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Review URL: http://webrtc-codereview.appspot.com/109006

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@383 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
hellner@google.com 2011-08-16 17:30:30 +00:00
parent 5daeae2e5f
commit 977c2966fc
10 changed files with 18 additions and 17 deletions
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1
src/modules/rtp_rtcp/source/forward_error_correction.cc
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@ -9,7 +9,6 @@
*/ */
#include "forward_error_correction.h" #include "forward_error_correction.h"
#include "fec_private_tables.h"
#include "rtp_utility.h" #include "rtp_utility.h"
#include "trace.h" #include "trace.h"

6
src/modules/rtp_rtcp/source/remote_rate_control.cc
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@ -236,10 +236,12 @@ WebRtc_UWord32 RemoteRateControl::ChangeBitRate(WebRtc_UWord32 currentBitRate,
#ifdef _DEBUG #ifdef _DEBUG
//char logStr[256]; //char logStr[256];
#ifdef _WIN32 #ifdef _WIN32
_snprintf(logStr,256, "New bitRate: %lu\n", currentBitRate / 1000); _snprintf(logStr,256, "New bitRate: %lu\n",
static_cast<long unsigned int> (currentBitRate / 1000));
OutputDebugStringA(logStr); OutputDebugStringA(logStr);
#else #else
snprintf(logStr,256, "New bitRate: %lu\n", currentBitRate / 1000); snprintf(logStr,256, "New bitRate: %lu\n",
static_cast<long unsigned int> (currentBitRate / 1000));
//TODO //TODO
#endif #endif
#endif #endif

2
src/modules/rtp_rtcp/source/rtp_receiver_video.cc
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@ -35,11 +35,11 @@ RTPReceiverVideo::RTPReceiverVideo(const WebRtc_Word32 id,
_criticalSectionReceiverVideo(*CriticalSectionWrapper::CreateCriticalSection()), _criticalSectionReceiverVideo(*CriticalSectionWrapper::CreateCriticalSection()),
_completeFrame(false), _completeFrame(false),
_receiveFEC(NULL),
_packetStartTimeMs(0), _packetStartTimeMs(0),
_receivedBW(), _receivedBW(),
_estimatedBW(0), _estimatedBW(0),
_currentFecFrameDecoded(false), _currentFecFrameDecoded(false),
_receiveFEC(NULL),
_h263InverseLogic(false), _h263InverseLogic(false),
_overUseDetector(), _overUseDetector(),
_videoBitRate(), _videoBitRate(),

4
src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -2187,10 +2187,10 @@ ModuleRtpRtcpImpl::OnBandwidthEstimateUpdate(WebRtc_UWord16 bandWidthKbit)
extensions. Overhead excludes any RTP payload headers and the extensions. Overhead excludes any RTP payload headers and the
payload itself. payload itself.
*/ */
WebRtc_UWord16 RTPpacketOH = _rtpReceiver.PacketOHReceived(); _rtpReceiver.PacketOHReceived();
// call RequestTMMBR when our localy created estimate changes // call RequestTMMBR when our localy created estimate changes
_rtcpSender.RequestTMMBR(bandWidthKbit, 0/*RTPpacketOH + _packetOverHead*/); _rtcpSender.RequestTMMBR(bandWidthKbit, 0);
} }
} }

2
src/modules/rtp_rtcp/source/rtp_sender_audio.cc
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@ -87,7 +87,7 @@ RTPSenderAudio::SetAudioFrequency(const WebRtc_UWord32 f)
_frequency = f; _frequency = f;
} }
WebRtc_UWord32 int
RTPSenderAudio::AudioFrequency() const RTPSenderAudio::AudioFrequency() const
{ {
CriticalSectionScoped cs(_sendAudioCritsect); CriticalSectionScoped cs(_sendAudioCritsect);

2
src/modules/rtp_rtcp/source/rtp_sender_audio.h
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@ -71,7 +71,7 @@ public:
void SetAudioFrequency(const WebRtc_UWord32 f); void SetAudioFrequency(const WebRtc_UWord32 f);
WebRtc_UWord32 AudioFrequency() const; int AudioFrequency() const;
// Set payload type for Redundant Audio Data RFC 2198 // Set payload type for Redundant Audio Data RFC 2198
WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType); WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType);

4
src/modules/rtp_rtcp/source/rtp_sender_video.cc
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@ -170,7 +170,7 @@ RTPSenderVideo::SendVideoPacket(const FrameType frameType,
// Add packet to FEC list // Add packet to FEC list
_rtpPacketListFec.PushBack(ptrGenericFEC); _rtpPacketListFec.PushBack(ptrGenericFEC);
// FEC can only protect up to kMaxMediaPackets packets // FEC can only protect up to kMaxMediaPackets packets
if (_mediaPacketListFec.GetSize() < if (static_cast<int>(_mediaPacketListFec.GetSize()) <
ForwardErrorCorrection::kMaxMediaPackets) ForwardErrorCorrection::kMaxMediaPackets)
{ {
_mediaPacketListFec.PushBack(ptrGenericFEC->pkt); _mediaPacketListFec.PushBack(ptrGenericFEC->pkt);
@ -195,7 +195,7 @@ RTPSenderVideo::SendVideoPacket(const FrameType frameType,
// Number of first partition packets cannot exceed kMaxMediaPackets // Number of first partition packets cannot exceed kMaxMediaPackets
if (_numberFirstPartition > if (_numberFirstPartition >
ForwardErrorCorrection::kMaxMediaPackets) ForwardErrorCorrection::kMaxMediaPackets)
{ {
_numberFirstPartition = _numberFirstPartition =
ForwardErrorCorrection::kMaxMediaPackets; ForwardErrorCorrection::kMaxMediaPackets;

2
src/modules/rtp_rtcp/source/rtp_sender_video.h
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@ -160,7 +160,7 @@ private:
bool _useUepProtectionDelta; bool _useUepProtectionDelta;
WebRtc_UWord8 _fecProtectionFactor; WebRtc_UWord8 _fecProtectionFactor;
bool _fecUseUepProtection; bool _fecUseUepProtection;
WebRtc_UWord32 _numberFirstPartition; int _numberFirstPartition;
ListWrapper _mediaPacketListFec; ListWrapper _mediaPacketListFec;
ListWrapper _rtpPacketListFec; ListWrapper _rtpPacketListFec;

8
src/modules/rtp_rtcp/source/rtp_utility.cc
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@ -567,11 +567,11 @@ ModuleRTPUtility::RTPHeaderParser::Parse(WebRtcRTPHeader& parsedPacket) const
*/ */
// Parse out the fields but only use it for debugging for now. // Parse out the fields but only use it for debugging for now.
const WebRtc_UWord8 ID = (*ptr & 0xf0) >> 4; //const WebRtc_UWord8 ID = (*ptr & 0xf0) >> 4;
const WebRtc_UWord8 len = (*ptr & 0x0f); //const WebRtc_UWord8 len = (*ptr & 0x0f);
ptr++; ptr++;
const WebRtc_UWord8 V = (*ptr & 0x80) >> 7; //const WebRtc_UWord8 V = (*ptr & 0x80) >> 7;
const WebRtc_UWord8 level = (*ptr & 0x7f); //const WebRtc_UWord8 level = (*ptr & 0x7f);
// DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u, level=%u", ID, len, V, level); // DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u, level=%u", ID, len, V, level);
} }
parsedPacket.header.headerLength += XLen; parsedPacket.header.headerLength += XLen;