Use RtpPacketToSend in RtpSenderVideo

webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h

Index: webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
index 4800a60c39d40c4ecc148d10cf27cde623aed7c0..0703c599c59d9d1edb08c4043fb3c13447bd98c1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -23,12 +23,8 @@ class AbsoluteSendTime {
   static constexpr uint8_t kValueSizeBytes = 3;
   static const char* kName;
   static bool IsSupportedFor(MediaType type);
-  static bool Parse(const uint8_t* data, uint32_t* time_24bits);
+  static bool Parse(const uint8_t* data, uint32_t* time_ms);
   static bool Write(uint8_t* data, int64_t time_ms);
-
-  static constexpr uint32_t MsTo24Bits(int64_t time_ms) {
-    return static_cast<uint32_t>(((time_ms << 18) + 500) / 1000) & 0x00FFFFFF;
-  }
 };
 
 class AudioLevel {
@@ -75,5 +71,22 @@ class VideoOrientation {
   static bool Write(uint8_t* data, uint8_t value);
 };
 
+class PlayoutDelayLimits {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionPlayoutDelay;
+  static constexpr uint8_t kValueSizeBytes = 3;
+  static const char* kName;
+  static bool IsSupportedFor(MediaType type);
+  // Playout delay in milliseconds. A playout delay limit (min or max)
+  // has 12 bits allocated. This allows a range of 0-4095 values which
+  // translates to a range of 0-40950 in milliseconds.
+  static constexpr int kGranularityMs = 10;
+  // Maximum playout delay value in milliseconds.
+  static constexpr int kMaxMs = 0xfff * kGranularityMs;  // 40950.
+
+  static bool Parse(const uint8_t* data, PlayoutDelay* playout_delay);
+  static bool Write(uint8_t* data, const PlayoutDelay& playout_delay);
+};
+
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_