Index: webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
index d15de162d9607f06cabcaa4e31372dc1e4b68ce0..55ca9a72f98a6b538d2955ce6b143329db551d8d 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
@@ -90,7 +90,6 @@ struct RTCPSender::RtcpContext { |
buffer_size(buffer_size), |
ntp_sec(0), |
ntp_frac(0), |
- jitter_transmission_offset(0), |
position(0) {} |
uint8_t* AllocateData(uint32_t bytes) { |
@@ -109,7 +108,6 @@ struct RTCPSender::RtcpContext { |
uint32_t buffer_size; |
uint32_t ntp_sec; |
uint32_t ntp_frac; |
- uint32_t jitter_transmission_offset; |
uint32_t position; |
}; |
@@ -146,7 +144,6 @@ RTCPSender::RTCPSender( |
using_nack_(false), |
sending_(false), |
remb_enabled_(false), |
- extended_jitter_report_enabled_(false), |
next_time_to_send_rtcp_(0), |
start_timestamp_(0), |
last_rtp_timestamp_(0), |
@@ -176,8 +173,6 @@ RTCPSender::RTCPSender( |
builders_[kRtcpSr] = &RTCPSender::BuildSR; |
builders_[kRtcpRr] = &RTCPSender::BuildRR; |
builders_[kRtcpSdes] = &RTCPSender::BuildSDES; |
- builders_[kRtcpTransmissionTimeOffset] = |
- &RTCPSender::BuildExtendedJitterReport; |
builders_[kRtcpPli] = &RTCPSender::BuildPLI; |
builders_[kRtcpFir] = &RTCPSender::BuildFIR; |
builders_[kRtcpSli] = &RTCPSender::BuildSLI; |
@@ -280,16 +275,6 @@ void RTCPSender::SetTMMBRStatus(bool enable) { |
} |
} |
-bool RTCPSender::IJ() const { |
- CriticalSectionScoped lock(critical_section_rtcp_sender_.get()); |
- return extended_jitter_report_enabled_; |
-} |
- |
-void RTCPSender::SetIJStatus(bool enable) { |
- CriticalSectionScoped lock(critical_section_rtcp_sender_.get()); |
- extended_jitter_report_enabled_ = enable; |
-} |
- |
void RTCPSender::SetStartTimestamp(uint32_t start_timestamp) { |
CriticalSectionScoped lock(critical_section_rtcp_sender_.get()); |
start_timestamp_ = start_timestamp; |
@@ -563,45 +548,6 @@ RTCPSender::BuildResult RTCPSender::BuildRR(RtcpContext* ctx) { |
return BuildResult::kSuccess; |
} |
-// From RFC 5450: Transmission Time Offsets in RTP Streams. |
-// 0 1 2 3 |
-// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
-// hdr |V=2|P| RC | PT=IJ=195 | length | |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
-// | inter-arrival jitter | |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
-// . . |
-// . . |
-// . . |
-// | inter-arrival jitter | |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
-// |
-// If present, this RTCP packet must be placed after a receiver report |
-// (inside a compound RTCP packet), and MUST have the same value for RC |
-// (reception report count) as the receiver report. |
- |
-RTCPSender::BuildResult RTCPSender::BuildExtendedJitterReport( |
- RtcpContext* ctx) { |
- // sanity |
- if (ctx->position + 8 >= IP_PACKET_SIZE) |
- return BuildResult::kTruncated; |
- |
- // add picture loss indicator |
- uint8_t RC = 1; |
- *ctx->AllocateData(1) = 0x80 + RC; |
- *ctx->AllocateData(1) = 195; |
- |
- // Used fixed length of 2 |
- *ctx->AllocateData(1) = 0; |
- *ctx->AllocateData(1) = 1; |
- |
- // Add inter-arrival jitter |
- ByteWriter<uint32_t>::WriteBigEndian(ctx->AllocateData(4), |
- ctx->jitter_transmission_offset); |
- return BuildResult::kSuccess; |
-} |
- |
RTCPSender::BuildResult RTCPSender::BuildPLI(RtcpContext* ctx) { |
// sanity |
if (ctx->position + 12 >= IP_PACKET_SIZE) |
@@ -1386,8 +1332,6 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state, |
AddReportBlock(report_block); |
} |
} |
- if (extended_jitter_report_enabled_) |
- SetFlag(kRtcpTransmissionTimeOffset, true); |
} |
} |