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finish 1.1.7 release

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This commit is contained in:
Christian R. Helmrich
2021-08-30 09:00:00 +02:00
parent 9d500c648d
commit acd53a2198
14 changed files with 80 additions and 103 deletions
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2
CMakeLists.txt
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@@ -16,7 +16,7 @@ if("${CMAKE_CURRENT_SOURCE_DIR}" STREQUAL "${CMAKE_CURRENT_BINARY_DIR}")
endif() 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) if(NOT CMAKE_BUILD_TYPE AND NOT CMAKE_CONFIGURATION_TYPES)
set(CMAKE_BUILD_TYPE Release set(CMAKE_BUILD_TYPE Release

23
README.md
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@@ -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 rates, so the lower-rate USAC coding tools (ACELP, TCX, Enhanced SBR
and MPEG Surround with Unified Stereo coding) won't be integrated. 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 based on the 3-clause BSD license but modified to address particular
aspects dictated by the nature and the output of this application. 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. 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 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. 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): ### Third-party stdin (foobar2000):

11
include/Release.htm
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@@ -25,9 +25,14 @@
<td valign="top"> <td valign="top">
<h1><br><span class="pink">exhale</span> - <span class="pink">e</span>codis e<span class="pink">x</span>tended <span class="pink">h</span>igh-efficiency <span class="pink">a</span>nd <span class="pink">l</span>ow-complexity <span class="pink">e</span>ncoder<br><span class="gray"><sup><br>Software Release Notes, Version History, Known Issues, Upcoming Feature Roadmap</sup></span><br><br></h1> <h1><br><span class="pink">exhale</span> - <span class="pink">e</span>codis e<span class="pink">x</span>tended <span class="pink">h</span>igh-efficiency <span class="pink">a</span>nd <span class="pink">l</span>ow-complexity <span class="pink">e</span>ncoder<br><span class="gray"><sup><br>Software Release Notes, Version History, Known Issues, Upcoming Feature Roadmap</sup></span><br><br></h1>
<h3>&nbsp; &nbsp;The version of this distribution of the &laquo;exhale&raquo; software release is <b>1.1.6</b> (official pub&shy;lic minor release) from May 30, 2021. Please check <a href="http://www.ecodis.de/audio.htm#mpeg">www.ecodis.de</a> 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.</h3> <h3>&nbsp; &nbsp;The version of this distribution of the &laquo;exhale&raquo; software release is <b>1.1.7</b> (official pub&shy;lic minor release) from August 2021. Please check <a href="http://www.ecodis.de/audio.htm#mpeg">www.ecodis.de</a> 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.</h3>
<h3><br><b>Chronological Version History</b></h3> <h3><br><b>Chronological Version History</b></h3>
<h3>&nbsp; &nbsp;Version <b>1.1.6 <span class="gray">&nbsp;May 2021, this release</span></b></h3> <h3>&nbsp; &nbsp;Version <b>1.1.7 <span class="gray">&nbsp;Aug. 2021, this release</span></b></h3>
<ul>
<li><h3>minor tuning at low SBR rates, enabled SBR coding at 22050 Hz input sample rate</h3></li>
<li><h3>exhaleApp: added expert modes for loudness leveling, custom Intra frame interval</h3></li>
</ul>
<h3>&nbsp; &nbsp;Version <b>1.1.6 <span class="gray">&nbsp;May 2021</span></b></h3>
<ul> <ul>
<li><h3>minor quality tuning and support for delayless operation (media time=0) with SBR</h3></li> <li><h3>minor quality tuning and support for delayless operation (media time=0) with SBR</h3></li>
<li><h3>exhaleApp: fixed very rare output file corruption after finishing encoding with SBR</h3></li> <li><h3>exhaleApp: fixed very rare output file corruption after finishing encoding with SBR</h3></li>
@@ -153,7 +158,7 @@
<li><h3>exhaleLib: speed-ups and further quality tuning for difficult signals, as necessary.</h3></li> <li><h3>exhaleLib: speed-ups and further quality tuning for difficult signals, as necessary.</h3></li>
</ul> </ul>
<h3><br></h3> <h3><br></h3>
<h4><span class="gray">Written by C. R. Helmrich for exhale 1.1.6, May 2021. Available at www.ecodis.de/exhale/release.htm.</span><br><br></h4> <h4><span class="gray">Written by C. R. Helmrich for exhale 1.1.7, Aug. 2021. Available at www.ecodis.de/exhale/release.htm.</span><br><br></h4>
</td> </td>
<td valign="top" colspan="2"> <td valign="top" colspan="2">

