webrtc_m130/pc/channelmanager.cc
Benjamin Wright a54daf162f Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h"
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.

Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:

void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);

In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.

This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.

Reland Fix:
- cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional
                    root level configuration.
- peerconnectionfactory - If this optional is set will now overwrite the
                          underyling value.

This along with the other field will be deprecated once dependent projects
are updated.

TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org

Bug: webrtc:9681
Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d
Reviewed-on: https://webrtc-review.googlesource.com/c/105560
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 23:09:07 +00:00

359 lines
11 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 "absl/memory/memory.h"
#include "media/base/rtpdataengine.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.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,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
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,
crypto_options, 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 = absl::make_unique<VoiceChannel>(
worker_thread_, network_thread_, signaling_thread, media_engine_.get(),
absl::WrapUnique(media_channel), content_name, srtp_required,
crypto_options);
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;
}
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,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
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,
crypto_options, 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 = absl::make_unique<VideoChannel>(
worker_thread_, network_thread_, signaling_thread,
absl::WrapUnique(media_channel), content_name, srtp_required,
crypto_options);
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;
}
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,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options) {
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<RtpDataChannel*>(RTC_FROM_HERE, [&] {
return CreateRtpDataChannel(media_config, rtp_transport, signaling_thread,
content_name, srtp_required, crypto_options);
});
}
// 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 = absl::make_unique<RtpDataChannel>(
worker_thread_, network_thread_, signaling_thread,
absl::WrapUnique(media_channel), content_name, srtp_required,
crypto_options);
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