From acd53a21980828f202792567694b47d4362a910f Mon Sep 17 00:00:00 2001 From: "Christian R. Helmrich" Date: Mon, 30 Aug 2021 09:00:00 +0200 Subject: [PATCH] finish 1.1.7 release --- CMakeLists.txt | 2 +- README.md | 23 ++++--- include/Release.htm | 11 +++- include/exhaleDecl.h | 2 +- include/version.h | 2 +- src/app/CMakeLists.txt | 2 +- src/app/basicWavReader.h | 2 +- src/app/exhaleApp.cpp | 127 ++++++++++++++------------------------ src/app/exhaleApp.rc | 2 +- src/app/exhaleAppPch.cpp | 2 +- src/app/exhaleAppPch.h | 2 +- src/app/loudnessEstim.cpp | 2 +- src/app/loudnessEstim.h | 2 +- src/app/makefile | 2 +- 14 files changed, 80 insertions(+), 103 deletions(-) diff --git a/CMakeLists.txt b/CMakeLists.txt index a7612ee..3eb4842 100644 --- a/CMakeLists.txt +++ b/CMakeLists.txt @@ -16,7 +16,7 @@ if("${CMAKE_CURRENT_SOURCE_DIR}" STREQUAL "${CMAKE_CURRENT_BINARY_DIR}") endif() -project(exhale VERSION 1.1.6 LANGUAGES CXX) +project(exhale VERSION 1.1.7 LANGUAGES CXX) if(NOT CMAKE_BUILD_TYPE AND NOT CMAKE_CONFIGURATION_TYPES) set(CMAKE_BUILD_TYPE Release diff --git a/README.md b/README.md index 2acf35b..87de6ed 100644 --- a/README.md +++ b/README.md @@ -17,6 +17,14 @@ quality mono, stereo, and multichannel coding at medium and high bit rates, so the lower-rate USAC coding tools (ACELP, TCX, Enhanced SBR and MPEG Surround with Unified Stereo coding) won't be integrated. +**Important:** Due to the missing lower-rate coding tools, the audio +quality at the lowest of exhale's bit-rate modes (18 kbit/s mono, 36 +kbit/s stereo) doesn't reflect the full capabilities of the Extended +HE-AAC standard. Therefore, use the lowest bit-rate modes *only when +required*. Also, please don't attempt to modify exhale's source code +or to configure the command-line encoder to produce lower bit-rates. +Use only existing presets and input sampling rates of 32...48 kHz. + ____________________________________________________________________ @@ -33,7 +41,7 @@ exhale is being made available under an open-source license which is based on the 3-clause BSD license but modified to address particular aspects dictated by the nature and the output of this application. -The license text and release notes for the current version 1.1.6 can +The license text and release notes for the current version 1.1.7 can be found in the `include` subdirectory of the exhale distribution. @@ -102,13 +110,14 @@ to print out usage information. As an example, the following command converts file Input.wav to file Output.m4a at roughly 128 kbit/s (if the input signal is 2-channel stereo) and in Extended HE-AAC format. -Note that, when calling the exhale application with a path (such as, -e.g., `bin/exhale` or `bin\exhale.exe`), but specifying the input or -output file without a file path (e.g., `Input.wav`), those files are -assumed to be located in the application path (here, `bin`). Use the -"dot prefix" to indicate files in the *current* directory instead of -the application directory (here, `./Input.wav` or `.\Input.wav`). +There is also an **expert mode** providing two additional arguments: + +`exhale.exe b s 42 C:\Music\Input.wav C:\Music\Output.m4a` + +e.g. encodes Input.wav to Output.m4a at roughly 48 kbit/s stereo and +with SBR enabled, seamless operation (`s` forces media time 0 in the +edit list), and an independent frame interval of 42 (range 10...99). ### Third-party stdin (foobar2000): diff --git a/include/Release.htm b/include/Release.htm index abc4a87..09188b1 100644 --- a/include/Release.htm +++ b/include/Release.htm @@ -25,9 +25,14 @@


exhale - ecodis extended high-efficiency and low-complexity encoder

Software Release Notes, Version History, Known Issues, Upcoming Feature Roadmap

-

   The version of this distribution of the «exhale» software release is 1.1.6 (official pub­lic minor release) from May 30, 2021. Please check www.ecodis.de regularly for new versions of this software. A summary of each version up to this release, a list of known issues with this release, and a roadmap of additional functionality are provided below.