2
include/exhaleDecl.h
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@@ -5,7 +5,7 @@
* and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * 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. * 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_ #ifndef _EXHALE_DECL_H_

2
include/version.h
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@@ -15,5 +15,5 @@
# define EXHALELIB_VERSION_MINOR "1" # define EXHALELIB_VERSION_MINOR "1"
#endif #endif
#ifndef EXHALELIB_VERSION_BUGFIX #ifndef EXHALELIB_VERSION_BUGFIX
# define EXHALELIB_VERSION_BUGFIX ".6" // "RC" or ".0", ".1", ... # define EXHALELIB_VERSION_BUGFIX ".7" // "RC" or ".0", ".1", ...
#endif #endif

2
src/app/CMakeLists.txt
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@@ -5,7 +5,7 @@
# and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- # 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. # 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 add_executable(exhaleApp

2
src/app/basicWavReader.h
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@@ -5,7 +5,7 @@
* and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * 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. * 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_ #ifndef _BASIC_WAV_READER_H_

127
src/app/exhaleApp.cpp
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@@ -72,10 +72,8 @@
#define EA_PEAK_NORM -96.33f // 20 * log10(2^-16), 16-bit normalization #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_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 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 ENABLE_STDOUT_LOAS 0 // 1: experimental LOAS packed pipe output
#define XHE_AAC_LOW_DELAY 0 // 1: allow encoding with 768 frame length #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 #define FULL_FRM_LOOKAHEAD // on: encoder delay = zero or frame length
static const int16_t usfc2x[32] = { // 2x upsampling filter coefficients 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, static uint16_t eaApplyLevelNorm (int32_t* /*o*/ pcmBuffer, uint16_t* oldLoudness, const uint16_t currLoudness,
const uint16_t frameSize, const uint16_t numChannels) 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); return (*oldLoudness = currLoudness);
} }
#if ENABLE_STDOUT_LOAS
static uint16_t eaInitLoasHeader (uint8_t* const loasHeader, // sets up LATM/LOAS header, returns payload offset 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) 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); const bool readStdin = (argc == 3 || argc == 5);
BasicWavReader wavReader; BasicWavReader wavReader;
int32_t* inPcmData = nullptr; // 24-bit WAVE audio input buffer int32_t* inPcmData = nullptr; // 24-bit WAVE audio input buffer
#if ENABLE_RESAMPLING
int32_t* inPcmRsmp = nullptr; // temporary buffer for resampler int32_t* inPcmRsmp = nullptr; // temporary buffer for resampler
#endif
uint8_t* outAuData = nullptr; // access unit (AU) output buffer uint8_t* outAuData = nullptr; // access unit (AU) output buffer
int inFileHandle = -1, outFileHandle = -1; int inFileHandle = -1, outFileHandle = -1;
uint32_t loudStats = EA_LOUD_INIT; // valid empty loudness data uint32_t loudStats = EA_LOUD_INIT; // valid empty loudness data
@@ -342,18 +337,17 @@ int main (const int argc, char* argv[])
# endif # endif
const char* exePath = argv[0]; const char* exePath = argv[0];
#endif #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 compatibleExtensionFlag = 0; // 0: disabled, 1: enabled
uint16_t coreSbrFrameLengthIndex = 1; // 0: 768, 1: 1024 samples uint16_t coreSbrFrameLengthIndex = 1; // 0: 768, 1: 1024 samples
