blob: 02a53f8877300b266c74402c090e3b93ede01140 [file] [log] [blame]
/*
* 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 "webrtc/pc/channelmanager.h"
#include <algorithm>
#include "webrtc/api/mediacontroller.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/media/base/device.h"
#include "webrtc/media/base/rtpdataengine.h"
#include "webrtc/pc/srtpfilter.h"
namespace cricket {
using rtc::Bind;
static DataEngineInterface* ConstructDataEngine() {
return new RtpDataEngine();
}
ChannelManager::ChannelManager(MediaEngineInterface* me,
DataEngineInterface* dme,
rtc::Thread* thread) {
Construct(me, dme, thread, thread);
}
ChannelManager::ChannelManager(MediaEngineInterface* me,
rtc::Thread* worker_thread,
rtc::Thread* network_thread) {
Construct(me, ConstructDataEngine(), worker_thread, network_thread);
}
void ChannelManager::Construct(MediaEngineInterface* me,
DataEngineInterface* dme,
rtc::Thread* worker_thread,
rtc::Thread* network_thread) {
media_engine_.reset(me);
data_media_engine_.reset(dme);
initialized_ = false;
main_thread_ = rtc::Thread::Current();
worker_thread_ = worker_thread;
network_thread_ = network_thread;
capturing_ = false;
enable_rtx_ = false;
crypto_options_ = rtc::CryptoOptions::NoGcm();
}
ChannelManager::~ChannelManager() {
if (initialized_) {
Terminate();
// If srtp is initialized (done by the Channel) then we must call
// srtp_shutdown to free all crypto kernel lists. But we need to make sure
// shutdown always called at the end, after channels are destroyed.
// ChannelManager d'tor is always called last, it's safe place to call
// shutdown.
ShutdownSrtp();
}
// 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;
}
}
bool ChannelManager::SetCryptoOptions(
const rtc::CryptoOptions& crypto_options) {
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, Bind(
&ChannelManager::SetCryptoOptions_w, this, crypto_options));
}
bool ChannelManager::SetCryptoOptions_w(
const rtc::CryptoOptions& crypto_options) {
if (!video_channels_.empty() || !voice_channels_.empty() ||
!data_channels_.empty()) {
LOG(LS_WARNING) << "Not changing crypto options in existing channels.";
}
crypto_options_ = crypto_options;
#if defined(ENABLE_EXTERNAL_AUTH)
if (crypto_options_.enable_gcm_crypto_suites) {
// TODO(jbauch): Re-enable once https://crbug.com/628400 is resolved.
crypto_options_.enable_gcm_crypto_suites = false;
LOG(LS_WARNING) << "GCM ciphers are not supported with " <<
"ENABLE_EXTERNAL_AUTH and will be disabled.";
}
#endif
return true;
}
void ChannelManager::GetSupportedAudioSendCodecs(
std::vector<AudioCodec>* codecs) const {
*codecs = media_engine_->audio_send_codecs();
}
void ChannelManager::GetSupportedAudioReceiveCodecs(
std::vector<AudioCodec>* codecs) const {
*codecs = media_engine_->audio_recv_codecs();
}
void ChannelManager::GetSupportedAudioRtpHeaderExtensions(
RtpHeaderExtensions* ext) const {
*ext = media_engine_->GetAudioCapabilities().header_extensions;
}
void ChannelManager::GetSupportedVideoCodecs(
std::vector<VideoCodec>* codecs) const {
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 {
*ext = media_engine_->GetVideoCapabilities().header_extensions;
}
void ChannelManager::GetSupportedDataCodecs(
std::vector<DataCodec>* codecs) const {
*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());
return media_engine_->Init();
}
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(NULL);
}
void ChannelManager::Terminate_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
// Need to destroy the voice/video channels
while (!video_channels_.empty()) {
DestroyVideoChannel_w(video_channels_.back());
}
while (!voice_channels_.empty()) {
DestroyVoiceChannel_w(voice_channels_.back());
}
}
VoiceChannel* ChannelManager::CreateVoiceChannel(
webrtc::MediaControllerInterface* media_controller,
TransportChannel* rtp_transport,
TransportChannel* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required,
const AudioOptions& options) {
return worker_thread_->Invoke<VoiceChannel*>(
RTC_FROM_HERE,
Bind(&ChannelManager::CreateVoiceChannel_w, this, media_controller,
rtp_transport, rtcp_transport, signaling_thread, content_name,
bundle_transport_name, rtcp_mux_required, srtp_required, options));
}
VoiceChannel* ChannelManager::CreateVoiceChannel_w(
webrtc::MediaControllerInterface* media_controller,
TransportChannel* rtp_transport,
TransportChannel* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required,
const AudioOptions& options) {
RTC_DCHECK(initialized_);
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
RTC_DCHECK(nullptr != media_controller);
VoiceMediaChannel* media_channel = media_engine_->CreateChannel(
media_controller->call_w(), media_controller->config(), options);
if (!media_channel)
return nullptr;
VoiceChannel* voice_channel = new VoiceChannel(
worker_thread_, network_thread_, signaling_thread, media_engine_.get(),
media_channel, content_name, rtcp_mux_required, srtp_required);
voice_channel->SetCryptoOptions(crypto_options_);
if (!voice_channel->Init_w(rtp_transport, rtcp_transport)) {
delete voice_channel;
return nullptr;
}
voice_channels_.push_back(voice_channel);
return voice_channel;
}
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");
// Destroy voice channel.