+

   The version of this distribution of the «exhale» software release is 1.1.7 (official pub­lic minor release) from August 2021. Please check www.ecodis.de regularly for new versions of this software. A summary of each version up to this release, a list of known issues with this release, and a roadmap of additional functionality are provided below.


Chronological Version History

-

   Version 1.1.6  May 2021, this release

+

   Version 1.1.7  Aug. 2021, this release

+ +

   Version 1.1.6  May 2021


-

Written by C. R. Helmrich for exhale 1.1.6, May 2021. Available at www.ecodis.de/exhale/release.htm.

+

Written by C. R. Helmrich for exhale 1.1.7, Aug. 2021. Available at www.ecodis.de/exhale/release.htm.

diff --git a/include/exhaleDecl.h b/include/exhaleDecl.h index dafee88..9f387d1 100644 --- a/include/exhaleDecl.h +++ b/include/exhaleDecl.h @@ -5,7 +5,7 @@ * and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * party rights, including patent rights. No such rights are granted under this License. * - * Copyright (c) 2018-2020 Christian R. Helmrich, project ecodis. All rights reserved. + * Copyright (c) 2018-2021 Christian R. Helmrich, project ecodis. All rights reserved. */ #ifndef _EXHALE_DECL_H_ diff --git a/include/version.h b/include/version.h index d33d4d0..29348e9 100644 --- a/include/version.h +++ b/include/version.h @@ -15,5 +15,5 @@ # define EXHALELIB_VERSION_MINOR "1" #endif #ifndef EXHALELIB_VERSION_BUGFIX -# define EXHALELIB_VERSION_BUGFIX ".6" // "RC" or ".0", ".1", ... +# define EXHALELIB_VERSION_BUGFIX ".7" // "RC" or ".0", ".1", ... #endif diff --git a/src/app/CMakeLists.txt b/src/app/CMakeLists.txt index b28588e..1ebe9b8 100644 --- a/src/app/CMakeLists.txt +++ b/src/app/CMakeLists.txt @@ -5,7 +5,7 @@ # and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- # party rights, including patent rights. No such rights are granted under this License. # - # Copyright (c) 2018-2020 Christian R. Helmrich, project ecodis. All rights reserved. + # Copyright (c) 2018-2021 Christian R. Helmrich, project ecodis. All rights reserved. ## add_executable(exhaleApp diff --git a/src/app/basicWavReader.h b/src/app/basicWavReader.h index 003b2ae..f72d634 100644 --- a/src/app/basicWavReader.h +++ b/src/app/basicWavReader.h @@ -5,7 +5,7 @@ * and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * party rights, including patent rights. No such rights are granted under this License. * - * Copyright (c) 2018-2020 Christian R. Helmrich, project ecodis. All rights reserved. + * Copyright (c) 2018-2021 Christian R. Helmrich, project ecodis. All rights reserved. */ #ifndef _BASIC_WAV_READER_H_ diff --git a/src/app/exhaleApp.cpp b/src/app/exhaleApp.cpp index 4a40bd3..e7d4165 100644 --- a/src/app/exhaleApp.cpp +++ b/src/app/exhaleApp.cpp @@ -72,10 +72,8 @@ #define EA_PEAK_NORM -96.33f // 20 * log10(2^-16), 16-bit normalization #define EA_PEAK_MIN 0.262f // 20 * log10() + EA_PEAK_NORM = -108 dbFS #define EA_USE_WORK_DIR 1 // 1: use working instead of app directory -#define ENABLE_RESAMPLING 1 // 1: automatic input up- and downsampling #define ENABLE_STDOUT_LOAS 0 // 1: experimental LOAS packed pipe output #define XHE_AAC_LOW_DELAY 0 // 1: allow encoding with 768 frame length -#if ENABLE_RESAMPLING #define FULL_FRM_LOOKAHEAD // on: encoder delay = zero or frame length static const int16_t usfc2x[32] = { // 2x upsampling filter coefficients @@ -247,9 +245,7 @@ static void eaApplyDownsampler (int32_t* const pcmBuffer, int32_t* const