uint16_t variableCoreBitRateMode = 3; // 0: lowest... 9: highest 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); uint8_t zeroDelayForSbrEncoding = (argc >= 5 && (argv[2][0] == 's' || argv[2][0] == 'S') && argv[2][1] == 0 ? 1 : 0);
#endif
#if ENABLE_STDOUT_LOAS #if ENABLE_STDOUT_LOAS
const bool writeStdout = (zeroDelayForSbrEncoding != 0 && argv[1][0] >= 'a' && argv[argc - 1][0] == '-' && argv[argc - 1][1] == 0); 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}; uint8_t loasHeader[64] = {0};
#endif #endif
bool enableLufsLevel = (argc >= 5 && (argv[2][0] == 'l' || argv[2][0] == 'L') && argv[2][1] == 0);
#ifdef EXHALE_APP_WIN #ifdef EXHALE_APP_WIN
const HANDLE hConsole = GetStdHandle (STD_OUTPUT_HANDLE); const HANDLE hConsole = GetStdHandle (STD_OUTPUT_HANDLE);
CONSOLE_SCREEN_BUFFER_INFO csbi; CONSOLE_SCREEN_BUFFER_INFO csbi;
@@ -704,10 +698,7 @@ int main (const int argc, char* argv[])
# endif # endif
if ((wavReader.getSampleRate () > 32100 + (unsigned) variableCoreBitRateMode * 12000 + (variableCoreBitRateMode >> 2) * 3900) && if ((wavReader.getSampleRate () > 32100 + (unsigned) variableCoreBitRateMode * 12000 + (variableCoreBitRateMode >> 2) * 3900) &&
# if ENABLE_RESAMPLING (variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) && !enableSbrCoding)
(variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) &&
# endif
!enableSbrCoding)
{ {
i = (variableCoreBitRateMode > 4 ? 96 : __min (64, 32 + variableCoreBitRateMode * 12)); i = (variableCoreBitRateMode > 4 ? 96 : __min (64, 32 + variableCoreBitRateMode * 12));
# ifdef EXHALE_APP_WCHAR # ifdef EXHALE_APP_WCHAR
@@ -721,14 +712,14 @@ int main (const int argc, char* argv[])
} }
if ((wavReader.getSampleRate () > 32000) && !enableSbrCoding && (variableCoreBitRateMode <= 1)) if ((wavReader.getSampleRate () > 32000) && !enableSbrCoding && (variableCoreBitRateMode <= 1))
{ {
# if ENABLE_RESAMPLING
if (wavReader.getSampleRate () == 48000) if (wavReader.getSampleRate () == 48000)
{ {
_ERROR2 (" NOTE: Downsampling the input audio from 48 kHz to 32 kHz with preset mode %d\n\n", variableCoreBitRateMode); _ERROR2 (" NOTE: Downsampling the input audio from 48 kHz to 32 kHz with preset mode %d\n\n", variableCoreBitRateMode);
} }
else 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 #endif
@@ -757,10 +748,7 @@ int main (const int argc, char* argv[])
} }
if ((wavReader.getSampleRate () > 32100 + (unsigned) variableCoreBitRateMode * 12000 + (variableCoreBitRateMode >> 2) * 3900) && if ((wavReader.getSampleRate () > 32100 + (unsigned) variableCoreBitRateMode * 12000 + (variableCoreBitRateMode >> 2) * 3900) &&
#if ENABLE_RESAMPLING (variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) && !enableSbrCoding)
(variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) &&
#endif
!enableSbrCoding)
{ {
i = (variableCoreBitRateMode > 4 ? 96 : __min (64, 32 + variableCoreBitRateMode * 12)); i = (variableCoreBitRateMode > 4 ? 96 : __min (64, 32 + variableCoreBitRateMode * 12));
#ifdef EXHALE_APP_WCHAR #ifdef EXHALE_APP_WCHAR
@@ -774,14 +762,14 @@ int main (const int argc, char* argv[])
} }
if ((wavReader.getSampleRate () > 32000) && !enableSbrCoding && (variableCoreBitRateMode <= 1)) if ((wavReader.getSampleRate () > 32000) && !enableSbrCoding && (variableCoreBitRateMode <= 1))
{ {
#if ENABLE_RESAMPLING
if (wavReader.getSampleRate () == 48000) 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); fprintf_s (stdout, " NOTE: Downsampling the input audio from 48 kHz to 32 kHz with preset mode %d\n\n", variableCoreBitRateMode);
