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/*
* 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
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