resampl } } } -#endif // ENABLE_RESAMPLING -#if ENABLE_STDOUT_LOAS static uint16_t eaApplyLevelNorm (int32_t* /*o*/ pcmBuffer, uint16_t* oldLoudness, const uint16_t currLoudness, const uint16_t frameSize, const uint16_t numChannels) { @@ -273,6 +269,7 @@ static uint16_t eaApplyLevelNorm (int32_t* /*o*/ pcmBuffer, uint16_t* oldLoudnes return (*oldLoudness = currLoudness); } +#if ENABLE_STDOUT_LOAS static uint16_t eaInitLoasHeader (uint8_t* const loasHeader, // sets up LATM/LOAS header, returns payload offset const uint8_t* const ascUcBuf, const uint32_t ascUcSize) { @@ -325,9 +322,7 @@ int main (const int argc, char* argv[]) const bool readStdin = (argc == 3 || argc == 5); BasicWavReader wavReader; int32_t* inPcmData = nullptr; // 24-bit WAVE audio input buffer -#if ENABLE_RESAMPLING int32_t* inPcmRsmp = nullptr; // temporary buffer for resampler -#endif uint8_t* outAuData = nullptr; // access unit (AU) output buffer int inFileHandle = -1, outFileHandle = -1; uint32_t loudStats = EA_LOUD_INIT; // valid empty loudness data @@ -342,18 +337,17 @@ int main (const int argc, char* argv[]) # endif const char* exePath = argv[0]; #endif - uint16_t i, exePathEnd = 0; + uint16_t i, exePathEnd = 0, loudMemory = 0; // for LUFS leveling uint16_t compatibleExtensionFlag = 0; // 0: disabled, 1: enabled uint16_t coreSbrFrameLengthIndex = 1; // 0: 768, 1: 1024 samples uint16_t variableCoreBitRateMode = 3; // 0: lowest... 9: highest -#if ENABLE_RESAMPLING uint8_t zeroDelayForSbrEncoding = (argc >= 5 && (argv[2][0] == 's' || argv[2][0] == 'S') && argv[2][1] == 0 ? 1 : 0); -#endif #if ENABLE_STDOUT_LOAS const bool writeStdout = (zeroDelayForSbrEncoding != 0 && argv[1][0] >= 'a' && argv[argc - 1][0] == '-' && argv[argc - 1][1] == 0); - uint16_t loasMuxOffset = 0, loudMemory = 0; + uint16_t loasMuxOffset = 0; uint8_t loasHeader[64] = {0}; #endif + bool enableLufsLevel = (argc >= 5 && (argv[2][0] == 'l' || argv[2][0] == 'L') && argv[2][1] == 0); #ifdef EXHALE_APP_WIN const HANDLE hConsole = GetStdHandle (STD_OUTPUT_HANDLE); CONSOLE_SCREEN_BUFFER_INFO csbi; @@ -704,10 +698,7 @@ int main (const int argc, char* argv[]) # endif if ((wavReader.getSampleRate () > 32100 + (unsigned) variableCoreBitRateMode * 12000 + (variableCoreBitRateMode >> 2) * 3900) && -# if ENABLE_RESAMPLING - (variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) && -# endif - !enableSbrCoding) + (variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) && !enableSbrCoding) { i = (variableCoreBitRateMode > 4 ? 96 : __min (64, 32 + variableCoreBitRateMode * 12)); # ifdef EXHALE_APP_WCHAR @@ -721,14 +712,14 @@ int main (const int argc, char* argv[]) } if ((wavReader.getSampleRate () > 32000) && !enableSbrCoding && (variableCoreBitRateMode <= 1)) { -# if ENABLE_RESAMPLING if (wavReader.getSampleRate () == 48000) { _ERROR2 (" NOTE: Downsampling the input audio from 48 kHz to 32 kHz with preset mode %d\n\n", variableCoreBitRateMode); } else -# endif - _ERROR2 (" WARNING: The input sampling rate should be 32 kHz or less for preset mode %d!\n\n", variableCoreBitRateMode); + { + _ERROR2 (" WARNING: The input sampling rate should be 32 kHz or less for preset mode %d!\n\n", variableCoreBitRateMode); + } } } #endif @@ -757,10 +748,7 @@ int main (const int argc, char* argv[]) } if ((wavReader.getSampleRate () > 32100 + (unsigned) variableCoreBitRateMode * 12000 + (variableCoreBitRateMode >> 2) * 3900) && -#if ENABLE_RESAMPLING - (variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) && -#endif - !enableSbrCoding) + (variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) && !enableSbrCoding) { i = (variableCoreBitRateMode > 4 ? 