Implement PlayoutDelay extension as a trait
to be used with rtp::Packet class
BUG=webrtc:1994
R=isheriff@chromium.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2224063004 .
Cr-Commit-Position: refs/heads/master@{#14105}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
index 2d5dbf9..a851bc3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -195,4 +195,52 @@
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
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
index 4800a60..95473f4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -75,5 +75,22 @@
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_
diff --git a/webrtc/test/fuzzers/rtp_packet_fuzzer.cc b/webrtc/test/fuzzers/rtp_packet_fuzzer.cc
index 2a67a8b..613f125 100644
--- a/webrtc/test/fuzzers/rtp_packet_fuzzer.cc
+++ b/webrtc/test/fuzzers/rtp_packet_fuzzer.cc
@@ -82,7 +82,8 @@
packet.GetExtension<TransportSequenceNumber>(&seqnum);
break;
case kRtpExtensionPlayoutDelay:
- // TODO(katrielc) Add this once it's written.
+ PlayoutDelay playout;
+ packet.GetExtension<PlayoutDelayLimits>(&playout);
break;
}
}