ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
118 lines
3.4 KiB
C++
118 lines
3.4 KiB
C++
/*
|
|
* Copyright (c) 2012 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_RTP_HEADER_EXTENSION_H_
|
|
#define WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
|
|
|
|
#include <map>
|
|
|
|
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
|
|
|
|
const size_t kRtpOneByteHeaderLength = 4;
|
|
const size_t kTransmissionTimeOffsetLength = 4;
|
|
const size_t kAudioLevelLength = 2;
|
|
const size_t kAbsoluteSendTimeLength = 4;
|
|
const size_t kVideoRotationLength = 2;
|
|
const size_t kTransportSequenceNumberLength = 3;
|
|
|
|
struct HeaderExtension {
|
|
HeaderExtension(RTPExtensionType extension_type)
|
|
: type(extension_type), length(0), active(true) {
|
|
Init();
|
|
}
|
|
|
|
HeaderExtension(RTPExtensionType extension_type, bool active)
|
|
: type(extension_type), length(0), active(active) {
|
|
Init();
|
|
}
|
|
|
|
void Init() {
|
|
// TODO(solenberg): Create handler classes for header extensions so we can
|
|
// get rid of switches like these as well as handling code spread out all
|
|
// over.
|
|
switch (type) {
|
|
case kRtpExtensionTransmissionTimeOffset:
|
|
length = kTransmissionTimeOffsetLength;
|
|
break;
|
|
case kRtpExtensionAudioLevel:
|
|
length = kAudioLevelLength;
|
|
break;
|
|
case kRtpExtensionAbsoluteSendTime:
|
|
length = kAbsoluteSendTimeLength;
|
|
break;
|
|
case kRtpExtensionVideoRotation:
|
|
length = kVideoRotationLength;
|
|
break;
|
|
case kRtpExtensionTransportSequenceNumber:
|
|
length = kTransportSequenceNumberLength;
|
|
break;
|
|
default:
|
|
assert(false);
|
|
}
|
|
}
|
|
|
|
const RTPExtensionType type;
|
|
uint8_t length;
|
|
bool active;
|
|
};
|
|
|
|
class RtpHeaderExtensionMap {
|
|
public:
|
|
RtpHeaderExtensionMap();
|
|
~RtpHeaderExtensionMap();
|
|
|
|
void Erase();
|
|
|
|
int32_t Register(const RTPExtensionType type, const uint8_t id);
|
|
|
|
// Active is a concept for a registered rtp header extension which doesn't
|
|
// take effect yet until being activated. Inactive RTP header extensions do
|
|
// not take effect and should not be included in size calculations until they
|
|
// are activated.
|
|
int32_t RegisterInactive(const RTPExtensionType type, const uint8_t id);
|
|
bool SetActive(const RTPExtensionType type, bool active);
|
|
|
|
int32_t Deregister(const RTPExtensionType type);
|
|
|
|
bool IsRegistered(RTPExtensionType type) const;
|
|
|
|
int32_t GetType(const uint8_t id, RTPExtensionType* type) const;
|
|
|
|
int32_t GetId(const RTPExtensionType type, uint8_t* id) const;
|
|
|
|
//
|
|
// Methods below ignore any inactive rtp header extensions.
|
|
//
|
|
|
|
size_t GetTotalLengthInBytes() const;
|
|
|
|
int32_t GetLengthUntilBlockStartInBytes(const RTPExtensionType type) const;
|
|
|
|
void GetCopy(RtpHeaderExtensionMap* map) const;
|
|
|
|
int32_t Size() const;
|
|
|
|
RTPExtensionType First() const;
|
|
|
|
RTPExtensionType Next(RTPExtensionType type) const;
|
|
|
|
private:
|
|
int32_t Register(const RTPExtensionType type, const uint8_t id, bool active);
|
|
std::map<uint8_t, HeaderExtension*> extensionMap_;
|
|
};
|
|
}
|
|
|
|
#endif // WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
|