Peter Boström ac547a6538 Remove channel ids from various interfaces.
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.

IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately

BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1335353005 .

Cr-Commit-Position: refs/heads/master@{#9978}
2015-09-17 21:06:02 +00:00

68 lines
2.3 KiB
C++

/*
* Copyright (c) 2012 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 "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
namespace webrtc {
// This class sends all its packet straight to the provided RtpRtcp module.
// with optional packet loss.
class LoopBackTransport : public webrtc::Transport {
public:
LoopBackTransport()
: count_(0),
packet_loss_(0),
rtp_payload_registry_(NULL),
rtp_receiver_(NULL),
rtp_rtcp_module_(NULL) {}
void SetSendModule(RtpRtcp* rtp_rtcp_module,
RTPPayloadRegistry* payload_registry,
RtpReceiver* receiver,
ReceiveStatistics* receive_statistics);
void DropEveryNthPacket(int n);
int SendPacket(const void* data, size_t len) override;
int SendRTCPPacket(const void* data, size_t len) override;
private:
int count_;
int packet_loss_;
ReceiveStatistics* receive_statistics_;
RTPPayloadRegistry* rtp_payload_registry_;
RtpReceiver* rtp_receiver_;
RtpRtcp* rtp_rtcp_module_;
};
class TestRtpReceiver : public NullRtpData {
public:
int32_t OnReceivedPayloadData(
const uint8_t* payload_data,
const size_t payload_size,
const webrtc::WebRtcRTPHeader* rtp_header) override;
const uint8_t* payload_data() const { return payload_data_; }
size_t payload_size() const { return payload_size_; }
webrtc::WebRtcRTPHeader rtp_header() const { return rtp_header_; }
private:
uint8_t payload_data_[1500];
size_t payload_size_;
webrtc::WebRtcRTPHeader rtp_header_;
};
} // namespace webrtc