RTC_DCHECK(initialized_);
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
VoiceChannels::iterator it = std::find(voice_channels_.begin(),
voice_channels_.end(), voice_channel);
RTC_DCHECK(it != voice_channels_.end());
if (it == voice_channels_.end())
return;
voice_channels_.erase(it);
delete voice_channel;
}
VideoChannel* ChannelManager::CreateVideoChannel(
webrtc::MediaControllerInterface* media_controller,
TransportChannel* rtp_transport,
TransportChannel* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required,
const VideoOptions& options) {
return worker_thread_->Invoke<VideoChannel*>(
RTC_FROM_HERE,
Bind(&ChannelManager::CreateVideoChannel_w, this, media_controller,
rtp_transport, rtcp_transport, signaling_thread, content_name,
bundle_transport_name, rtcp_mux_required, srtp_required, options));
}
VideoChannel* ChannelManager::CreateVideoChannel_w(
webrtc::MediaControllerInterface* media_controller,
TransportChannel* rtp_transport,
TransportChannel* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required,
const VideoOptions& options) {
RTC_DCHECK(initialized_);
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
RTC_DCHECK(nullptr != media_controller);
VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel(
media_controller->call_w(), media_controller->config(), options);
if (media_channel == NULL) {
return NULL;
}
VideoChannel* video_channel = new VideoChannel(
worker_thread_, network_thread_, signaling_thread, media_channel,
content_name, rtcp_mux_required, srtp_required);
video_channel->SetCryptoOptions(crypto_options_);
if (!video_channel->Init_w(rtp_transport, rtcp_transport)) {
delete video_channel;
return NULL;
}
video_channels_.push_back(video_channel);
return video_channel;
}
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");
// Destroy video channel.
RTC_DCHECK(initialized_);
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
VideoChannels::iterator it = std::find(video_channels_.begin(),
video_channels_.end(), video_channel);
RTC_DCHECK(it != video_channels_.end());
if (it == video_channels_.end())
return;
video_channels_.erase(it);
delete video_channel;
}
RtpDataChannel* ChannelManager::CreateRtpDataChannel(
webrtc::MediaControllerInterface* media_controller,
TransportChannel* rtp_transport,
TransportChannel* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required) {
return worker_thread_->Invoke<RtpDataChannel*>(
RTC_FROM_HERE,
Bind(&ChannelManager::CreateRtpDataChannel_w, this, media_controller,
rtp_transport, rtcp_transport, signaling_thread, content_name,
bundle_transport_name, rtcp_mux_required, srtp_required));
}
RtpDataChannel* ChannelManager::CreateRtpDataChannel_w(
webrtc::MediaControllerInterface* media_controller,
TransportChannel* rtp_transport,
TransportChannel* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required) {
// This is ok to alloc from a thread other than the worker thread.
RTC_DCHECK(initialized_);
MediaConfig config;
if (media_controller) {
config = media_controller->config();
}
DataMediaChannel* media_channel = data_media_engine_->CreateChannel(config);
if (!media_channel) {
LOG(LS_WARNING) << "Failed to create RTP data channel.";
return nullptr;
}
RtpDataChannel* data_channel = new RtpDataChannel(
worker_thread_, network_thread_, signaling_thread, media_channel,
content_name, rtcp_mux_required, srtp_required);
data_channel->SetCryptoOptions(crypto_options_);
if (!data_channel->Init_w(rtp_transport, rtcp_transport)) {
LOG(LS_WARNING) << "Failed to init data channel.";
delete data_channel;
return nullptr;
}
data_channels_.push_back(data_channel);
return data_channel;
}
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");
// Destroy data channel.
RTC_DCHECK(initialized_);
RtpDataChannels::iterator it =
std::find(data_channels_.begin(), data_channels_.end(), data_channel);
RTC_DCHECK(it != data_channels_.end());
if (it == data_channels_.end())
return;
data_channels_.erase(it);
delete data_channel;
}
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