7d8c72e2db
This change basicly re-enables the change of r1220, which was reverted in r1235 due to Clang issues. The difference from r1220 is that the TickTimeInterface was renamed to TickTimeClass, and no longer inherits from TickTime. Review URL: http://webrtc-codereview.appspot.com/335006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1267 4adac7df-926f-26a2-2b94-8c16560cd09d
140 lines
3.5 KiB
C++
140 lines
3.5 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "mt_test_common.h"
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#include <cmath>
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#include "modules/video_coding/main/source/tick_time_base.h"
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#include "rtp_dump.h"
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namespace webrtc {
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TransportCallback::TransportCallback(webrtc::RtpRtcp* rtp,
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TickTimeBase* clock,
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const char* filename):
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RTPSendCompleteCallback(rtp, clock, filename)
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{
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//
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}
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TransportCallback::~TransportCallback()
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{
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//
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}
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int
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TransportCallback::SendPacket(int channel, const void *data, int len)
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{
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_sendCount++;
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_totalSentLength += len;
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if (_rtpDump != NULL)
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{
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if (_rtpDump->DumpPacket((const WebRtc_UWord8*)data, len) != 0)
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{
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return -1;
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}
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}
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bool transmitPacket = true;
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// Off-line tests, don't drop first Key frame (approx.)
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if (_sendCount > 20)
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{
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transmitPacket = PacketLoss();
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}
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TickTimeBase clock;
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int64_t now = clock.MillisecondTimestamp();
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// Insert outgoing packet into list
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if (transmitPacket)
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{
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rtpPacket* newPacket = new rtpPacket();
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memcpy(newPacket->data, data, len);
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newPacket->length = len;
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// Simulate receive time = network delay + packet jitter
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// simulated as a Normal distribution random variable with
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// mean = networkDelay and variance = jitterVar
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WebRtc_Word32
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simulatedDelay = (WebRtc_Word32)NormalDist(_networkDelayMs,
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sqrt(_jitterVar));
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newPacket->receiveTime = now + simulatedDelay;
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_rtpPackets.PushBack(newPacket);
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}
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return 0;
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}
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int
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TransportCallback::TransportPackets()
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{
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// Are we ready to send packets to the receiver?
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rtpPacket* packet = NULL;
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TickTimeBase clock;
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int64_t now = clock.MillisecondTimestamp();
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while (!_rtpPackets.Empty())
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{
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// Take first packet in list
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packet = static_cast<rtpPacket*>((_rtpPackets.First())->GetItem());
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WebRtc_Word64 timeToReceive = packet->receiveTime - now;
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if (timeToReceive > 0)
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{
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// No available packets to send
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break;
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}
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_rtpPackets.PopFront();
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// Send to receive side
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if (_rtp->IncomingPacket((const WebRtc_UWord8*)packet->data,
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packet->length) < 0)
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{
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delete packet;
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packet = NULL;
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// Will return an error after the first packet that goes wrong
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return -1;
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}
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delete packet;
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packet = NULL;
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}
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return 0; // OK
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}
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bool VCMProcessingThread(void* obj)
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{
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SharedRTPState* state = static_cast<SharedRTPState*>(obj);
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if (state->_vcm.TimeUntilNextProcess() <= 0)
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{
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if (state->_vcm.Process() < 0)
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{
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return false;
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}
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}
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return true;
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}
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bool VCMDecodeThread(void* obj)
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{
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SharedRTPState* state = static_cast<SharedRTPState*>(obj);
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state->_vcm.Decode();
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return true;
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}
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bool TransportThread(void *obj)
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{
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SharedTransportState* state = static_cast<SharedTransportState*>(obj);
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state->_transport.TransportPackets();
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return true;
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}
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} // namespace webrtc
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