96 : __min (64, 32 + variableCoreBitRateMode * 12)); #ifdef EXHALE_APP_WCHAR @@ -774,14 +762,14 @@ int main (const int argc, char* argv[]) } if ((wavReader.getSampleRate () > 32000) && !enableSbrCoding && (variableCoreBitRateMode <= 1)) { -#if ENABLE_RESAMPLING if (wavReader.getSampleRate () == 48000) { fprintf_s (stdout, " NOTE: Downsampling the input audio from 48 kHz to 32 kHz with preset mode %d\n\n", variableCoreBitRateMode); } else -#endif - _ERROR2 (" WARNING: The input sampling rate should be 32 kHz or less for preset mode %d!\n\n", variableCoreBitRateMode); + { + _ERROR2 (" WARNING: The input sampling rate should be 32 kHz or less for preset mode %d!\n\n", variableCoreBitRateMode); + } } if (outPathEnd == 0) // name has no path @@ -847,7 +835,6 @@ int main (const int argc, char* argv[]) const unsigned numChannels = wavReader.getNumChannels (); const unsigned inSampDepth = wavReader.getBitDepth (); const unsigned sbrEncDelay = (enableSbrCoding ? 962 : 0); -#if ENABLE_RESAMPLING const bool enableUpsampler = eaInitUpsampler2x (&inPcmRsmp, variableCoreBitRateMode, i, frameLength, numChannels); const bool enableResampler = (enableSbrCoding ? false : // no 3:2 downsampling required when encoding in SBR mode eaInitDownsampler (&inPcmRsmp, variableCoreBitRateMode, i, frameLength, numChannels)); @@ -855,32 +842,29 @@ int main (const int argc, char* argv[]) const unsigned inFrameSize = (enableResampler ? startLength : frameLength) * sizeof (int32_t); // max buffer size const unsigned resampRatio = (enableResampler ? 3 : 1); // for resampling ratio const unsigned resampShift = (enableResampler || enableUpsampler ? 1 : 0); -# ifdef FULL_FRM_LOOKAHEAD +#ifdef FULL_FRM_LOOKAHEAD const uint16_t inPadLength = uint16_t ((((frameLength << 1) - startLength) * resampRatio) >> resampShift) + (enableSbrCoding ? firstLength - (sbrEncDelay >> resampShift) - (enableUpsampler ? 32 : 0) : 0); -# endif +#endif const int64_t expectLength = (wavReader.getDataBytesLeft () << resampShift) / int64_t ((numChannels * inSampDepth * resampRatio) >> 3); if (enableUpsampler) // notify by printf { -# if ENABLE_STDOUT_LOAS +#if ENABLE_STDOUT_LOAS if (writeStdout) // relocate to stderr { -# ifdef EXHALE_APP_WCHAR +# ifdef EXHALE_APP_WCHAR fwprintf_s (stderr, L" NOTE: Upsampling the input audio from %d kHz to %d kHz with preset mode %d\n\n", -# else +# else fprintf_s (stderr, " NOTE: Upsampling the input audio from %d kHz to %d kHz with preset mode %d\n\n", -# endif +# endif i / 1000, i / 500, variableCoreBitRateMode); } else -# endif +#endif fprintf_s (stdout, " NOTE: Upsampling the input audio from %d kHz to %d kHz with preset mode %d\n\n", i / 1000, i / 500, variableCoreBitRateMode); } -#else - const unsigned inFrameSize = frameLength * sizeof (int32_t); // max buffer size - const int64_t expectLength = wavReader.getDataBytesLeft () / int64_t ((numChannels * inSampDepth) >> 3); -#endif + // allocate dynamic frame memory buffers inPcmData = (int32_t*) malloc (inFrameSize * numChannels); // max frame in size #ifdef NO_PREROLL_DATA @@ -896,16 +880,12 @@ int main (const int argc, char* argv[]) goto mainFinish; // memory alloc error } -#if ENABLE_RESAMPLING -# ifdef FULL_FRM_LOOKAHEAD +#ifdef FULL_FRM_LOOKAHEAD memset (inPcmData, 0, inPadLength * numChannels * sizeof (int32_t)); // padding if (inPadLength + wavReader.read (inPcmData + inPadLength * numChannels, firstLength - inPadLength) != firstLength) -# else - if (wavReader.read (inPcmData, firstLength) != firstLength) // full first frame -# endif #else - if (wavReader.read (inPcmData, frameLength) != frameLength) // full first frame + if (wavReader.read (inPcmData, firstLength) != firstLength) // full first frame #endif { _ERROR1 (" ERROR while trying to encode input audio data! The audio stream is too short!