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/*
* Copyright (c) 2011 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.
*/
Move talk/media to webrtc/media I removed the 'libjingle' target in talk/libjingle.gyp and replaced all users of it with base/base.gyp:rtc_base. It seems the jsoncpp and expat dependencies were not used by it's previous references. The files in talk/media/testdata were uploaded to Google Storage and added .sha1 files in resources/media instead of simply moving them. The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL in order to not break Git history. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc. * Unused GYP reference to libjingle_tests_additional_deps was removed. * Removed duplicated GYP entries of webrtc/base/testutils.cc webrtc/base/testutils.h The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media, so they were moved to the media.gyp. I also checked that none of EXPAT_RELATIVE_PATH, FEATURE_ENABLE_VOICEMAIL, GTEST_RELATIVE_PATH, JSONCPP_RELATIVE_PATH, LOGGING=1, SRTP_RELATIVE_PATH, FEATURE_ENABLE_SSL, FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN, HAVE_SCTP, HAVE_SRTP, are used by the talk/media code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/ BUG=webrtc:5420 NOPRESUBMIT=True TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1587193006 Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
#ifndef WEBRTC_MEDIA_BASE_RTPUTILS_H_
#define WEBRTC_MEDIA_BASE_RTPUTILS_H_
#include "webrtc/rtc_base/byteorder.h"
namespace rtc {
struct PacketTimeUpdateParams;
} // namespace rtc
namespace cricket {
const size_t kMinRtpPacketLen = 12;
const size_t kMaxRtpPacketLen = 2048;
const size_t kMinRtcpPacketLen = 4;
struct RtpHeader {
int payload_type;
int seq_num;
uint32_t timestamp;
uint32_t ssrc;
};
enum RtcpTypes {
kRtcpTypeSR = 200, // Sender report payload type.
kRtcpTypeRR = 201, // Receiver report payload type.
kRtcpTypeSDES = 202, // SDES payload type.
kRtcpTypeBye = 203, // BYE payload type.
kRtcpTypeApp = 204, // APP payload type.
kRtcpTypeRTPFB = 205, // Transport layer Feedback message payload type.
kRtcpTypePSFB = 206, // Payload-specific Feedback message payload type.
};
bool GetRtpPayloadType(const void* data, size_t len, int* value);
bool GetRtpSeqNum(const void* data, size_t len, int* value);
bool GetRtpTimestamp(const void* data, size_t len, uint32_t* value);
bool GetRtpSsrc(const void* data, size_t len, uint32_t* value);
bool GetRtpHeaderLen(const void* data, size_t len, size_t* value);
bool GetRtcpType(const void* data, size_t len, int* value);
bool GetRtcpSsrc(const void* data, size_t len, uint32_t* value);
bool GetRtpHeader(const void* data, size_t len, RtpHeader* header);
bool SetRtpSsrc(void* data, size_t len, uint32_t value);
// Assumes version 2, no padding, no extensions, no csrcs.
bool SetRtpHeader(void* data, size_t len, const RtpHeader& header);
bool IsRtpPacket(const void* data, size_t len);
// True if |payload type| is 0-127.
bool IsValidRtpPayloadType(int payload_type);
// True if |size| is appropriate for the indicated packet type.
bool IsValidRtpRtcpPacketSize(bool rtcp, size_t size);
// TODO(zstein): Consider using an enum instead of a bool to differentiate
// between RTP and RTCP.
// Returns "RTCP" or "RTP" according to |rtcp|.
const char* RtpRtcpStringLiteral(bool rtcp);
// Verifies that a packet has a valid RTP header.
bool ValidateRtpHeader(const uint8_t* rtp,
size_t length,
size_t* header_length);
// Helper method which updates the absolute send time extension if present.
bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
size_t length,
int extension_id,
uint64_t time_us);
// Applies specified |options| to the packet. It updates the absolute send time
// extension header if it is present present then updates HMAC.
bool ApplyPacketOptions(uint8_t* data,
size_t length,
const rtc::PacketTimeUpdateParams& packet_time_params,
uint64_t time_us);
} // namespace cricket
Move talk/media to webrtc/media I removed the 'libjingle' target in talk/libjingle.gyp and replaced all users of it with base/base.gyp:rtc_base. It seems the jsoncpp and expat dependencies were not used by it's previous references. The files in talk/media/testdata were uploaded to Google Storage and added .sha1 files in resources/media instead of simply moving them. The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL in order to not break Git history. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc. * Unused GYP reference to libjingle_tests_additional_deps was removed. * Removed duplicated GYP entries of webrtc/base/testutils.cc webrtc/base/testutils.h The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media, so they were moved to the media.gyp. I also checked that none of EXPAT_RELATIVE_PATH, FEATURE_ENABLE_VOICEMAIL, GTEST_RELATIVE_PATH, JSONCPP_RELATIVE_PATH, LOGGING=1, SRTP_RELATIVE_PATH, FEATURE_ENABLE_SSL, FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN, HAVE_SCTP, HAVE_SRTP, are used by the talk/media code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/ BUG=webrtc:5420 NOPRESUBMIT=True TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1587193006 Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
#endif // WEBRTC_MEDIA_BASE_RTPUTILS_H_