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Break out glue for old->new Transport.

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Reduces multiple inheritance and code duplication.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4774 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2013-09-18 11:52:42 +00:00
parent fe84fda488
commit e75a1bf45f
8 changed files with 84 additions and 60 deletions
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4
webrtc/video_engine/internal/call.h
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@ -7,8 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_VIDEO_ENGINE_CALL_IMPL_H_
#define WEBRTC_VIDEO_ENGINE_CALL_IMPL_H_
#ifndef WEBRTC_VIDEO_ENGINE_INTERNAL_CALL_H_
#define WEBRTC_VIDEO_ENGINE_INTERNAL_CALL_H_
#include <map>
#include <vector>

36
webrtc/video_engine/internal/transport_adapter.cc Normal file
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@ -0,0 +1,36 @@
/*
* 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 "webrtc/video_engine/internal/transport_adapter.h"
namespace webrtc {
namespace internal {
TransportAdapter::TransportAdapter(newapi::Transport* transport)
: transport_(transport) {}
int TransportAdapter::SendPacket(int /*channel*/,
const void* packet,
int length) {
bool success = transport_->SendRTP(static_cast<const uint8_t*>(packet),
static_cast<size_t>(length));
return success ? length : -1;
}
int TransportAdapter::SendRTCPPacket(int /*channel*/,
const void* packet,
int length) {
bool success = transport_->SendRTCP(static_cast<const uint8_t*>(packet),
static_cast<size_t>(length));
return success ? length : -1;
}
} // namespace internal
} // namespace webrtc

34
webrtc/video_engine/internal/transport_adapter.h Normal file
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@ -0,0 +1,34 @@
/*
* 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.
*/
#ifndef WEBRTC_VIDEO_ENGINE_INTERNAL_TRANSPORT_ADAPTER_H_
#define WEBRTC_VIDEO_ENGINE_INTERNAL_TRANSPORT_ADAPTER_H_
#include "webrtc/common_types.h"
#include "webrtc/video_engine/new_include/transport.h"
namespace webrtc {
namespace internal {
class TransportAdapter : public webrtc::Transport {
public:
explicit TransportAdapter(newapi::Transport* transport);
virtual int SendPacket(int /*channel*/, const void* packet, int length)
OVERRIDE;
virtual int SendRTCPPacket(int /*channel*/, const void* packet, int length)
OVERRIDE;
private:
newapi::Transport *transport_;
};
} // namespace internal
} // namespace webrtc
#endif // WEBRTC_VIDEO_ENGINE_INTERNAL_TRANSPORT_ADAPTER_H_

21
webrtc/video_engine/internal/video_receive_stream.cc
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@ -30,7 +30,7 @@ namespace internal {
VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
const VideoReceiveStream::Config& config,
newapi::Transport* transport)
: transport_(transport), config_(config), channel_(-1) {
: transport_adapter_(transport), config_(config), channel_(-1) {
video_engine_base_ = ViEBase::GetInterface(video_engine);
// TODO(mflodman): Use the other CreateChannel method.
video_engine_base_->CreateChannel(channel_);
@ -48,7 +48,7 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
network_ = ViENetwork::GetInterface(video_engine);
assert(network_ != NULL);
network_->RegisterSendTransport(channel_, *this);
network_->RegisterSendTransport(channel_, transport_adapter_);
codec_ = ViECodec::GetInterface(video_engine);
@ -177,22 +177,5 @@ int VideoReceiveStream::DeliverFrame(uint8_t* frame,
bool VideoReceiveStream::IsTextureSupported() { return false; }
int VideoReceiveStream::SendPacket(int /*channel*/,
const void* packet,
int length) {
assert(length >= 0);
bool success = transport_->SendRTP(static_cast<const uint8_t*>(packet),
static_cast<size_t>(length));
return success ? 0 : -1;
}
int VideoReceiveStream::SendRTCPPacket(int /*channel*/,
const void* packet,
int length) {
assert(length >= 0);
bool success = transport_->SendRTCP(static_cast<const uint8_t*>(packet),
static_cast<size_t>(length));
return success ? 0 : -1;
}
} // internal
} // webrtc

