Revert of Simplify and extend RtpHeaderExtensionMap

webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc

 /*
  *  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.
  */
 
@@ skipping 15 matching lines @@
 //
 // 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               |
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 constexpr RTPExtensionType AbsoluteSendTime::kId;
 constexpr uint8_t AbsoluteSendTime::kValueSizeBytes;
-constexpr const char* AbsoluteSendTime::kUri;
+const char* const AbsoluteSendTime::kUri =
+    "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
 
 bool AbsoluteSendTime::Parse(const uint8_t* data, uint32_t* time_24bits) {
   *time_24bits = ByteReader<uint32_t, 3>::ReadBigEndian(data);
   return true;
 }
 
 bool AbsoluteSendTime::Write(uint8_t* data, int64_t time_ms) {
   ByteWriter<uint32_t, 3>::WriteBigEndian(data, MsTo24Bits(time_ms));
   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     |
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //
 constexpr RTPExtensionType AudioLevel::kId;
 constexpr uint8_t AudioLevel::kValueSizeBytes;
-constexpr const char* AudioLevel::kUri;
+const char* const AudioLevel::kUri =
+    "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
 
 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,
@@ skipping 15 matching lines @@
 //
 // 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              |
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 constexpr RTPExtensionType TransmissionOffset::kId;
 constexpr uint8_t TransmissionOffset::kValueSizeBytes;
-constexpr const char* TransmissionOffset::kUri;
+const char* const TransmissionOffset::kUri =
+    "urn:ietf:params:rtp-hdrext:toffset";
 
 bool TransmissionOffset::Parse(const uint8_t* data, int32_t* rtp_time) {
   *rtp_time = ByteReader<int32_t, 3>::ReadBigEndian(data);
   return true;
 }
 
 bool TransmissionOffset::Write(uint8_t* data, int32_t rtp_time) {
   RTC_DCHECK_LE(rtp_time, 0x00ffffff);
   ByteWriter<int32_t, 3>::WriteBigEndian(data, rtp_time);
   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 |
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 constexpr RTPExtensionType TransportSequenceNumber::kId;
 constexpr uint8_t TransportSequenceNumber::kValueSizeBytes;
-constexpr const char* TransportSequenceNumber::kUri;
+const char* const TransportSequenceNumber::kUri =
+    "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
 
 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|
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 constexpr RTPExtensionType VideoOrientation::kId;
 constexpr uint8_t VideoOrientation::kValueSizeBytes;
-constexpr const char* VideoOrientation::kUri;
+const char* const VideoOrientation::kUri = "urn:3gpp:video-orientation";
 
 bool VideoOrientation::Parse(const uint8_t* data, VideoRotation* rotation) {
   *rotation = ConvertCVOByteToVideoRotation(data[0]);
   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;
 }
 
 //   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;
-constexpr const char* PlayoutDelayLimits::kUri;
+const char* const PlayoutDelayLimits::kUri =
+    "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
 
 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