\n\n"); @@ -915,11 +895,7 @@ int main (const int argc, char* argv[]) } else // start coding loop, show progress { -#if ENABLE_RESAMPLING const unsigned sampleRate = (wavReader.getSampleRate () << resampShift) / resampRatio; -#else - const unsigned sampleRate = wavReader.getSampleRate (); -#endif const bool userIndepPeriod = (argc >= 5 && argv[3][0] > '0' && argv[3][0] <= '9' && argv[3][1] >= '0' && argv[3][1] <= '9' && argv[3][2] == 0); const unsigned indepPeriod = (userIndepPeriod ? 10 * (argv[3][0] - 48) + (argv[3][1] - 48) : (sampleRate < 48000 ? sampleRate - 320u : 50u << 10u) / frameLength); #if ENABLE_STDOUT_LOAS @@ -939,9 +915,7 @@ int main (const int argc, char* argv[]) #else ExhaleEncoder exhaleEnc (inPcmData, outAuData, sampleRate, numChannels, frameLength, indepPeriod, variableCoreBitRateMode #endif -#if ENABLE_RESAMPLING + (enableUpsampler && (variableCoreBitRateMode < 9) ? 1 : 0) -#endif #if !RESTRICT_TO_AAC , !(argc >= 5 && (argv[2][0] == 'n' || argv[2][0] == 'N') && argv[2][1] == 0), compatibleExtensionFlag > 0 #endif @@ -976,6 +950,7 @@ int main (const int argc, char* argv[]) { br = 0; // init frame count & header if ((loasMuxOffset = eaInitLoasHeader (loasHeader, outAuData, bw)) == 0) i = 1; + else enableLufsLevel = true; } else #endif @@ -1046,12 +1021,11 @@ int main (const int argc, char* argv[]) i = 1; // for progress bar -#if ENABLE_RESAMPLING // resample initial frame if necessary if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels, !zeroDelayForSbrEncoding); else if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels, !zeroDelayForSbrEncoding); -#endif + // initial frame, encode look-ahead AU if ((bw = exhaleEnc.encodeLookahead ()) < 3) { @@ -1093,6 +1067,7 @@ int main (const int argc, char* argv[]) #endif #ifdef NO_PREROLL_DATA if (bwMax < bw) bwMax = bw; + // write first AU, add frame to header if ((mp4Writer.addFrameAU (outAuData, bw) != (int) bw) || loudnessEst.addNewPcmData (frameLength)) { @@ -1111,23 +1086,18 @@ int main (const int argc, char* argv[]) goto mainFinish; // estimation error } #endif -#if ENABLE_RESAMPLING + while (wavReader.read (inPcmData, (frameLength * resampRatio) >> resampShift) > 0) // read a new audio frame -#else - while (wavReader.read (inPcmData, frameLength) > 0) // read a new audio frame -#endif { -#if ENABLE_RESAMPLING // resample audio frame if necessary if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels); else if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels); -#endif + // frame coding loop, encode next AU loudnessEst.addNewPcmData (frameLength); -#if ENABLE_STDOUT_LOAS - if (writeStdout) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels); -#endif + if (enableLufsLevel) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels); + if ((bw = exhaleEnc.encodeFrame ()) < 3) { _ERROR2 ("\n ERROR while trying to create xHE-AAC frame: error value %d was returned!