} }
else 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 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 numChannels = wavReader.getNumChannels ();
const unsigned inSampDepth = wavReader.getBitDepth (); const unsigned inSampDepth = wavReader.getBitDepth ();
const unsigned sbrEncDelay = (enableSbrCoding ? 962 : 0); const unsigned sbrEncDelay = (enableSbrCoding ? 962 : 0);
#if ENABLE_RESAMPLING
const bool enableUpsampler = eaInitUpsampler2x (&inPcmRsmp, variableCoreBitRateMode, i, frameLength, numChannels); const bool enableUpsampler = eaInitUpsampler2x (&inPcmRsmp, variableCoreBitRateMode, i, frameLength, numChannels);
const bool enableResampler = (enableSbrCoding ? false : // no 3:2 downsampling required when encoding in SBR mode const bool enableResampler = (enableSbrCoding ? false : // no 3:2 downsampling required when encoding in SBR mode
eaInitDownsampler (&inPcmRsmp, variableCoreBitRateMode, i, frameLength, numChannels)); 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 inFrameSize = (enableResampler ? startLength : frameLength) * sizeof (int32_t); // max buffer size
const unsigned resampRatio = (enableResampler ? 3 : 1); // for resampling ratio const unsigned resampRatio = (enableResampler ? 3 : 1); // for resampling ratio
const unsigned resampShift = (enableResampler || enableUpsampler ? 1 : 0); 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) + const uint16_t inPadLength = uint16_t ((((frameLength << 1) - startLength) * resampRatio) >> resampShift) +
(enableSbrCoding ? firstLength - (sbrEncDelay >> resampShift) - (enableUpsampler ? 32 : 0) : 0); (enableSbrCoding ? firstLength - (sbrEncDelay >> resampShift) - (enableUpsampler ? 32 : 0) : 0);
# endif #endif
const int64_t expectLength = (wavReader.getDataBytesLeft () << resampShift) / int64_t ((numChannels * inSampDepth * resampRatio) >> 3); const int64_t expectLength = (wavReader.getDataBytesLeft () << resampShift) / int64_t ((numChannels * inSampDepth * resampRatio) >> 3);
if (enableUpsampler) // notify by printf if (enableUpsampler) // notify by printf
{ {
# if ENABLE_STDOUT_LOAS #if ENABLE_STDOUT_LOAS
if (writeStdout) // relocate to stderr 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", 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", 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); i / 1000, i / 500, variableCoreBitRateMode);
} }
else 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); 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 // allocate dynamic frame memory buffers
inPcmData = (int32_t*) malloc (inFrameSize * numChannels); // max frame in size inPcmData = (int32_t*) malloc (inFrameSize * numChannels); // max frame in size
#ifdef NO_PREROLL_DATA #ifdef NO_PREROLL_DATA
@@ -896,16 +880,12 @@ int main (const int argc, char* argv[])
goto mainFinish; // memory alloc error 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 memset (inPcmData, 0, inPadLength * numChannels * sizeof (int32_t)); // padding
if (inPadLength + wavReader.read (inPcmData + inPadLength * numChannels, firstLength - inPadLength) != firstLength) if (inPadLength + wavReader.read (inPcmData + inPadLength * numChannels, firstLength - inPadLength) != firstLength)
# else
if (wavReader.read (inPcmData, firstLength) != firstLength) // full first frame
# endif
#else #else
if (wavReader.read (inPcmData, frameLength) != frameLength) // full first frame if (wavReader.read (inPcmData, firstLength) != firstLength) // full first frame
#endif #endif
{ {
