| /* |
| * 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* 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 | |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| // |
| 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* 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 | |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 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]); |
| 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; |
| 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 |