e2a0177255
renamed kName to kUri and make it more const. remove IsSupportedBy to reduce header dependency. BUG=webrtc:1994 Review-Url: https://codereview.webrtc.org/2457783005 Cr-Commit-Position: refs/heads/master@{#14825}
205 lines
7.7 KiB
C++
205 lines
7.7 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
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#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
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namespace webrtc {
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// Absolute send time in RTP streams.
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//
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// The absolute send time is signaled to the receiver in-band using the
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// general mechanism for RTP header extensions [RFC5285]. The payload
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// of this extension (the transmitted value) is a 24-bit unsigned integer
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// containing the sender's current time in seconds as a fixed point number
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// with 18 bits fractional part.
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//
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// The form of the absolute send time extension block:
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//
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// 0 1 2 3
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// 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
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | len=2 | absolute send time |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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constexpr RTPExtensionType AbsoluteSendTime::kId;
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constexpr uint8_t AbsoluteSendTime::kValueSizeBytes;
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const char* const AbsoluteSendTime::kUri =
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"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
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bool AbsoluteSendTime::Parse(const uint8_t* data, uint32_t* time_24bits) {
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*time_24bits = ByteReader<uint32_t, 3>::ReadBigEndian(data);
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return true;
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}
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bool AbsoluteSendTime::Write(uint8_t* data, int64_t time_ms) {
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ByteWriter<uint32_t, 3>::WriteBigEndian(data, MsTo24Bits(time_ms));
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return true;
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}
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// An RTP Header Extension for Client-to-Mixer Audio Level Indication
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//
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// https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
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//
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// The form of the audio level extension block:
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//
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// 0 1
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | len=0 |V| level |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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//
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constexpr RTPExtensionType AudioLevel::kId;
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constexpr uint8_t AudioLevel::kValueSizeBytes;
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const char* const AudioLevel::kUri =
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"urn:ietf:params:rtp-hdrext:ssrc-audio-level";
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bool AudioLevel::Parse(const uint8_t* data,
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bool* voice_activity,
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uint8_t* audio_level) {
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*voice_activity = (data[0] & 0x80) != 0;
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*audio_level = data[0] & 0x7F;
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return true;
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}
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bool AudioLevel::Write(uint8_t* data,
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bool voice_activity,
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uint8_t audio_level) {
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RTC_CHECK_LE(audio_level, 0x7f);
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data[0] = (voice_activity ? 0x80 : 0x00) | audio_level;
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return true;
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}
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// From RFC 5450: Transmission Time Offsets in RTP Streams.
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//
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// The transmission time is signaled to the receiver in-band using the
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// general mechanism for RTP header extensions [RFC5285]. The payload
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// of this extension (the transmitted value) is a 24-bit signed integer.
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// When added to the RTP timestamp of the packet, it represents the
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// "effective" RTP transmission time of the packet, on the RTP
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// timescale.
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//
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// The form of the transmission offset extension block:
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//
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// 0 1 2 3
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// 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
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | len=2 | transmission offset |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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constexpr RTPExtensionType TransmissionOffset::kId;
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constexpr uint8_t TransmissionOffset::kValueSizeBytes;
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const char* const TransmissionOffset::kUri =
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"urn:ietf:params:rtp-hdrext:toffset";
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bool TransmissionOffset::Parse(const uint8_t* data, int32_t* rtp_time) {
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*rtp_time = ByteReader<int32_t, 3>::ReadBigEndian(data);
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return true;
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}
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bool TransmissionOffset::Write(uint8_t* data, int32_t rtp_time) {
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RTC_DCHECK_LE(rtp_time, 0x00ffffff);
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ByteWriter<int32_t, 3>::WriteBigEndian(data, rtp_time);
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return true;
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}
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// 0 1 2
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | L=1 |transport wide sequence number |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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constexpr RTPExtensionType TransportSequenceNumber::kId;
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constexpr uint8_t TransportSequenceNumber::kValueSizeBytes;
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const char* const TransportSequenceNumber::kUri =
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"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
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bool TransportSequenceNumber::Parse(const uint8_t* data, uint16_t* value) {
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*value = ByteReader<uint16_t>::ReadBigEndian(data);
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return true;
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}
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bool TransportSequenceNumber::Write(uint8_t* data, uint16_t value) {
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ByteWriter<uint16_t>::WriteBigEndian(data, value);
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return true;
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}
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// Coordination of Video Orientation in RTP streams.
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//
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// Coordination of Video Orientation consists in signaling of the current
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// orientation of the image captured on the sender side to the receiver for
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// appropriate rendering and displaying.
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//
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// 0 1
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | len=0 |0 0 0 0 C F R R|
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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constexpr RTPExtensionType VideoOrientation::kId;
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constexpr uint8_t VideoOrientation::kValueSizeBytes;
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const char* const VideoOrientation::kUri = "urn:3gpp:video-orientation";
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bool VideoOrientation::Parse(const uint8_t* data, VideoRotation* rotation) {
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*rotation = ConvertCVOByteToVideoRotation(data[0]);
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return true;
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}
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bool VideoOrientation::Write(uint8_t* data, VideoRotation rotation) {
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data[0] = ConvertVideoRotationToCVOByte(rotation);
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return true;
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}
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bool VideoOrientation::Parse(const uint8_t* data, uint8_t* value) {
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*value = data[0];
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return true;
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}
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bool VideoOrientation::Write(uint8_t* data, uint8_t value) {
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data[0] = value;
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return true;
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}
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// 0 1 2 3
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// 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
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | len=2 | MIN delay | MAX delay |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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constexpr RTPExtensionType PlayoutDelayLimits::kId;
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constexpr uint8_t PlayoutDelayLimits::kValueSizeBytes;
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const char* const PlayoutDelayLimits::kUri =
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"http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
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bool PlayoutDelayLimits::Parse(const uint8_t* data,
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PlayoutDelay* playout_delay) {
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RTC_DCHECK(playout_delay);
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uint32_t raw = ByteReader<uint32_t, 3>::ReadBigEndian(data);
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uint16_t min_raw = (raw >> 12);
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uint16_t max_raw = (raw & 0xfff);
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if (min_raw > max_raw)
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return false;
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playout_delay->min_ms = min_raw * kGranularityMs;
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playout_delay->max_ms = max_raw * kGranularityMs;
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return true;
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}
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bool PlayoutDelayLimits::Write(uint8_t* data,
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const PlayoutDelay& playout_delay) {
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RTC_DCHECK_LE(0, playout_delay.min_ms);
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RTC_DCHECK_LE(playout_delay.min_ms, playout_delay.max_ms);
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RTC_DCHECK_LE(playout_delay.max_ms, kMaxMs);
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// Convert MS to value to be sent on extension header.
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uint32_t min_delay = playout_delay.min_ms / kGranularityMs;
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uint32_t max_delay = playout_delay.max_ms / kGranularityMs;
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ByteWriter<uint32_t, 3>::WriteBigEndian(data, (min_delay << 12) | max_delay);
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return true;
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}
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} // namespace webrtc
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