Implement PlayoutDelay extension as a trait

webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
index 441c548906b07cc646faa08844dfc0d03f18832b..a3153d2190cacac464c9dbc5517254e9c73fb4c4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -200,4 +200,52 @@ bool VideoOrientation::Write(uint8_t* data, uint8_t value) {
   data[0] = value;
   return true;
 }
+
+//   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
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  ID   | len=2 |   MIN delay           |   MAX delay           |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr RTPExtensionType PlayoutDelayLimits::kId;
+constexpr uint8_t PlayoutDelayLimits::kValueSizeBytes;
+const char* PlayoutDelayLimits::kName =
+    "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
+bool PlayoutDelayLimits::IsSupportedFor(MediaType type) {
+  switch (type) {
+    case MediaType::ANY:
+    case MediaType::VIDEO:
+      return true;
+    case MediaType::AUDIO:
+    case MediaType::DATA:
+      return false;
+  }
+  RTC_NOTREACHED();
+  return false;
+}
+
+bool PlayoutDelayLimits::Parse(const uint8_t* data,
+                               PlayoutDelay* playout_delay) {
+  RTC_DCHECK(playout_delay);
+  uint32_t raw = ByteReader<uint32_t, 3>::ReadBigEndian(data);
+  uint16_t min_raw = (raw >> 12);
+  uint16_t max_raw = (raw & 0xfff);
+  if (min_raw > max_raw)
+    return false;
+  playout_delay->min_ms = min_raw * kGranularityMs;
+  playout_delay->max_ms = max_raw * kGranularityMs;
+  return true;
+}
+
+bool PlayoutDelayLimits::Write(uint8_t* data,
+                               const PlayoutDelay& playout_delay) {
+  RTC_DCHECK_LE(0, playout_delay.min_ms);
+  RTC_DCHECK_LE(playout_delay.min_ms, playout_delay.max_ms);
+  RTC_DCHECK_LE(playout_delay.max_ms, kMaxMs);
+  // Convert MS to value to be sent on extension header.
+  uint32_t min_delay = playout_delay.min_ms / kGranularityMs;
+  uint32_t max_delay = playout_delay.max_ms / kGranularityMs;
+  ByteWriter<uint32_t, 3>::WriteBigEndian(data, (min_delay << 12) | max_delay);
+  return true;
+}
+
 }  // namespace webrtc