_ERROR1 (" ERROR while trying to encode input audio data! The audio stream is too short!\n\n"); _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 else // start coding loop, show progress
{ {
#if ENABLE_RESAMPLING
const unsigned sampleRate = (wavReader.getSampleRate () << resampShift) / resampRatio; 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 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); const unsigned indepPeriod = (userIndepPeriod ? 10 * (argv[3][0] - 48) + (argv[3][1] - 48) : (sampleRate < 48000 ? sampleRate - 320u : 50u << 10u) / frameLength);
#if ENABLE_STDOUT_LOAS #if ENABLE_STDOUT_LOAS
@@ -939,9 +915,7 @@ int main (const int argc, char* argv[])
#else #else
ExhaleEncoder exhaleEnc (inPcmData, outAuData, sampleRate, numChannels, frameLength, indepPeriod, variableCoreBitRateMode ExhaleEncoder exhaleEnc (inPcmData, outAuData, sampleRate, numChannels, frameLength, indepPeriod, variableCoreBitRateMode
#endif #endif
#if ENABLE_RESAMPLING
+ (enableUpsampler && (variableCoreBitRateMode < 9) ? 1 : 0) + (enableUpsampler && (variableCoreBitRateMode < 9) ? 1 : 0)
#endif
#if !RESTRICT_TO_AAC #if !RESTRICT_TO_AAC
, !(argc >= 5 && (argv[2][0] == 'n' || argv[2][0] == 'N') && argv[2][1] == 0), compatibleExtensionFlag > 0 , !(argc >= 5 && (argv[2][0] == 'n' || argv[2][0] == 'N') && argv[2][1] == 0), compatibleExtensionFlag > 0
#endif #endif
@@ -976,6 +950,7 @@ int main (const int argc, char* argv[])
{ {
br = 0; // init frame count & header br = 0; // init frame count & header
if ((loasMuxOffset = eaInitLoasHeader (loasHeader, outAuData, bw)) == 0) i = 1; if ((loasMuxOffset = eaInitLoasHeader (loasHeader, outAuData, bw)) == 0) i = 1;
else enableLufsLevel = true;
} }
else else
#endif #endif
@@ -1046,12 +1021,11 @@ int main (const int argc, char* argv[])
i = 1; // for progress bar i = 1; // for progress bar
#if ENABLE_RESAMPLING
// resample initial frame if necessary // resample initial frame if necessary
if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels, !zeroDelayForSbrEncoding); if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels, !zeroDelayForSbrEncoding);
else else
if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels, !zeroDelayForSbrEncoding); if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels, !zeroDelayForSbrEncoding);
#endif
// initial frame, encode look-ahead AU // initial frame, encode look-ahead AU
if ((bw = exhaleEnc.encodeLookahead ()) < 3) if ((bw = exhaleEnc.encodeLookahead ()) < 3)
{ {
@@ -1093,6 +1067,7 @@ int main (const int argc, char* argv[])
#endif #endif
#ifdef NO_PREROLL_DATA #ifdef NO_PREROLL_DATA
if (bwMax < bw) bwMax = bw; if (bwMax < bw) bwMax = bw;
// write first AU, add frame to header // write first AU, add frame to header
if ((mp4Writer.addFrameAU (outAuData, bw) != (int) bw) || loudnessEst.addNewPcmData (frameLength)) 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 goto mainFinish; // estimation error
} }
#endif #endif
#if ENABLE_RESAMPLING
while (wavReader.read (inPcmData, (frameLength * resampRatio) >> resampShift) > 0) // read a new audio frame 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 // resample audio frame if necessary
if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels); if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels);
else else
if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels); if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels);
#endif
// frame coding loop, encode next AU // frame coding loop, encode next AU
loudnessEst.addNewPcmData (frameLength); loudnessEst.addNewPcmData (frameLength);
#if ENABLE_STDOUT_LOAS if (enableLufsLevel) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels);