\n\n", bw); @@ -1138,6 +1108,7 @@ int main (const int argc, char* argv[]) goto mainFinish; // encoding error } if (bwMax < bw) bwMax = bw; + // write new AU, add frame to header #if ENABLE_STDOUT_LOAS if (writeStdout) @@ -1171,17 +1142,15 @@ int main (const int argc, char* argv[]) } } // frame loop -#if ENABLE_RESAMPLING // resample the last frame if necessary if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels); else if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels); -#endif + // end of coding loop, encode final AU loudnessEst.addNewPcmData (frameLength); -#if ENABLE_STDOUT_LOAS - if (writeStdout) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels); -#endif + if (enableLufsLevel) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels); + if ((bw = exhaleEnc.encodeFrame ()) < 3) { _ERROR2 ("\n ERROR while trying to create xHE-AAC frame: error value %d was returned!\n\n", bw); @@ -1192,6 +1161,7 @@ int main (const int argc, char* argv[]) goto mainFinish; // coder-time error } if (bwMax < bw) bwMax = bw; + // write final AU, add frame to header #if ENABLE_STDOUT_LOAS if (writeStdout) @@ -1216,38 +1186,30 @@ int main (const int argc, char* argv[]) } byteCount += bw; -#if ENABLE_RESAMPLING const int64_t actualLength = (wavReader.getDataBytesRead () << resampShift) / int64_t ((numChannels * inSampDepth * resampRatio) >> 3); const int64_t inFileLength = wavReader.getDataBytesRead () / int64_t ((numChannels * inSampDepth) >> 3); const unsigned inFrmLength = (frameLength * resampRatio) >> resampShift; const unsigned resampDelay = (enableUpsampler ? 32 : (enableResampler ? 64 : 0)); -# ifdef FULL_FRM_LOOKAHEAD +#ifdef FULL_FRM_LOOKAHEAD const unsigned flushLength = (inFileLength - resampDelay + inPadLength) % inFrmLength; -# else +#else const unsigned flushLength = (inFileLength - resampDelay) % inFrmLength; -# endif +#endif if ((flushLength + ((startLength * resampRatio) >> resampShift) - inFrmLength + resampDelay - (resampDelay >> 6)/*rnd*/+ wavReader.getSampleRate () / 200 > inFrmLength -#else - const int64_t actualLength = wavReader.getDataBytesRead () / int64_t ((numChannels * inSampDepth) >> 3); - const unsigned flushLength = actualLength % frameLength; - - if ((flushLength + startLength - frameLength + sampleRate / 200 > frameLength -#endif - ((2 + sbrEncDelay * 3) >> 2)) || (flushLength == 0)) // flush last frame { memset (inPcmData, 0, inFrameSize * numChannels); -#if ENABLE_RESAMPLING + // resample flush frame if necessary if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels); else if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels); -#endif + // flush remaining audio into new AU // no loudnessEst.addNewPcmData call -#if ENABLE_STDOUT_LOAS - if (writeStdout) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels); -#endif + if (enableLufsLevel) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels); + if ((bw = exhaleEnc.encodeFrame ()) < 3) { _ERROR2 ("\n ERROR while trying to create last xHE-AAC frame: error value %d was returned!\n\n", bw); @@ -1258,6 +1220,7 @@ int main (const int argc, char* argv[]) goto mainFinish; // encoding error } if (bwMax < bw) bwMax = bw; + // the flush AU, add frame to header #if ENABLE_STDOUT_LOAS if (writeStdout) @@ -1329,8 +1292,10 @@ int main (const int argc, char* argv[]) #endif { // quantize for loudnessInfo() reset - const uint32_t qLoud = uint32_t (4.0f * __max (0.0f, (loudStats >> 16) / 512.f + EA_LOUD_NORM) + 0.5f); + const uint32_t qLoud = uint32_t (enableLufsLevel ? 