RtpPacket class introduced.

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
new file mode 100644
index 0000000000000000000000000000000000000000..a551b15617b55677efdf80050426d077b41af311
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+// Absolute send time in RTP streams.
+//
+// The absolute send time is signaled to the receiver in-band using the
+// general mechanism for RTP header extensions [RFC5285]. The payload
+// of this extension (the transmitted value) is a 24-bit unsigned integer
+// containing the sender's current time in seconds as a fixed point number
+// with 18 bits fractional part.
+//
+// The form of the absolute send time extension block:
+//
+//    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 |              absolute send time               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+const char* AbsoluteSendTime::kName =
+    "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
+bool AbsoluteSendTime::IsSupportedFor(MediaType type) {
+  return true;
+}
+
+bool AbsoluteSendTime::Parse(const uint8_t* data, uint32_t* value) {
+  *value = ByteReader<uint32_t, 3>::ReadBigEndian(data);
+  return true;
+}
+
+bool AbsoluteSendTime::Write(uint8_t* data, int64_t time_ms) {
+  const uint32_t kAbsSendTimeFraction = 18;
+  uint32_t time_24_bits =
+      static_cast<uint32_t>(((time_ms << kAbsSendTimeFraction) + 500) / 1000) &
+      0x00FFFFFF;
+
+  ByteWriter<uint32_t, 3>::WriteBigEndian(data, time_24_bits);
+  return true;
+}
+
+// An RTP Header Extension for Client-to-Mixer Audio Level Indication
+//
+// https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
+//
+// The form of the audio level extension block:
+//
+//    0                   1
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=0 |V|   level     |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+const char* AudioLevel::kName = "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
+bool AudioLevel::IsSupportedFor(MediaType type) {
+  switch (type) {
+    case MediaType::ANY:
+    case MediaType::AUDIO:
+      return true;
+    case MediaType::VIDEO:
+    case MediaType::DATA:
+      return false;
+  }
+  RTC_NOTREACHED();
+  return false;
+}
+
+bool AudioLevel::Parse(const uint8_t* data,
+                       bool* voice_activity,
+                       uint8_t* audio_level) {
+  *voice_activity = (data[0] & 0x80) != 0;
+  *audio_level = data[0] & 0x7F;
+  return true;
+}
+
+bool AudioLevel::Write(uint8_t* data,
+                       bool voice_activity,
+                       uint8_t audio_level) {
+  RTC_CHECK_LE(audio_level, 0x7f);
+  data[0] = (voice_activity ? 0x80 : 0x00) | audio_level;
+  return true;
+}
+
+// From RFC 5450: Transmission Time Offsets in RTP Streams.
+//
+// The transmission time is signaled to the receiver in-band using the
+// general mechanism for RTP header extensions [RFC5285]. The payload
+// of this extension (the transmitted value) is a 24-bit signed integer.
+// When added to the RTP timestamp of the packet, it represents the
+// "effective" RTP transmission time of the packet, on the RTP
+// timescale.
+//
+// The form of the transmission offset extension block:
+//
+//    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 |              transmission offset              |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+const char* TransmissionOffset::kName = "urn:ietf:params:rtp-hdrext:toffset";
+bool TransmissionOffset::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 TransmissionOffset::Parse(const uint8_t* data, int32_t* value) {
+  *value = ByteReader<int32_t, 3>::ReadBigEndian(data);
+  return true;
+}
+
+bool TransmissionOffset::Write(uint8_t* data, int64_t value) {
+  RTC_CHECK_LE(value, 0x00ffffff);
+  ByteWriter<int32_t, 3>::WriteBigEndian(data, value);
+  return true;
+}
+
+//   0                   1                   2
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  ID   | L=1   |transport wide sequence number |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+const char* TransportSequenceNumber::kName =
+    "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
+bool TransportSequenceNumber::IsSupportedFor(MediaType type) {
+  return true;
+}
+
+bool TransportSequenceNumber::Parse(const uint8_t* data, uint16_t* value) {
+  *value = ByteReader<uint16_t>::ReadBigEndian(data);
+  return true;
+}
+
+bool TransportSequenceNumber::Write(uint8_t* data, uint16_t value) {
+  ByteWriter<uint16_t>::WriteBigEndian(data, value);
+  return true;
+}
+
+// Coordination of Video Orientation in RTP streams.
+//
+// Coordination of Video Orientation consists in signaling of the current
+// orientation of the image captured on the sender side to the receiver for
+// appropriate rendering and displaying.
+//
+//    0                   1
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=0 |0 0 0 0 C F R R|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+const char* VideoOrientation::kName = "urn:3gpp:video-orientation";
+bool VideoOrientation::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 VideoOrientation::Parse(const uint8_t* data, VideoRotation* rotation) {
+  *rotation = ConvertCVOByteToVideoRotation(data[0] & 0x03);
+  return true;
+}
+
+bool VideoOrientation::Write(uint8_t* data, VideoRotation rotation) {
+  data[0] = ConvertVideoRotationToCVOByte(rotation);
+  return true;
+}
+
+bool VideoOrientation::Parse(const uint8_t* data, uint8_t* value) {
+  *value = data[0];
+  return true;
+}
+
+bool VideoOrientation::Write(uint8_t* data, uint8_t value) {
+  data[0] = value;
+  return true;
+}
+}  // namespace webrtc