move(dme);
initialized_ = false;
main_thread_ = rtc::Thread::Current();
worker_thread_ = worker_thread;
network_thread_ = network_thread;
capturing_ = false;
enable_rtx_ = false;
}
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, Bind(&ChannelManager::DestructorDeletes_w, this));
}
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 {
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 {
if (!data_media_engine_) {
return;
}
*codecs = data_media_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<bool>(
RTC_FROM_HERE,
rtc::Bind(&rtc::Thread::SetAllowBlockingCalls, network_thread_, false));
}
initialized_ = worker_thread_->Invoke<bool>(
RTC_FROM_HERE, Bind(&ChannelManager::InitMediaEngine_w, this));
RTC_DCHECK(initialized_);
return initialized_;
}
bool ChannelManager::InitMediaEngine_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
if (media_engine_) {
return media_engine_->Init();
}
return true;
}
void ChannelManager::Terminate() {
RTC_DCHECK(initialized_);
if (!initialized_) {
return;
}
worker_thread_->Invoke<void>(RTC_FROM_HERE,
Bind(&ChannelManager::Terminate_w, this));
initialized_ = false;
}
void ChannelManager::DestructorDeletes_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
media_engine_.reset(nullptr);
}
void ChannelManager::Terminate_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
// Need to destroy the voice/video channels
video_channels_.clear();
voice_channels_.clear();
}
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,
Bind(&ChannelManager::CreateVoiceChannel_w, this, 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,
Bind(&ChannelManager::CreateVoiceChannel_w, this, call, media_config,
nullptr, nullptr, rtp_transport, rtcp_transport, signaling_thread,
content_name, srtp_required, options));
}
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(initialized_);
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
RTC_DCHECK(nullptr != call);
if (!media_engine_) {
return nullptr;
}
VoiceMediaChannel* media_channel = media_engine_->CreateChannel(
call, media_config, options);
if (!media_channel) {
return nullptr;
}
std::unique_ptr<VoiceChannel> voice_channel(
new VoiceChannel(worker_thread_, network_thread_, signaling_thread,
media_engine_.get(), media_channel, content_name,
rtcp_packet_transport == nullptr, srtp_required));
if (!voice_channel->Init_w(rtp_dtls_transport, rtcp_dtls_transport,
rtp_packet_transport, rtcp_packet_transport)) {
return nullptr;
}
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) {
worker_thread_->Invoke<void>(
RTC_FROM_HERE,
Bind(&ChannelManager::DestroyVoiceChannel_w, this, voice_channel));
}
}
void ChannelManager::DestroyVoiceChannel_w(VoiceChannel* voice_channel) {
TRACE_EVENT0("webrtc", "ChannelManager::DestroyVoiceChannel_w");
RTC_DCHECK(initialized_);
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
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,
Bind(&ChannelManager::CreateVideoChannel_w, this, 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,
Bind(&ChannelManager::CreateVideoChannel_w, this, call, media_config,
nullptr, nullptr, rtp_transport, rtcp_transport, signaling_thread,
content_name, srtp_required, options));
}
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(initialized_);
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
RTC_DCHECK(nullptr != call);
VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel(
call, media_config, options);
if (!media_channel) {
return nullptr;
}
std::unique_ptr<VideoChannel> video_channel(new VideoChannel(
worker_thread_, network_thread_, signaling_thread, media_channel,
content_name, rtcp_packet_transport == nullptr, srtp_required));
if (!video_channel->Init_w(rtp_dtls_transport, rtcp_dtls_transport,
rtp_packet_transport, rtcp_packet_transport)) {
return nullptr;
}
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) {
worker_thread_->Invoke<void>(
RTC_FROM_HERE,
Bind(&ChannelManager::DestroyVideoChannel_w, this, video_channel));
}
}
void ChannelManager::DestroyVideoChannel_w(VideoChannel* video_channel) {
TRACE_EVENT0("webrtc", "ChannelManager::DestroyVideoChannel_w");
RTC_DCHECK(initialized_);
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
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) {
return worker_thread_->Invoke<RtpDataChannel*>(
RTC_FROM_HERE, Bind(&ChannelManager::CreateRtpDataChannel_w, this,
media_config, rtp_transport, rtcp_transport,
signaling_thread, content_name, srtp_required));
}
RtpDataChannel* ChannelManager::CreateRtpDataChannel_w(
const cricket::MediaConfig& media_config,
DtlsTransportInternal* rtp_transport,
DtlsTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required) {
// This is ok to alloc from a thread other than the worker thread.
RTC_DCHECK(initialized_);
DataMediaChannel* media_channel
= data_media_engine_->CreateChannel(media_config);
if (!media_channel) {
LOG(LS_WARNING) << "Failed to create RTP data channel.";
return nullptr;
}
std::unique_ptr<RtpDataChannel> data_channel(new RtpDataChannel(
worker_thread_, network_thread_, signaling_thread, media_channel,
content_name, rtcp_transport == nullptr, srtp_required));
if (!data_channel->Init_w(rtp_transport, rtcp_transport, rtp_transport,
rtcp_transport)) {
LOG(LS_WARNING) << "Failed to init data channel.";
return nullptr;
}
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) {
worker_thread_->Invoke<void>(
RTC_FROM_HERE,
Bind(&ChannelManager::DestroyRtpDataChannel_w, this, data_channel));
}
}
void ChannelManager::DestroyRtpDataChannel_w(RtpDataChannel* data_channel) {
TRACE_EVENT0("webrtc", "ChannelManager::DestroyRtpDataChannel_w");
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, Bind(&MediaEngineInterface::StartAecDump,
media_engine_.get(), file, max_size_bytes));
}
void ChannelManager::StopAecDump() {
worker_thread_->Invoke<void>(
RTC_FROM_HERE,
Bind(&MediaEngineInterface::StopAecDump, media_engine_.get()));
}
} // namespace cricket