/* * 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::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(((time_ms << kAbsSendTimeFraction) + 500) / 1000) & 0x00FFFFFF; ByteWriter::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::ReadBigEndian(data); return true; } bool TransmissionOffset::Write(uint8_t* data, int64_t value) { RTC_CHECK_LE(value, 0x00ffffff); ByteWriter::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::ReadBigEndian(data); return true; } bool TransportSequenceNumber::Write(uint8_t* data, uint16_t value) { ByteWriter::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