11
webrtc/video_engine/internal/video_receive_stream.h
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@ -16,6 +16,7 @@
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/video_engine/include/vie_render.h"
#include "webrtc/video_engine/internal/transport_adapter.h"
#include "webrtc/video_engine/new_include/video_receive_stream.h"
namespace webrtc {
@ -31,8 +32,7 @@ class ViERTP_RTCP;
namespace internal {
class VideoReceiveStream : public webrtc::VideoReceiveStream,
public webrtc::ExternalRenderer,
public webrtc::Transport {
public webrtc::ExternalRenderer {
public:
VideoReceiveStream(webrtc::VideoEngine* video_engine,
const VideoReceiveStream::Config& config,
@ -51,17 +51,12 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
virtual bool IsTextureSupported() OVERRIDE;
virtual int SendPacket(int /*channel*/, const void* packet, int length)
OVERRIDE;
virtual int SendRTCPPacket(int /*channel*/, const void* packet, int length)
OVERRIDE;
public:
virtual bool DeliverRtcp(const uint8_t* packet, size_t length);
virtual bool DeliverRtp(const uint8_t* packet, size_t length);
private:
newapi::Transport* transport_;
TransportAdapter transport_adapter_;
VideoReceiveStream::Config config_;
Clock* clock_;

24
webrtc/video_engine/internal/video_send_stream.cc
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@ -80,7 +80,7 @@ VideoSendStream::VideoSendStream(newapi::Transport* transport,
bool overuse_detection,
webrtc::VideoEngine* video_engine,
const VideoSendStream::Config& config)
: transport_(transport), config_(config), external_codec_(NULL) {
: transport_adapter_(transport), config_(config), external_codec_(NULL) {
if (config_.codec.numberOfSimulcastStreams > 0) {
assert(config_.rtp.ssrcs.size() == config_.codec.numberOfSimulcastStreams);
@ -146,7 +146,7 @@ VideoSendStream::VideoSendStream(newapi::Transport* transport,
network_ = ViENetwork::GetInterface(video_engine);
assert(network_ != NULL);
network_->RegisterSendTransport(channel_, *this);
network_->RegisterSendTransport(channel_, transport_adapter_);
if (config.encoder) {
external_codec_ = ViEExternalCodec::GetInterface(video_engine);
@ -250,26 +250,6 @@ void VideoSendStream::GetSendCodec(VideoCodec* send_codec) {
*send_codec = config_.codec;
}
int VideoSendStream::SendPacket(int /*channel*/,
const void* packet,
int length) {
// TODO(pbos): Lock these methods and the destructor so it can't be processing
// a packet when the destructor has been called.
assert(length >= 0);
bool success = transport_->SendRTP(static_cast<const uint8_t*>(packet),
static_cast<size_t>(length));
return success ? length : -1;
}
int VideoSendStream::SendRTCPPacket(int /*channel*/,
const void* packet,
int length) {
assert(length >= 0);
bool success = transport_->SendRTCP(static_cast<const uint8_t*>(packet),
static_cast<size_t>(length));
return success ? length : -1;
}
bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
return network_->ReceivedRTCPPacket(
channel_, packet, static_cast<int>(length)) == 0;

12
webrtc/video_engine/internal/video_send_stream.h
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@ -14,6 +14,7 @@
#include <vector>
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/video_engine/internal/transport_adapter.h"
#include "webrtc/video_engine/new_include/video_receive_stream.h"
#include "webrtc/video_engine/new_include/video_send_stream.h"
@ -33,8 +34,7 @@ namespace internal {
class ResolutionAdaptor;
class VideoSendStream : public webrtc::VideoSendStream,
public VideoSendStreamInput,
public webrtc::Transport {
public VideoSendStreamInput {
public:
VideoSendStream(newapi::Transport* transport,
bool overuse_detection,
@ -58,17 +58,11 @@ class VideoSendStream : public webrtc::VideoSendStream,
virtual void GetSendCodec(VideoCodec* send_codec) OVERRIDE;
virtual int SendPacket(int /*channel*/, const void* packet, int length)
OVERRIDE;
virtual int SendRTCPPacket(int /*channel*/, const void* packet, int length)
OVERRIDE;
public:
bool DeliverRtcp(const uint8_t* packet, size_t length);
private:
newapi::Transport* transport_;
TransportAdapter transport_adapter_;
VideoSendStream::Config config_;
ViEBase* video_engine_base_;

2
webrtc/video_engine/video_engine_core.gypi
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@ -117,6 +117,8 @@
# New VideoEngine API
'internal/call.cc',
'internal/call.h',
'internal/transport_adapter.cc',
'internal/transport_adapter.h',
'internal/video_receive_stream.cc',
'internal/video_receive_stream.h',
'internal/video_send_stream.cc',