blob: 2ba16927783173414cc20263ed9a76932e5fedb4 [file] [log] [blame]
/*
* Copyright (c) 2013 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 <assert.h>
#include <string.h>
#include <map>
#include <vector>
#include "webrtc/call.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/video/video_receive_stream.h"
#include "webrtc/video/video_send_stream.h"
#include "webrtc/video_engine/include/vie_base.h"
#include "webrtc/video_engine/include/vie_codec.h"
#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
namespace webrtc {
namespace internal {
class Call : public webrtc::Call, public PacketReceiver {
public:
Call(webrtc::VideoEngine* video_engine, const Call::Config& config);
virtual ~Call();
virtual PacketReceiver* Receiver() OVERRIDE;
virtual std::vector<VideoCodec> GetVideoCodecs() OVERRIDE;
virtual VideoSendStream::Config GetDefaultSendConfig() OVERRIDE;
virtual VideoSendStream* CreateVideoSendStream(
const VideoSendStream::Config& config) OVERRIDE;
virtual void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream)
OVERRIDE;
virtual VideoReceiveStream::Config GetDefaultReceiveConfig() OVERRIDE;
virtual VideoReceiveStream* CreateVideoReceiveStream(
const VideoReceiveStream::Config& config) OVERRIDE;
virtual void DestroyVideoReceiveStream(
webrtc::VideoReceiveStream* receive_stream) OVERRIDE;
virtual uint32_t SendBitrateEstimate() OVERRIDE;
virtual uint32_t ReceiveBitrateEstimate() OVERRIDE;
virtual bool DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE;
private:
bool DeliverRtcp(const uint8_t* packet, size_t length);
bool DeliverRtp(const RTPHeader& header,
const uint8_t* packet,
size_t length);
Call::Config config_;
std::map<uint32_t, VideoReceiveStream*> receive_ssrcs_;
scoped_ptr<RWLockWrapper> receive_lock_;
std::map<uint32_t, VideoSendStream*> send_ssrcs_;
scoped_ptr<RWLockWrapper> send_lock_;
scoped_ptr<RtpHeaderParser> rtp_header_parser_;
webrtc::VideoEngine* video_engine_;
ViERTP_RTCP* rtp_rtcp_;
ViECodec* codec_;
DISALLOW_COPY_AND_ASSIGN(Call);
};
} // internal
class TraceDispatcher : public TraceCallback {
public:
TraceDispatcher()
: crit_(CriticalSectionWrapper::CreateCriticalSection()),
initialized_(false),
filter_(kTraceNone) {}
~TraceDispatcher() {
if (initialized_) {
Trace::ReturnTrace();
VideoEngine::SetTraceCallback(NULL);
}
}
virtual void Print(TraceLevel level,
const char* message,
int length) OVERRIDE {
CriticalSectionScoped lock(crit_.get());
for (std::map<Call*, Call::Config*>::iterator it = callbacks_.begin();
it != callbacks_.end();
++it) {
if ((level & it->second->trace_filter) != kTraceNone)
it->second->trace_callback->Print(level, message, length);
}
}
void RegisterCallback(Call* call, Call::Config* config) {
if (config->trace_callback == NULL)
return;
CriticalSectionScoped lock(crit_.get());
callbacks_[call] = config;
filter_ |= config->trace_filter;
if (filter_ != kTraceNone && !initialized_) {
initialized_ = true;
Trace::CreateTrace();
VideoEngine::SetTraceCallback(this);
}
VideoEngine::SetTraceFilter(filter_);
}
void DeregisterCallback(Call* call) {
CriticalSectionScoped lock(crit_.get());
callbacks_.erase(call);
filter_ = kTraceNone;
for (std::map<Call*, Call::Config*>::iterator it = callbacks_.begin();
it != callbacks_.end();
++it) {
filter_ |= it->second->trace_filter;
}
VideoEngine::SetTraceFilter(filter_);
}
private:
scoped_ptr<CriticalSectionWrapper> crit_;
bool initialized_;
unsigned int filter_;
std::map<Call*, Call::Config*> callbacks_;
};
namespace internal {
TraceDispatcher* global_trace_dispatcher = NULL;
} // internal
Call* Call::Create(const Call::Config& config) {
if (internal::global_trace_dispatcher == NULL) {
TraceDispatcher* dispatcher = new TraceDispatcher();
// TODO(pbos): Atomic compare and exchange.
