Index: webrtc/modules/audio_coding/neteq/rtcp.cc |
diff --git a/webrtc/modules/audio_coding/neteq/rtcp.cc b/webrtc/modules/audio_coding/neteq/rtcp.cc |
index 0263e763efb56dc377d4ef8308a0cb1185409297..3f8ef0e4e9e9d11f76932327f188f0af45818e85 100644 |
--- a/webrtc/modules/audio_coding/neteq/rtcp.cc |
+++ b/webrtc/modules/audio_coding/neteq/rtcp.cc |
@@ -56,24 +56,24 @@ void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) { |
void Rtcp::GetStatistics(bool no_reset, RtcpStatistics* stats) { |
// Extended highest sequence number received. |
- stats->extended_max_sequence_number = |
+ stats->extended_highest_sequence_number = |
(static_cast<int>(cycles_) << 16) + max_seq_no_; |
// Calculate expected number of packets and compare it with the number of |
// packets that were actually received. The cumulative number of lost packets |
// can be extracted. |
uint32_t expected_packets = |
- stats->extended_max_sequence_number - base_seq_no_ + 1; |
+ stats->extended_highest_sequence_number - base_seq_no_ + 1; |
if (received_packets_ == 0) { |
// No packets received, assume none lost. |
- stats->cumulative_lost = 0; |
+ stats->packets_lost = 0; |
} else if (expected_packets > received_packets_) { |
- stats->cumulative_lost = expected_packets - received_packets_; |
- if (stats->cumulative_lost > 0xFFFFFF) { |
- stats->cumulative_lost = 0xFFFFFF; |
+ stats->packets_lost = expected_packets - received_packets_; |
+ if (stats->packets_lost > 0xFFFFFF) { |
+ stats->packets_lost = 0xFFFFFF; |
} |
} else { |
- stats->cumulative_lost = 0; |
+ stats->packets_lost = 0; |
} |
// Fraction lost since last report. |