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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

8
webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
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@ -184,7 +184,7 @@ int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
const PayloadUnion& specific_payload,
bool is_red,
const uint8_t* payload,
uint16_t payload_length,
size_t payload_length,
int64_t timestamp_ms,
bool is_first_packet) {
TRACE_EVENT2("webrtc_rtp", "Audio::ParseRtp",
@ -288,7 +288,7 @@ int32_t RTPReceiverAudio::InvokeOnInitializeDecoder(
int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
WebRtcRTPHeader* rtp_header,
const uint8_t* payload_data,
uint16_t payload_length,
size_t payload_length,
const AudioPayload& audio_specific,
bool is_red) {
@ -311,13 +311,13 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
if (payload_length % 4 != 0) {
return -1;
}
uint8_t number_of_events = payload_length / 4;
size_t number_of_events = payload_length / 4;
// sanity
if (number_of_events >= MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS) {
number_of_events = MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS;
}
for (int n = 0; n < number_of_events; ++n) {
for (size_t n = 0; n < number_of_events; ++n) {
bool end = (payload_data[(4 * n) + 1] & 0x80) ? true : false;
std::set<uint8_t>::iterator event =