if (internal::global_trace_dispatcher == NULL) {
internal::global_trace_dispatcher = dispatcher;
} else {
delete dispatcher;
}
}
VideoEngine* video_engine = VideoEngine::Create();
assert(video_engine != NULL);
return new internal::Call(video_engine, config);
}
namespace internal {
Call::Call(webrtc::VideoEngine* video_engine, const Call::Config& config)
: config_(config),
receive_lock_(RWLockWrapper::CreateRWLock()),
send_lock_(RWLockWrapper::CreateRWLock()),
rtp_header_parser_(RtpHeaderParser::Create()),
video_engine_(video_engine) {
assert(video_engine != NULL);
assert(config.send_transport != NULL);
global_trace_dispatcher->RegisterCallback(this, &config_);
rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine_);
assert(rtp_rtcp_ != NULL);
codec_ = ViECodec::GetInterface(video_engine_);
assert(codec_ != NULL);
}
Call::~Call() {
global_trace_dispatcher->DeregisterCallback(this);
codec_->Release();
rtp_rtcp_->Release();
webrtc::VideoEngine::Delete(video_engine_);
}
PacketReceiver* Call::Receiver() { return this; }
std::vector<VideoCodec> Call::GetVideoCodecs() {
std::vector<VideoCodec> codecs;
VideoCodec codec;
for (size_t i = 0; i < static_cast<size_t>(codec_->NumberOfCodecs()); ++i) {
if (codec_->GetCodec(static_cast<unsigned char>(i), codec) == 0) {
codecs.push_back(codec);
}
}
return codecs;
}
VideoSendStream::Config Call::GetDefaultSendConfig() {
VideoSendStream::Config config;
codec_->GetCodec(0, config.codec);
return config;
}
VideoSendStream* Call::CreateVideoSendStream(
const VideoSendStream::Config& config) {
assert(config.rtp.ssrcs.size() > 0);
assert(config.codec.numberOfSimulcastStreams == 0 ||
config.codec.numberOfSimulcastStreams == config.rtp.ssrcs.size());
VideoSendStream* send_stream = new VideoSendStream(
config_.send_transport, config_.overuse_detection, video_engine_, config);
WriteLockScoped write_lock(*send_lock_);
for (size_t i = 0; i < config.rtp.ssrcs.size(); ++i) {
assert(send_ssrcs_.find(config.rtp.ssrcs[i]) == send_ssrcs_.end());
send_ssrcs_[config.rtp.ssrcs[i]] = send_stream;
}
return send_stream;
}
void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
assert(send_stream != NULL);
VideoSendStream* send_stream_impl = NULL;
{
WriteLockScoped write_lock(*send_lock_);
for (std::map<uint32_t, VideoSendStream*>::iterator it =
send_ssrcs_.begin();
it != send_ssrcs_.end();
++it) {
if (it->second == static_cast<VideoSendStream*>(send_stream)) {
send_stream_impl = it->second;
send_ssrcs_.erase(it);
break;
}
}
}
assert(send_stream_impl != NULL);
delete send_stream_impl;
}
VideoReceiveStream::Config Call::GetDefaultReceiveConfig() {
return VideoReceiveStream::Config();
}
VideoReceiveStream* Call::CreateVideoReceiveStream(
const VideoReceiveStream::Config& config) {
VideoReceiveStream* receive_stream = new VideoReceiveStream(
video_engine_, config, config_.send_transport, config_.voice_engine);
WriteLockScoped write_lock(*receive_lock_);
assert(receive_ssrcs_.find(config.rtp.ssrc) == receive_ssrcs_.end());
receive_ssrcs_[config.rtp.ssrc] = receive_stream;
return receive_stream;
}
void Call::DestroyVideoReceiveStream(
webrtc::VideoReceiveStream* receive_stream) {
assert(receive_stream != NULL);
VideoReceiveStream* receive_stream_impl = NULL;
{
WriteLockScoped write_lock(*receive_lock_);
for (std::map<uint32_t, VideoReceiveStream*>::iterator it =
receive_ssrcs_.begin();
it != receive_ssrcs_.end();
++it) {
if (it->second == static_cast<VideoReceiveStream*>(receive_stream)) {
receive_stream_impl = it->second;
receive_ssrcs_.erase(it);
break;
}
}
}
assert(receive_stream_impl != NULL);
delete receive_stream_impl;
}
uint32_t Call::SendBitrateEstimate() {
// TODO(pbos): Return send-bitrate estimate
return 0;
}
uint32_t Call::ReceiveBitrateEstimate() {
// TODO(pbos): Return receive-bitrate estimate
return 0;
}
bool Call::DeliverRtcp(const uint8_t* packet, size_t length) {
// TODO(pbos): Figure out what channel needs it actually.
// Do NOT broadcast! Also make sure it's a valid packet.
bool rtcp_delivered = false;
{
ReadLockScoped read_lock(*receive_lock_);
for (std::map<uint32_t, VideoReceiveStream*>::iterator it =
receive_ssrcs_.begin();
it != receive_ssrcs_.end();
++it) {
if (it->second->DeliverRtcp(packet, length))
rtcp_delivered = true;
}
}
{
ReadLockScoped read_lock(*send_lock_);
for (std::map<uint32_t, VideoSendStream*>::iterator it =
send_ssrcs_.begin();
it != send_ssrcs_.end();
++it) {
if (it->second->DeliverRtcp(packet, length))
rtcp_delivered = true;
}
}
return rtcp_delivered;
}
bool Call::DeliverRtp(const RTPHeader& header,
const uint8_t* packet,
size_t length) {
VideoReceiveStream* receiver;
{
ReadLockScoped read_lock(*receive_lock_);
std::map<uint32_t, VideoReceiveStream*>::iterator it =
receive_ssrcs_.find(header.ssrc);
if (it == receive_ssrcs_.end()) {
// TODO(pbos): Log some warning, SSRC without receiver.
return false;
}
receiver = it->second;
}
return receiver->DeliverRtp(static_cast<const uint8_t*>(packet), length);
}
bool Call::DeliverPacket(const uint8_t* packet, size_t length) {
// TODO(pbos): ExtensionMap if there are extensions.
if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length)))
return DeliverRtcp(packet, length);
RTPHeader rtp_header;
if (!rtp_header_parser_->Parse(packet, static_cast<int>(length), &rtp_header))
return false;
return DeliverRtp(rtp_header, packet, length);
}
} // namespace internal
} // namespace webrtc