blob: 95473f4e390fe053dfbc691afb1654d51b042faf [file] [log] [blame]
/*
* 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 "webrtc/base/basictypes.h"
#include "webrtc/call.h"
#include "webrtc/common_video/rotation.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 const char* kName;
static bool IsSupportedFor(MediaType type);
static bool Parse(const uint8_t* data, uint32_t* time_24bits);
static bool Write(uint8_t* data, int64_t time_ms);
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 const char* kName;
static bool IsSupportedFor(MediaType type);
static bool Parse(const uint8_t* data,
bool* voice_activity,
uint8_t* audio_level);
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 const char* kName;
static bool IsSupportedFor(MediaType type);
static bool Parse(const uint8_t* data, int32_t* rtp_time);
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 const char* kName;
static bool IsSupportedFor(MediaType type);
static bool Parse(const uint8_t* data, uint16_t* value);
static bool Write(uint8_t* data, uint16_t value);
};
class VideoOrientation {
public:
static constexpr RTPExtensionType kId = kRtpExtensionVideoRotation;
static constexpr uint8_t kValueSizeBytes = 1;
static const char* kName;
static bool IsSupportedFor(MediaType type);
static bool Parse(const uint8_t* data, VideoRotation* value);
static bool Write(uint8_t* data, VideoRotation value);
static bool Parse(const uint8_t* data, uint8_t* value);
static bool Write(uint8_t* data, uint8_t value);
};
class PlayoutDelayLimits {
public:
static constexpr RTPExtensionType kId = kRtpExtensionPlayoutDelay;
static constexpr uint8_t kValueSizeBytes = 3;
static const char* kName;
static bool IsSupportedFor(MediaType type);
// 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(const uint8_t* data, PlayoutDelay* playout_delay);
static bool Write(uint8_t* data, const PlayoutDelay& playout_delay);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_