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const shuffleArray = function(array) { | |
const a = array.slice(); | |
for (let i = a.length - 1; i > 0; i--) { | |
const j = Math.floor(Math.random() * (i + 1)); | |
[a[i], a[j]] = [a[j], a[i]]; | |
} | |
return a; | |
}; |
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//"avcoti" hexadecimal representation of the following three bytes in the (subset) sequence parameter set Network Abstraction Layer (NAL) unit specified in AVC: 1.profile_idc, 2.the byte containing the constraint_set flags (currently constraint_set0_flag through constraint_set5_flag, and the reserved_zero_2bits), 3.level_idc. | |
AVC1_CODEC_MAP = { | |
,"avc1.66.30": {profile:"Baseline", level:3.0, max_bit_rate:10000} //iOS friendly variation (iOS 3.0-3.1.2) | |
"avc1.42001e": {profile:"Baseline", level:3.0, max_bit_rate:10000} | |
,"avc1.42001f": {profile:"Baseline", level:3.1, max_bit_rate:14000} | |
//other variations | |
,"avc1.77.30": {profile:"Main", level:3.0, max_bit_rate:10000} //iOS friendly variation (iOS 3.0-3.1.2) | |
,"avc1.4d001e": {profile:"Main", level:3.0, max_bit_rate:10000} | |
,"avc1.4d001f": {profile:"Main", level:3.1, max_bit_rate:14000} |
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#include <websocketpp/config/asio_no_tls_client.hpp> | |
#include <websocketpp/client.hpp> | |
#include <iostream> | |
typedef websocketpp::client<websocketpp::config::asio_client> client; | |
using websocketpp::lib::placeholders::_1; | |
using websocketpp::lib::placeholders::_2; |
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@echo off | |
setlocal | |
set _WEBRTC_VERSION=86 | |
set _BRANCH_NUMBER=4240 | |
set _DEPOT_TOOLS_COMMIT=master | |
set ARGS_RELEASE="use_rtti=true is_debug=false use_custom_libcxx=false rtc_use_h264=true ffmpeg_branding=\"Chrome\" rtc_include_tests=false rtc_build_tools=false rtc_build_examples=false" | |
set ARGS_DEBUG="use_rtti=true is_debug=true use_custom_libcxx=false rtc_use_h264=true ffmpeg_branding=\"Chrome\" enable_iterator_debugging=true rtc_include_tests=false rtc_build_tools=false rtc_build_examples=false" |
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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, |
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#include "audiorenderer.h" | |
#include "videorenderer.h" | |
#include "audiocapturemodule.h" | |
#include "yuvframecapture.h" | |
#include "conductor.h" | |
#include "webrtc/api/test/fakeconstraints.h" | |
#include "webrtc/video_encoder.h" | |
#include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h" |
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#include <chrono> | |
auto t1 = std::chrono::high_resolution_clock::now(); | |
// code to measure | |
auto t2 = std::chrono::high_resolution_clock::now(); | |
auto dur = t2 - t1; | |
cout << dur.count() / rtc::kNumNanosecsPerMicrosec; |
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script_ver = 2.5 | |
obs = obslua | |
source_name = "" | |
last_text = "" | |
current_frame = 0 | |
time_format_string = "" | |
output_format_string = "" |
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script_ver = 2.7 | |
obs = obslua | |
source_name = "" | |
last_text = "" | |
current_frame = 0 | |
frame_rate = 0 | |
time_format_string = "" | |
output_format_string = "" |
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#!/bin/bash | |
#************************************************ | |
# hdrtosdr.sh * | |
# v1.0.0 * | |
# developer: SCG82 ([email protected]) * | |
#************************************************ | |
USAGE_S="Usage: $0 -s [start time hh:mm:ss] -e [end time hh:mm:ss] -w width -h height -p preset FILE" | |
USAGE="Usage (arguments are optional): $0 -s [start time hh:mm:ss] -e [end time hh:mm:ss] -w [output width] -h [output height] -p [preset: ultrafast,superfast,veryfast,faster,fast,medium,slow,slower,veryslow,placebo] FILE" |
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