c7c26a0e64
Reason for revert: Doing a reland where systeminfo.cc includes basictypes.h so that CPU_X86 etc. are defined when they are checked/used. Original issue's description: > Revert of Replace basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2604043002/ ) > > Reason for revert: > Very likely cause of Chromium import bot breakage (unused function '__cpuid'), TBD why. > > Original issue's description: > > Replace basictypes.h with stdint.h for int_t types. > > > > Removes basictypes.h for types that only makes use of it for fixed-size-int > > typedefs and replaces it with stdint.h. > > > > BUG=webrtc:6853 > > R=tommi@webrtc.org > > > > Review-Url: https://codereview.webrtc.org/2604043002 > > Cr-Commit-Position: refs/heads/master@{#15867} > > Committed:7fd1a75300> > TBR=tommi@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6853 > > Review-Url: https://codereview.webrtc.org/2603203003 > Cr-Commit-Position: refs/heads/master@{#15869} > Committed:7eb0e23bcfBUG=webrtc:6853 TBR=tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2609783002 Cr-Commit-Position: refs/heads/master@{#15873}
101 lines
3.7 KiB
C++
101 lines
3.7 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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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
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#include <stdint.h>
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#include "webrtc/common_video/rotation.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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class AbsoluteSendTime {
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public:
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static constexpr RTPExtensionType kId = kRtpExtensionAbsoluteSendTime;
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static constexpr uint8_t kValueSizeBytes = 3;
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static constexpr const char* kUri =
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"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
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static bool Parse(const uint8_t* data, uint32_t* time_24bits);
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static bool Write(uint8_t* data, int64_t time_ms);
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static constexpr uint32_t MsTo24Bits(int64_t time_ms) {
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return static_cast<uint32_t>(((time_ms << 18) + 500) / 1000) & 0x00FFFFFF;
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}
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};
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class AudioLevel {
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public:
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static constexpr RTPExtensionType kId = kRtpExtensionAudioLevel;
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static constexpr uint8_t kValueSizeBytes = 1;
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static constexpr const char* kUri =
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"urn:ietf:params:rtp-hdrext:ssrc-audio-level";
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static bool Parse(const uint8_t* data,
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bool* voice_activity,
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uint8_t* audio_level);
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static bool Write(uint8_t* data, bool voice_activity, uint8_t audio_level);
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};
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class TransmissionOffset {
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public:
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static constexpr RTPExtensionType kId = kRtpExtensionTransmissionTimeOffset;
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static constexpr uint8_t kValueSizeBytes = 3;
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static constexpr const char* kUri = "urn:ietf:params:rtp-hdrext:toffset";
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static bool Parse(const uint8_t* data, int32_t* rtp_time);
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static bool Write(uint8_t* data, int32_t rtp_time);
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};
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class TransportSequenceNumber {
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public:
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static constexpr RTPExtensionType kId = kRtpExtensionTransportSequenceNumber;
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static constexpr uint8_t kValueSizeBytes = 2;
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static constexpr const char* kUri =
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"http://www.ietf.org/id/"
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"draft-holmer-rmcat-transport-wide-cc-extensions-01";
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static bool Parse(const uint8_t* data, uint16_t* value);
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static bool Write(uint8_t* data, uint16_t value);
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};
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class VideoOrientation {
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public:
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static constexpr RTPExtensionType kId = kRtpExtensionVideoRotation;
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static constexpr uint8_t kValueSizeBytes = 1;
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static constexpr const char* kUri = "urn:3gpp:video-orientation";
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static bool Parse(const uint8_t* data, VideoRotation* value);
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static bool Write(uint8_t* data, VideoRotation value);
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static bool Parse(const uint8_t* data, uint8_t* value);
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static bool Write(uint8_t* data, uint8_t value);
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};
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class PlayoutDelayLimits {
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public:
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static constexpr RTPExtensionType kId = kRtpExtensionPlayoutDelay;
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static constexpr uint8_t kValueSizeBytes = 3;
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static constexpr const char* kUri =
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"http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
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// Playout delay in milliseconds. A playout delay limit (min or max)
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// has 12 bits allocated. This allows a range of 0-4095 values which
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// translates to a range of 0-40950 in milliseconds.
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static constexpr int kGranularityMs = 10;
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// Maximum playout delay value in milliseconds.
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static constexpr int kMaxMs = 0xfff * kGranularityMs; // 40950.
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static bool Parse(const uint8_t* data, PlayoutDelay* playout_delay);
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static bool Write(uint8_t* data, const PlayoutDelay& playout_delay);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
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