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Fix mismatch between different NACK list lengths and packet buffers.

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This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors.

BUG=1289

Review URL: https://webrtc-codereview.appspot.com/1065007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2013-02-01 15:09:57 +00:00
parent b586507986
commit becf9c897c
36 changed files with 316 additions and 130 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/video_coding/main/test/jitter_buffer_test.cc
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@ -106,7 +106,7 @@ int JitterBufferTest(CmdArgs& args)
WebRtc_UWord8 data[1500];
VCMPacket packet(data, size, seqNum, timeStamp, true);
VCMJitterBuffer jb(clock);
VCMJitterBuffer jb(clock, -1, -1, true);
seqNum = 1234;
timeStamp = 123*90;

2
webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc
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@ -239,7 +239,7 @@ int MTRxTxTest(CmdArgs& args)
FecProtectionParams delta_params = protectionCallback.DeltaFecParameters();
FecProtectionParams key_params = protectionCallback.KeyFecParameters();
rtp->SetFecParameters(&delta_params, &key_params);
rtp->SetNACKStatus(nackEnabled ? kNackRtcp : kNackOff);
rtp->SetNACKStatus(nackEnabled ? kNackRtcp : kNackOff, kMaxPacketAgeToNack);
vcm->SetChannelParameters((WebRtc_UWord32) bitRate,
(WebRtc_UWord8) lossRate, rttMS);

10
webrtc/modules/video_coding/main/test/rtp_player.cc
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "rtp_player.h"
#include "webrtc/modules/video_coding/main/test/rtp_player.h"
#include <cstdlib>
#ifdef WIN32
@ -18,9 +18,10 @@
#include <arpa/inet.h>
#endif
#include "../source/internal_defines.h"
#include "gtest/gtest.h"
#include "rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/video_coding/main/source/internal_defines.h"
#include "webrtc/modules/video_coding/main/test/test_util.h"
#include "webrtc/system_wrappers/interface/clock.h"
using namespace webrtc;
@ -191,7 +192,8 @@ WebRtc_Word32 RTPPlayer::Initialize(const PayloadTypeList* payloadList)
_randVec[i] = rand();
}
_randVecPos = 0;
WebRtc_Word32 ret = _rtpModule->SetNACKStatus(kNackOff);
WebRtc_Word32 ret = _rtpModule->SetNACKStatus(kNackOff,
kMaxPacketAgeToNack);
if (ret < 0)
{
return -1;

3
webrtc/modules/video_coding/main/test/test_util.h
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@ -22,6 +22,9 @@
#include "module_common_types.h"
#include "testsupport/fileutils.h"
enum { kMaxNackListSize = 250 };
enum { kMaxPacketAgeToNack = 450 };
// Class used for passing command line arguments to tests
class CmdArgs
{

1
webrtc/modules/video_coding/main/test/video_rtp_play.cc
View File

@ -194,6 +194,7 @@ int RtpPlay(CmdArgs& args)
vcm->SetVideoProtection(protectionMethod, protectionEnabled);
vcm->SetRenderDelay(renderDelayMs);
vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
vcm->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack);
ret = 0;

4
webrtc/modules/video_coding/main/test/video_rtp_play_mt.cc
View File

@ -84,8 +84,7 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
protection == kProtectionNack ||
kProtectionNackFEC));
Clock* clock = Clock::GetRealTimeClock();
VideoCodingModule* vcm =
VideoCodingModule::Create(1, clock);
VideoCodingModule* vcm = VideoCodingModule::Create(1, clock);
RtpDataCallback dataCallback(vcm);
std::string rtpFilename;
rtpFilename = args.inputFile;
@ -227,6 +226,7 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
vcm->SetVideoProtection(protection, protectionEnabled);
vcm->SetRenderDelay(renderDelayMs);
vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
vcm->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack);
EventWrapper& waitEvent = *EventWrapper::Create();