if (writeStdout) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels);
#endif
if ((bw = exhaleEnc.encodeFrame ()) < 3) if ((bw = exhaleEnc.encodeFrame ()) < 3)
{ {
_ERROR2 ("\n ERROR while trying to create xHE-AAC frame: error value %d was returned!\n\n", bw); _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 goto mainFinish; // encoding error
} }
if (bwMax < bw) bwMax = bw; if (bwMax < bw) bwMax = bw;
// write new AU, add frame to header // write new AU, add frame to header
#if ENABLE_STDOUT_LOAS #if ENABLE_STDOUT_LOAS
if (writeStdout) if (writeStdout)
@@ -1171,17 +1142,15 @@ int main (const int argc, char* argv[])
} }
} // frame loop } // frame loop
#if ENABLE_RESAMPLING
// resample the last frame if necessary // resample the last frame if necessary
if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels); if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels);
else else
if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels); if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels);
#endif
// end of coding loop, encode final AU // end of coding loop, encode final AU
loudnessEst.addNewPcmData (frameLength); loudnessEst.addNewPcmData (frameLength);
#if ENABLE_STDOUT_LOAS if (enableLufsLevel) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels);
if (writeStdout) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels);
#endif
if ((bw = exhaleEnc.encodeFrame ()) < 3) if ((bw = exhaleEnc.encodeFrame ()) < 3)
{ {
_ERROR2 ("\n ERROR while trying to create xHE-AAC frame: error value %d was returned!\n\n", bw); _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 goto mainFinish; // coder-time error
} }
if (bwMax < bw) bwMax = bw; if (bwMax < bw) bwMax = bw;
// write final AU, add frame to header // write final AU, add frame to header
#if ENABLE_STDOUT_LOAS #if ENABLE_STDOUT_LOAS
if (writeStdout) if (writeStdout)
@@ -1216,38 +1186,30 @@ int main (const int argc, char* argv[])
} }
byteCount += bw; byteCount += bw;
#if ENABLE_RESAMPLING
const int64_t actualLength = (wavReader.getDataBytesRead () << resampShift) / int64_t ((numChannels * inSampDepth * resampRatio) >> 3); 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 int64_t inFileLength = wavReader.getDataBytesRead () / int64_t ((numChannels * inSampDepth) >> 3);
const unsigned inFrmLength = (frameLength * resampRatio) >> resampShift; const unsigned inFrmLength = (frameLength * resampRatio) >> resampShift;
const unsigned resampDelay = (enableUpsampler ? 32 : (enableResampler ? 64 : 0)); const unsigned resampDelay = (enableUpsampler ? 32 : (enableResampler ? 64 : 0));
# ifdef FULL_FRM_LOOKAHEAD #ifdef FULL_FRM_LOOKAHEAD
const unsigned flushLength = (inFileLength - resampDelay + inPadLength) % inFrmLength; const unsigned flushLength = (inFileLength - resampDelay + inPadLength) % inFrmLength;
# else #else
const unsigned flushLength = (inFileLength - resampDelay) % inFrmLength; const unsigned flushLength = (inFileLength - resampDelay) % inFrmLength;
# endif #endif
if ((flushLength + ((startLength * resampRatio) >> resampShift) - inFrmLength + resampDelay if ((flushLength + ((startLength * resampRatio) >> resampShift) - inFrmLength + resampDelay
- (resampDelay >> 6)/*rnd*/+ wavReader.getSampleRate () / 200 > inFrmLength - (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 - ((2 + sbrEncDelay * 3) >> 2)) || (flushLength == 0)) // flush last frame
{ {
memset (inPcmData, 0, inFrameSize * numChannels); memset (inPcmData, 0, inFrameSize * numChannels);
#if ENABLE_RESAMPLING
// resample flush frame if necessary // resample flush frame if necessary
