webrtc_m130/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
ilnik 04f4d126f8 Implement timing frames.
Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}
2017-06-19 14:18:55 +00:00

180 lines
6.8 KiB
C++

/*
* 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.
*/
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
#include <stdint.h>
#include <string>
#include "webrtc/api/video/video_content_type.h"
#include "webrtc/api/video/video_rotation.h"
#include "webrtc/api/video/video_timing.h"
#include "webrtc/base/array_view.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
class AbsoluteSendTime {
public:
static constexpr RTPExtensionType kId = kRtpExtensionAbsoluteSendTime;
static constexpr uint8_t kValueSizeBytes = 3;
static constexpr const char* kUri =
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
static bool Parse(rtc::ArrayView<const uint8_t> data, uint32_t* time_24bits);
static size_t ValueSize(uint32_t time_24bits) { return kValueSizeBytes; }
static bool Write(uint8_t* data, uint32_t time_24bits);
static constexpr uint32_t MsTo24Bits(int64_t time_ms) {
return static_cast<uint32_t>(((time_ms << 18) + 500) / 1000) & 0x00FFFFFF;
}
};
class AudioLevel {
public:
static constexpr RTPExtensionType kId = kRtpExtensionAudioLevel;
static constexpr uint8_t kValueSizeBytes = 1;
static constexpr const char* kUri =
"urn:ietf:params:rtp-hdrext:ssrc-audio-level";
static bool Parse(rtc::ArrayView<const uint8_t> data,
bool* voice_activity,
uint8_t* audio_level);
static size_t ValueSize(bool voice_activity, uint8_t audio_level) {
return kValueSizeBytes;
}
static bool Write(uint8_t* data, bool voice_activity, uint8_t audio_level);
};
class TransmissionOffset {
public:
static constexpr RTPExtensionType kId = kRtpExtensionTransmissionTimeOffset;
static constexpr uint8_t kValueSizeBytes = 3;
static constexpr const char* kUri = "urn:ietf:params:rtp-hdrext:toffset";
static bool Parse(rtc::ArrayView<const uint8_t> data, int32_t* rtp_time);
static size_t ValueSize(int32_t rtp_time) { return kValueSizeBytes; }
static bool Write(uint8_t* data, int32_t rtp_time);
};
class TransportSequenceNumber {
public:
static constexpr RTPExtensionType kId = kRtpExtensionTransportSequenceNumber;
static constexpr uint8_t kValueSizeBytes = 2;
static constexpr const char* kUri =
"http://www.ietf.org/id/"
"draft-holmer-rmcat-transport-wide-cc-extensions-01";
static bool Parse(rtc::ArrayView<const uint8_t> data, uint16_t* value);
static size_t ValueSize(uint16_t value) { return kValueSizeBytes; }
static bool Write(uint8_t* data, uint16_t value);
};
class VideoOrientation {
public:
static constexpr RTPExtensionType kId = kRtpExtensionVideoRotation;
static constexpr uint8_t kValueSizeBytes = 1;
static constexpr const char* kUri = "urn:3gpp:video-orientation";
static bool Parse(rtc::ArrayView<const uint8_t> data, VideoRotation* value);
static size_t ValueSize(VideoRotation) { return kValueSizeBytes; }
static bool Write(uint8_t* data, VideoRotation value);
static bool Parse(rtc::ArrayView<const uint8_t> data, uint8_t* value);
static size_t ValueSize(uint8_t value) { return kValueSizeBytes; }
static bool Write(uint8_t* data, uint8_t value);
};
class PlayoutDelayLimits {
public:
static constexpr RTPExtensionType kId = kRtpExtensionPlayoutDelay;
static constexpr uint8_t kValueSizeBytes = 3;
static constexpr const char* kUri =
"http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
// Playout delay in milliseconds. A playout delay limit (min or max)
// has 12 bits allocated. This allows a range of 0-4095 values which
// translates to a range of 0-40950 in milliseconds.
static constexpr int kGranularityMs = 10;
// Maximum playout delay value in milliseconds.
static constexpr int kMaxMs = 0xfff * kGranularityMs; // 40950.
static bool Parse(rtc::ArrayView<const uint8_t> data,
PlayoutDelay* playout_delay);
static size_t ValueSize(const PlayoutDelay&) {
return kValueSizeBytes;
}
static bool Write(uint8_t* data, const PlayoutDelay& playout_delay);
};
class VideoContentTypeExtension {
public:
static constexpr RTPExtensionType kId = kRtpExtensionVideoContentType;
static constexpr uint8_t kValueSizeBytes = 1;
static constexpr const char* kUri =
"http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
static bool Parse(rtc::ArrayView<const uint8_t> data,
VideoContentType* content_type);
static size_t ValueSize(VideoContentType) {
return kValueSizeBytes;
}
static bool Write(uint8_t* data, VideoContentType content_type);
};
class VideoTimingExtension {
public:
static constexpr RTPExtensionType kId = kRtpExtensionVideoTiming;
static constexpr uint8_t kValueSizeBytes = 12;
static constexpr const char* kUri =
"http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
static bool Parse(rtc::ArrayView<const uint8_t> data, VideoTiming* timing);
static size_t ValueSize(const VideoTiming&) { return kValueSizeBytes; }
static bool Write(uint8_t* data, const VideoTiming& timing);
static size_t ValueSize(uint16_t time_delta_ms, uint8_t idx) {
return kValueSizeBytes;
}
// Writes only single time delta to position idx.
static bool Write(uint8_t* data, uint16_t time_delta_ms, uint8_t idx);
};
class RtpStreamId {
public:
static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
static constexpr const char* kUri =
"urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
static bool Parse(rtc::ArrayView<const uint8_t> data, StreamId* rsid);
static size_t ValueSize(const StreamId& rsid) { return rsid.size(); }
static bool Write(uint8_t* data, const StreamId& rsid);
static bool Parse(rtc::ArrayView<const uint8_t> data, std::string* rsid);
static size_t ValueSize(const std::string& rsid) { return rsid.size(); }
static bool Write(uint8_t* data, const std::string& rsid);
};
class RepairedRtpStreamId {
public:
static constexpr RTPExtensionType kId = kRtpExtensionRepairedRtpStreamId;
static constexpr const char* kUri =
"urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id";
static bool Parse(rtc::ArrayView<const uint8_t> data, StreamId* rsid);
static size_t ValueSize(const StreamId& rsid);
static bool Write(uint8_t* data, const StreamId& rsid);
static bool Parse(rtc::ArrayView<const uint8_t> data, std::string* rsid);
static size_t ValueSize(const std::string& rsid);
static bool Write(uint8_t* data, const std::string& rsid);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_