webrtc_m130/webrtc/media/base/constants.cc

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/*
* libjingle
* Copyright 2012 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
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
#include "webrtc/media/base/constants.h"
#include <string>
namespace cricket {
const int kVideoCodecClockrate = 90000;
const int kDataCodecClockrate = 90000;
const int kDataMaxBandwidth = 30720; // bps
const float kHighSystemCpuThreshold = 0.85f;
const float kLowSystemCpuThreshold = 0.65f;
const float kProcessCpuThreshold = 0.10f;
const char kRtxCodecName[] = "rtx";
const char kRedCodecName[] = "red";
const char kUlpfecCodecName[] = "ulpfec";
const char kCodecParamAssociatedPayloadType[] = "apt";
const char kOpusCodecName[] = "opus";
const char kIsacCodecName[] = "isac";
const char kL16CodecName[] = "l16";
const char kG722CodecName[] = "g722";
const char kIlbcCodecName[] = "ilbc";
const char kPcmuCodecName[] = "pcmu";
const char kPcmaCodecName[] = "pcma";
const char kCnCodecName[] = "cn";
const char kDtmfCodecName[] = "telephone-event";
// draft-spittka-payload-rtp-opus-03.txt
const char kCodecParamPTime[] = "ptime";
const char kCodecParamMaxPTime[] = "maxptime";
const char kCodecParamMinPTime[] = "minptime";
const char kCodecParamSPropStereo[] = "sprop-stereo";
const char kCodecParamStereo[] = "stereo";
const char kCodecParamUseInbandFec[] = "useinbandfec";
const char kCodecParamUseDtx[] = "usedtx";
const char kCodecParamMaxAverageBitrate[] = "maxaveragebitrate";
const char kCodecParamMaxPlaybackRate[] = "maxplaybackrate";
const char kCodecParamSctpProtocol[] = "protocol";
const char kCodecParamSctpStreams[] = "streams";
const char kParamValueTrue[] = "1";
const char kParamValueEmpty[] = "";
const int kOpusDefaultMaxPTime = 120;
const int kOpusDefaultPTime = 20;
const int kOpusDefaultMinPTime = 3;
const int kOpusDefaultSPropStereo = 0;
const int kOpusDefaultStereo = 0;
const int kOpusDefaultUseInbandFec = 0;
const int kOpusDefaultUseDtx = 0;
const int kOpusDefaultMaxPlaybackRate = 48000;
const int kPreferredMaxPTime = 60;
const int kPreferredMinPTime = 10;
const int kPreferredSPropStereo = 0;
const int kPreferredStereo = 0;
const int kPreferredUseInbandFec = 0;
const char kRtcpFbParamNack[] = "nack";
const char kRtcpFbNackParamPli[] = "pli";
const char kRtcpFbParamRemb[] = "goog-remb";
const char kRtcpFbParamTransportCc[] = "transport-cc";
const char kRtcpFbParamCcm[] = "ccm";
const char kRtcpFbCcmParamFir[] = "fir";
const char kCodecParamMaxBitrate[] = "x-google-max-bitrate";
const char kCodecParamMinBitrate[] = "x-google-min-bitrate";
const char kCodecParamStartBitrate[] = "x-google-start-bitrate";
const char kCodecParamMaxQuantization[] = "x-google-max-quantization";
const char kCodecParamPort[] = "x-google-port";
const int kGoogleRtpDataCodecId = 101;
const char kGoogleRtpDataCodecName[] = "google-data";
const int kGoogleSctpDataCodecId = 108;
const char kGoogleSctpDataCodecName[] = "google-sctp-data";
const char kComfortNoiseCodecName[] = "CN";
const int kRtpAudioLevelHeaderExtensionDefaultId = 1;
const char kRtpAudioLevelHeaderExtension[] =
"urn:ietf:params:rtp-hdrext:ssrc-audio-level";
const int kRtpTimestampOffsetHeaderExtensionDefaultId = 2;
const char kRtpTimestampOffsetHeaderExtension[] =
"urn:ietf:params:rtp-hdrext:toffset";
const int kRtpAbsoluteSenderTimeHeaderExtensionDefaultId = 3;
const char kRtpAbsoluteSenderTimeHeaderExtension[] =
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
const int kRtpVideoRotationHeaderExtensionDefaultId = 4;
const char kRtpVideoRotationHeaderExtension[] = "urn:3gpp:video-orientation";
const char kRtpVideoRotation6BitsHeaderExtensionForTesting[] =
"urn:3gpp:video-orientation:6";
const int kRtpTransportSequenceNumberHeaderExtensionDefaultId = 5;
const char kRtpTransportSequenceNumberHeaderExtension[] =
"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
const char kVp8CodecName[] = "VP8";
const char kVp9CodecName[] = "VP9";
const char kH264CodecName[] = "H264";
const int kDefaultVp8PlType = 100;
const int kDefaultVp9PlType = 101;
const int kDefaultH264PlType = 107;
const int kDefaultRedPlType = 116;
const int kDefaultUlpfecType = 117;
const int kDefaultRtxVp8PlType = 96;
const int kDefaultRtxVp9PlType = 97;
const int kDefaultRtxRedPlType = 98;
const int kDefaultVideoMaxWidth = 640;
const int kDefaultVideoMaxHeight = 400;
const int kDefaultVideoMaxFramerate = 30;
} // namespace cricket