139/* -23 LUFS */ : 4.0f * __max (0.0f, (loudStats >> 16) / 512.f + EA_LOUD_NORM) + 0.5f); const uint32_t qPeak = uint32_t (32.0f * (20.0f - 20.0f * log10 (__max (EA_PEAK_MIN, float (loudStats & USHRT_MAX))) - EA_PEAK_NORM) + 0.5f); + // NOTE: In case of enableLufsLevel, the input peak is also the approximate + // target peak - just as the -23.0 LUFS is the approximate target loudness. // recreate ASC + UC + loudness data bw |= (qPeak << 18) | (qLoud << 6) | 11; // measurementSystem & reliability @@ -1404,13 +1369,11 @@ mainFinish: free ((void*) inPcmData); inPcmData = nullptr; } -#if ENABLE_RESAMPLING if (inPcmRsmp != nullptr) { free ((void*) inPcmRsmp); inPcmRsmp = nullptr; } -#endif #if EA_USE_WORK_DIR if (currPath != nullptr) { diff --git a/src/app/exhaleApp.rc b/src/app/exhaleApp.rc index f971950..6c79c20 100644 --- a/src/app/exhaleApp.rc +++ b/src/app/exhaleApp.rc @@ -13,7 +13,7 @@ 0 ICON "exhaleApp.ico" VS_VERSION_INFO VERSIONINFO -FILEVERSION 1,1,6,2 +FILEVERSION 1,1,7 BEGIN BLOCK "StringFileInfo" BEGIN diff --git a/src/app/exhaleAppPch.cpp b/src/app/exhaleAppPch.cpp index 65b8a9e..e667c39 100644 --- a/src/app/exhaleAppPch.cpp +++ b/src/app/exhaleAppPch.cpp @@ -5,7 +5,7 @@ * and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * party rights, including patent rights. No such rights are granted under this License. * - * Copyright (c) 2018-2020 Christian R. Helmrich, project ecodis. All rights reserved. + * Copyright (c) 2018-2021 Christian R. Helmrich, project ecodis. All rights reserved. */ #include "exhaleAppPch.h" diff --git a/src/app/exhaleAppPch.h b/src/app/exhaleAppPch.h index 55e33d8..370cef6 100644 --- a/src/app/exhaleAppPch.h +++ b/src/app/exhaleAppPch.h @@ -5,7 +5,7 @@ * and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * party rights, including patent rights. No such rights are granted under this License. * - * Copyright (c) 2018-2020 Christian R. Helmrich, project ecodis. All rights reserved. + * Copyright (c) 2018-2021 Christian R. Helmrich, project ecodis. All rights reserved. */ #ifndef _EXHALE_APP_PCH_H_ diff --git a/src/app/loudnessEstim.cpp b/src/app/loudnessEstim.cpp index 197ce72..9f74e39 100644 --- a/src/app/loudnessEstim.cpp +++ b/src/app/loudnessEstim.cpp @@ -5,7 +5,7 @@ * and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * party rights, including patent rights. No such rights are granted under this License. * - * Copyright (c) 2018-2020 Christian R. Helmrich, project ecodis. All rights reserved. + * Copyright (c) 2018-2021 Christian R. Helmrich, project ecodis. All rights reserved. */ #include "exhaleAppPch.h" diff --git a/src/app/loudnessEstim.h b/src/app/loudnessEstim.h index cd67481..c1813f2 100644 --- a/src/app/loudnessEstim.h +++ b/src/app/loudnessEstim.h @@ -5,7 +5,7 @@ * and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * party rights, including patent rights. No such rights are granted under this License. * - * Copyright (c) 2018-2020 Christian R. Helmrich, project ecodis. All rights reserved. + * Copyright (c) 2018-2021 Christian R. Helmrich, project ecodis. All rights reserved. */ #ifndef _LOUDNESS_ESTIM_H_ diff --git a/src/app/makefile b/src/app/makefile index 1806c85..8625a0c 100644 --- a/src/app/makefile +++ b/src/app/makefile @@ -5,7 +5,7 @@ # and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- # party rights, including patent rights. No such rights are granted under this License. # - # Copyright (c) 2018-2020 Christian R. Helmrich, project ecodis. All rights reserved. + # Copyright (c) 2018-2021 Christian R. Helmrich, project ecodis. All rights reserved. ## # define as console application