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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

18
webrtc/modules/video_coding/main/test/rtp_player.cc
View File

@ -41,7 +41,7 @@ enum {
class RawRtpPacket {
public:
RawRtpPacket(const uint8_t* data, uint32_t length, uint32_t ssrc,
RawRtpPacket(const uint8_t* data, size_t length, uint32_t ssrc,
uint16_t seq_num)
: data_(new uint8_t[length]),
length_(length),
@ -53,7 +53,7 @@ class RawRtpPacket {
}
const uint8_t* data() const { return data_.get(); }
uint32_t length() const { return length_; }
size_t length() const { return length_; }
int64_t resend_time_ms() const { return resend_time_ms_; }
void set_resend_time_ms(int64_t timeMs) { resend_time_ms_ = timeMs; }
uint32_t ssrc() const { return ssrc_; }
@ -61,7 +61,7 @@ class RawRtpPacket {
private:
scoped_ptr<uint8_t[]> data_;
uint32_t length_;
size_t length_;
int64_t resend_time_ms_;
uint32_t ssrc_;
uint16_t seq_num_;
@ -251,7 +251,7 @@ class SsrcHandlers {
return 0;
}
void IncomingPacket(const uint8_t* data, uint32_t length) {
void IncomingPacket(const uint8_t* data, size_t length) {
for (HandlerMapIt it = handlers_.begin(); it != handlers_.end(); ++it) {
if (!it->second->rtp_header_parser_->IsRtcp(data, length)) {
RTPHeader header;
@ -375,14 +375,10 @@ class RtpPlayerImpl : public RtpPlayerInterface {
if (reordering_ && reorder_buffer_.get() == NULL) {
reorder_buffer_.reset(
new RawRtpPacket(next_packet_.data,
static_cast<uint32_t>(next_packet_.length),
0,
0));
new RawRtpPacket(next_packet_.data, next_packet_.length, 0, 0));
return 0;
}
int ret = SendPacket(next_packet_.data,
static_cast<uint32_t>(next_packet_.length));
int ret = SendPacket(next_packet_.data, next_packet_.length);
if (reorder_buffer_.get()) {
SendPacket(reorder_buffer_->data(), reorder_buffer_->length());
reorder_buffer_.reset(NULL);
@ -421,7 +417,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
}
private:
int SendPacket(const uint8_t* data, uint32_t length) {
int SendPacket(const uint8_t* data, size_t length) {
assert(data);
assert(length > 0);