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/*
* 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 "webrtc/modules/video_coding/main/test/mt_test_common.h"
#include <math.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/utility/interface/rtp_dump.h"
#include "webrtc/system_wrappers/interface/clock.h"
namespace webrtc {
TransportCallback::TransportCallback(Clock* clock, const char* filename)
: RTPSendCompleteCallback(clock, filename) {
}
TransportCallback::~TransportCallback()
{
//
}
int
TransportCallback::SendPacket(int channel, const void *data, size_t len)
{
_sendCount++;
_totalSentLength += len;
if (_rtpDump != NULL)
{
if (_rtpDump->DumpPacket((const uint8_t*)data, len) != 0)
{
return -1;
}
}
bool transmitPacket = true;
// Off-line tests, don't drop first Key frame (approx.)
if (_sendCount > 20)
{
transmitPacket = PacketLoss();
}
Clock* clock = Clock::GetRealTimeClock();
int64_t now = clock->TimeInMilliseconds();
// Insert outgoing packet into list
if (transmitPacket)
{
RtpPacket* newPacket = new RtpPacket();
memcpy(newPacket->data, data, len);
newPacket->length = len;
// Simulate receive time = network delay + packet jitter
// simulated as a Normal distribution random variable with
// mean = networkDelay and variance = jitterVar
int32_t
simulatedDelay = (int32_t)NormalDist(_networkDelayMs,
sqrt(_jitterVar));
newPacket->receiveTime = now + simulatedDelay;
_rtpPackets.push_back(newPacket);
}
return 0;
}
int
TransportCallback::TransportPackets()
{
// Are we ready to send packets to the receiver?
RtpPacket* packet = NULL;
Clock* clock = Clock::GetRealTimeClock();
int64_t now = clock->TimeInMilliseconds();
while (!_rtpPackets.empty())
{
// Take first packet in list
packet = _rtpPackets.front();
int64_t timeToReceive = packet->receiveTime - now;
if (timeToReceive > 0)
{
// No available packets to send
break;
}
_rtpPackets.pop_front();
// Send to receive side
RTPHeader header;
scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
if (!parser->Parse(packet->data, packet->length, &header)) {
delete packet;
return -1;
}
PayloadUnion payload_specific;
if (!rtp_payload_registry_->GetPayloadSpecifics(
header.payloadType, &payload_specific)) {
return -1;
}
if (!rtp_receiver_->IncomingRtpPacket(header, packet->data,
packet->length, payload_specific,
true))
{
delete packet;
return -1;
}
delete packet;
packet = NULL;
}
return 0; // OK
}
bool VCMProcessingThread(void* obj)
{
SharedRTPState* state = static_cast<SharedRTPState*>(obj);
if (state->_vcm.TimeUntilNextProcess() <= 0)
{
if (state->_vcm.Process() < 0)
{
return false;
}
}
return true;
}
bool VCMDecodeThread(void* obj)
{
SharedRTPState* state = static_cast<SharedRTPState*>(obj);
state->_vcm.Decode();
return true;
}
bool TransportThread(void *obj)
{
SharedTransportState* state = static_cast<SharedTransportState*>(obj);
state->_transport.TransportPackets();
return true;
}
} // namespace webrtc
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