pbos@webrtc.org 6488761f2e Implement VideoSendStream::SetCodec().
Removing assertion that SSRC count should be the same as the number of
streams in the codec. It makes sense that you don't always use the same
number of streams under one call. Dropping resolution due to CPU overuse
for instance can require less streams, but the SSRCs should stay
allocated so that operations can resume when not overusing any more.

This change also means we can get rid of the ugly SendStreamState whose
content wasn't defined. Instead we use SetCodec to change resolution
etc. on the fly. Should something else have to be replaced on the fly
then that functionality simply has to be implemented.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-14 08:58:14 +00:00

359 lines
10 KiB
C++

/*
* 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* CreateSendStream(
const VideoSendStream::Config& config) OVERRIDE;
virtual void DestroySendStream(webrtc::VideoSendStream* send_stream) OVERRIDE;
virtual VideoReceiveStream::Config GetDefaultReceiveConfig() OVERRIDE;
virtual VideoReceiveStream* CreateReceiveStream(
const VideoReceiveStream::Config& config) OVERRIDE;
virtual void DestroyReceiveStream(
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::CreateSendStream(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::DestroySendStream(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::CreateReceiveStream(
const VideoReceiveStream::Config& config) {
VideoReceiveStream* receive_stream =
new VideoReceiveStream(video_engine_, config, config_.send_transport);
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::DestroyReceiveStream(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