if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels); if (enableUpsampler) eaApplyUpsampler2x (inPcmData, inPcmRsmp, frameLength, numChannels);
else else
if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels); if (enableResampler) eaApplyDownsampler (inPcmData, inPcmRsmp, frameLength, numChannels);
#endif
// flush remaining audio into new AU // flush remaining audio into new AU
// no loudnessEst.addNewPcmData call // no loudnessEst.addNewPcmData call
#if ENABLE_STDOUT_LOAS if (enableLufsLevel) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels);
if (writeStdout) eaApplyLevelNorm (inPcmData, &loudMemory, loudnessEst.getStatistics () >> 16, frameLength, numChannels);
#endif
if ((bw = exhaleEnc.encodeFrame ()) < 3) if ((bw = exhaleEnc.encodeFrame ()) < 3)
{ {
_ERROR2 ("\n ERROR while trying to create last xHE-AAC frame: error value %d was returned!\n\n", bw); _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 goto mainFinish; // encoding error
} }
if (bwMax < bw) bwMax = bw; if (bwMax < bw) bwMax = bw;
// the flush AU, add frame to header // the flush AU, add frame to header
#if ENABLE_STDOUT_LOAS #if ENABLE_STDOUT_LOAS
if (writeStdout) if (writeStdout)
@@ -1329,8 +1292,10 @@ int main (const int argc, char* argv[])
#endif #endif
{ {
// quantize for loudnessInfo() reset // 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); 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 // recreate ASC + UC + loudness data
bw |= (qPeak << 18) | (qLoud << 6) | 11; // measurementSystem & reliability bw |= (qPeak << 18) | (qLoud << 6) | 11; // measurementSystem & reliability
@@ -1404,13 +1369,11 @@ mainFinish:
free ((void*) inPcmData); free ((void*) inPcmData);
inPcmData = nullptr; inPcmData = nullptr;
} }
#if ENABLE_RESAMPLING
if (inPcmRsmp != nullptr) if (inPcmRsmp != nullptr)
{ {
free ((void*) inPcmRsmp); free ((void*) inPcmRsmp);
inPcmRsmp = nullptr; inPcmRsmp = nullptr;
} }
#endif
#if EA_USE_WORK_DIR #if EA_USE_WORK_DIR
if (currPath != nullptr) if (currPath != nullptr)
{ {

2
src/app/exhaleApp.rc
View File

@@ -13,7 +13,7 @@
0 ICON "exhaleApp.ico" 0 ICON "exhaleApp.ico"
VS_VERSION_INFO VERSIONINFO VS_VERSION_INFO VERSIONINFO
FILEVERSION 1,1,6,2 FILEVERSION 1,1,7
BEGIN BEGIN
BLOCK "StringFileInfo" BLOCK "StringFileInfo"
BEGIN BEGIN

2
src/app/exhaleAppPch.cpp
View File

@@ -5,7 +5,7 @@
* and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * 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. * 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" #include "exhaleAppPch.h"

2
src/app/exhaleAppPch.h
View File

@@ -5,7 +5,7 @@
* and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * 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. * 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_ #ifndef _EXHALE_APP_PCH_H_

2
src/app/loudnessEstim.cpp
View File

@@ -5,7 +5,7 @@
* and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * 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. * 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" #include "exhaleAppPch.h"

2
src/app/loudnessEstim.h
View File

@@ -5,7 +5,7 @@
* and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- * 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. * 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_ #ifndef _LOUDNESS_ESTIM_H_

2
src/app/makefile
View File

@@ -5,7 +5,7 @@
# and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third- # 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. # 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 # define as console application