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Some refactoring inside rtp_rtcp/.

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Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2014-07-08 12:10:51 +00:00
parent 241a9b0b65
commit 62bafae661
45 changed files with 384 additions and 400 deletions
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11
webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
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@ -133,11 +133,12 @@ void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header,
void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
uint32_t receive_time_secs,
uint32_t receive_time_frac) {
uint32_t receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP(
receive_time_secs, receive_time_frac, header.payload_type_frequency);
uint32_t last_receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP(
last_receive_time_secs_, last_receive_time_frac_,
header.payload_type_frequency);
uint32_t receive_time_rtp = RtpUtility::ConvertNTPTimeToRTP(
receive_time_secs, receive_time_frac, header.payload_type_frequency);
uint32_t last_receive_time_rtp =
RtpUtility::ConvertNTPTimeToRTP(last_receive_time_secs_,
last_receive_time_frac_,
header.payload_type_frequency);
int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
(header.timestamp - last_received_timestamp_);