Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
index 648e25dbfec8acb53576298a1307b0df1627168a..203d45df1c106d95f02dca21e1b5cab86ab63f87 100644 |
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
@@ -22,7 +22,6 @@ |
#define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination |
#define IP_PACKET_SIZE 1500 // we assume ethernet |
#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10 |
-#define TIMEOUT_SEI_MESSAGES_MS 30000 // in milliseconds |
namespace webrtc { |
namespace rtcp { |
@@ -30,6 +29,10 @@ class TransportFeedback; |
} |
const int kVideoPayloadTypeFrequency = 90000; |
+// TODO(solenberg): RTP time stamp rate for RTCP is fixed at 8k, this is legacy |
+// and should be fixed. |
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6458 |
+const int kBogusRtpRateForAudioRtcp = 8000; |
// Minimum RTP header size in bytes. |
const uint8_t kRtpHeaderSize = 12; |