Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
index fe0e49f79e6a2df6194a307095fe8c40beb49062..5852a73f1982ee6f0c8876c428c75c4520e3cb68 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -1240,7 +1240,7 @@ size_t RTPSender::CreateRtpHeader(uint8_t* header, |
} |
uint16_t len = |
- BuildRTPHeaderExtension(header + rtp_header_length, marker_bit); |
+ BuildRtpHeaderExtension(header + rtp_header_length, marker_bit); |
if (len > 0) { |
header[0] |= 0x10; // Set extension bit. |
rtp_header_length += len; |
@@ -1255,17 +1255,19 @@ int32_t RTPSender::BuildRTPheader(uint8_t* data_buffer, |
int64_t capture_time_ms, |
bool timestamp_provided, |
bool inc_sequence_number) { |
+ return BuildRtpHeader(data_buffer, payload_type, marker_bit, |
+ capture_timestamp, capture_time_ms); |
+} |
+ |
+int32_t RTPSender::BuildRtpHeader(uint8_t* data_buffer, |
+ int8_t payload_type, |
+ bool marker_bit, |
+ uint32_t capture_timestamp, |
+ int64_t capture_time_ms) { |
assert(payload_type >= 0); |
rtc::CritScope lock(&send_critsect_); |
- if (timestamp_provided) { |
- timestamp_ = start_timestamp_ + capture_timestamp; |
- } else { |
- // Make a unique time stamp. |
- // We can't inc by the actual time, since then we increase the risk of back |
- // timing. |
- timestamp_++; |
- } |
+ timestamp_ = start_timestamp_ + capture_timestamp; |
last_timestamp_time_ms_ = clock_->TimeInMilliseconds(); |
uint32_t sequence_number = sequence_number_++; |
capture_time_ms_ = capture_time_ms; |
@@ -1274,7 +1276,7 @@ int32_t RTPSender::BuildRTPheader(uint8_t* data_buffer, |
timestamp_, sequence_number, csrcs_); |
} |
-uint16_t RTPSender::BuildRTPHeaderExtension(uint8_t* data_buffer, |
+uint16_t RTPSender::BuildRtpHeaderExtension(uint8_t* data_buffer, |
bool marker_bit) const { |
if (rtp_header_extension_map_.Size() <= 0) { |
return 0; |