zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Use size_t more consistently for packet/payload lengths.

Browse Source 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pkasting@chromium.org
2014-11-20 22:28:14 +00:00
parent edc6e57a92
commit 4591fbd09f
341 changed files with 2610 additions and 2613 deletions
Download Patch File Download Diff File Expand all files Collapse all files

23
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
View File

@ -233,11 +233,11 @@ int32_t RTPSenderAudio::SendAudio(
const int8_t payloadType,
const uint32_t captureTimeStamp,
const uint8_t* payloadData,
const uint32_t dataSize,
const size_t dataSize,
const RTPFragmentationHeader* fragmentation) {
// TODO(pwestin) Breakup function in smaller functions.
uint16_t payloadSize = static_cast<uint16_t>(dataSize);
uint16_t maxPayloadLength = _rtpSender->MaxPayloadLength();
size_t payloadSize = dataSize;
size_t maxPayloadLength = _rtpSender->MaxPayloadLength();
bool dtmfToneStarted = false;
uint16_t dtmfLengthMS = 0;
uint8_t key = 0;
@ -383,7 +383,7 @@ int32_t RTPSenderAudio::SendAudio(
// only 0x80 if we have multiple blocks
dataBuffer[rtpHeaderLength++] = 0x80 +
fragmentation->fragmentationPlType[1];
uint32_t blockLength = fragmentation->fragmentationLength[1];
size_t blockLength = fragmentation->fragmentationLength[1];
// sanity blockLength
if(blockLength > 0x3ff) { // block length 10 bits 1023 bytes
@ -406,9 +406,8 @@ int32_t RTPSenderAudio::SendAudio(
payloadData + fragmentation->fragmentationOffset[0],
fragmentation->fragmentationLength[0]);
payloadSize = static_cast<uint16_t>(
fragmentation->fragmentationLength[0] +
fragmentation->fragmentationLength[1]);
payloadSize = fragmentation->fragmentationLength[0] +
fragmentation->fragmentationLength[1];
} else {
// silence for too long send only new data
dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
@ -416,8 +415,7 @@ int32_t RTPSenderAudio::SendAudio(
payloadData + fragmentation->fragmentationOffset[0],
fragmentation->fragmentationLength[0]);
payloadSize = static_cast<uint16_t>(
fragmentation->fragmentationLength[0]);
payloadSize = fragmentation->fragmentationLength[0];
}
} else {
if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
@ -427,8 +425,7 @@ int32_t RTPSenderAudio::SendAudio(
payloadData + fragmentation->fragmentationOffset[0],
fragmentation->fragmentationLength[0]);
payloadSize = static_cast<uint16_t>(
fragmentation->fragmentationLength[0]);
payloadSize = fragmentation->fragmentationLength[0];
} else {
memcpy(dataBuffer+rtpHeaderLength, payloadData, payloadSize);
}
@ -437,7 +434,7 @@ int32_t RTPSenderAudio::SendAudio(
// Update audio level extension, if included.
{
uint16_t packetSize = payloadSize + rtpHeaderLength;
size_t packetSize = payloadSize + rtpHeaderLength;
RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
RTPHeader rtp_header;
rtp_parser.Parse(rtp_header);
@ -451,7 +448,7 @@ int32_t RTPSenderAudio::SendAudio(
"seqnum", _rtpSender->SequenceNumber());
return _rtpSender->SendToNetwork(dataBuffer,
payloadSize,
static_cast<uint16_t>(rtpHeaderLength),
rtpHeaderLength,
-1,
kAllowRetransmission,
PacedSender::kHighPriority);