mirror of
https://gitlab.com/ecodis/exhale.git
synced 2025-02-08 15:48:46 +01:00
finish 1.1.7 release
This commit is contained in:
Christian R. Helmrich
parent
9d500c648d
commit
acd53a2198
@ -16,7 +16,7 @@ if("${CMAKE_CURRENT_SOURCE_DIR}" STREQUAL "${CMAKE_CURRENT_BINARY_DIR}")
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endif()
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project(exhale VERSION 1.1.6 LANGUAGES CXX)
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project(exhale VERSION 1.1.7 LANGUAGES CXX)
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if(NOT CMAKE_BUILD_TYPE AND NOT CMAKE_CONFIGURATION_TYPES)
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set(CMAKE_BUILD_TYPE Release
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23
README.md
23
README.md
@ -17,6 +17,14 @@ quality mono, stereo, and multichannel coding at medium and high bit
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rates, so the lower-rate USAC coding tools (ACELP, TCX, Enhanced SBR
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and MPEG Surround with Unified Stereo coding) won't be integrated.
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**Important:** Due to the missing lower-rate coding tools, the audio
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quality at the lowest of exhale's bit-rate modes (18 kbit/s mono, 36
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kbit/s stereo) doesn't reflect the full capabilities of the Extended
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HE-AAC standard. Therefore, use the lowest bit-rate modes *only when
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required*. Also, please don't attempt to modify exhale's source code
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or to configure the command-line encoder to produce lower bit-rates.
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Use only existing presets and input sampling rates of 32...48 kHz.
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____________________________________________________________________
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@ -33,7 +41,7 @@ exhale is being made available under an open-source license which is
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based on the 3-clause BSD license but modified to address particular
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aspects dictated by the nature and the output of this application.
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The license text and release notes for the current version 1.1.6 can
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The license text and release notes for the current version 1.1.7 can
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be found in the `include` subdirectory of the exhale distribution.
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@ -102,13 +110,14 @@ to print out usage information. As an example, the following command
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converts file Input.wav to file Output.m4a at roughly 128 kbit/s (if
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the input signal is 2-channel stereo) and in Extended HE-AAC format.
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Note that, when calling the exhale application with a path (such as,
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e.g., `bin/exhale` or `bin\exhale.exe`), but specifying the input or
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output file without a file path (e.g., `Input.wav`), those files are
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assumed to be located in the application path (here, `bin`). Use the
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"dot prefix" to indicate files in the *current* directory instead of
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the application directory (here, `./Input.wav` or `.\Input.wav`).
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There is also an **expert mode** providing two additional arguments:
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`exhale.exe b s 42 C:\Music\Input.wav C:\Music\Output.m4a`
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e.g. encodes Input.wav to Output.m4a at roughly 48 kbit/s stereo and
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with SBR enabled, seamless operation (`s` forces media time 0 in the
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edit list), and an independent frame interval of 42 (range 10...99).
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### Third-party stdin (foobar2000):
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@ -25,9 +25,14 @@
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<td valign="top">
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<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>
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<h3> The version of this distribution of the «exhale» software release is <b>1.1.6</b> (official pub­lic 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>
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<h3> The version of this distribution of the «exhale» software release is <b>1.1.7</b> (official pub­lic 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>
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<h3><br><b>Chronological Version History</b></h3>
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<h3> Version <b>1.1.6 <span class="gray"> May 2021, this release</span></b></h3>
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<h3> Version <b>1.1.7 <span class="gray"> Aug. 2021, this release</span></b></h3>
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<ul>
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<li><h3>minor tuning at low SBR rates, enabled SBR coding at 22050 Hz input sample rate</h3></li>
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<li><h3>exhaleApp: added expert modes for loudness leveling, custom Intra frame interval</h3></li>
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</ul>
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<h3> Version <b>1.1.6 <span class="gray"> May 2021</span></b></h3>
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<ul>
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<li><h3>minor quality tuning and support for delayless operation (media time=0) with SBR</h3></li>
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<li><h3>exhaleApp: fixed very rare output file corruption after finishing encoding with SBR</h3></li>
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@ -153,7 +158,7 @@
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<li><h3>exhaleLib: speed-ups and further quality tuning for difficult signals, as necessary.</h3></li>
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</ul>
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<h3><br></h3>
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<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>
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<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>
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</td>
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<td valign="top" colspan="2">
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@ -5,7 +5,7 @@
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* and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third-
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* party rights, including patent rights. No such rights are granted under this License.
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*
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* Copyright (c) 2018-2020 Christian R. Helmrich, project ecodis. All rights reserved.
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* Copyright (c) 2018-2021 Christian R. Helmrich, project ecodis. All rights reserved.
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*/
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#ifndef _EXHALE_DECL_H_
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@ -15,5 +15,5 @@
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# define EXHALELIB_VERSION_MINOR "1"
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#endif
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#ifndef EXHALELIB_VERSION_BUGFIX
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# define EXHALELIB_VERSION_BUGFIX ".6" // "RC" or ".0", ".1", ...
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# define EXHALELIB_VERSION_BUGFIX ".7" // "RC" or ".0", ".1", ...
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#endif
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@ -5,7 +5,7 @@
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# and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third-
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# party rights, including patent rights. No such rights are granted under this License.
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#
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# Copyright (c) 2018-2020 Christian R. Helmrich, project ecodis. All rights reserved.
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# Copyright (c) 2018-2021 Christian R. Helmrich, project ecodis. All rights reserved.
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##
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add_executable(exhaleApp
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@ -5,7 +5,7 @@
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* and comes with ABSOLUTELY NO WARRANTY. This software may be subject to other third-
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* party rights, including patent rights. No such rights are granted under this License.
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*
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* Copyright (c) 2018-2020 Christian R. Helmrich, project ecodis. All rights reserved.
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* Copyright (c) 2018-2021 Christian R. Helmrich, project ecodis. All rights reserved.
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*/
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#ifndef _BASIC_WAV_READER_H_
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@ -72,10 +72,8 @@
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#define EA_PEAK_NORM -96.33f // 20 * log10(2^-16), 16-bit normalization
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#define EA_PEAK_MIN 0.262f // 20 * log10() + EA_PEAK_NORM = -108 dbFS
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#define EA_USE_WORK_DIR 1 // 1: use working instead of app directory
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#define ENABLE_RESAMPLING 1 // 1: automatic input up- and downsampling
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#define ENABLE_STDOUT_LOAS 0 // 1: experimental LOAS packed pipe output
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#define XHE_AAC_LOW_DELAY 0 // 1: allow encoding with 768 frame length
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#if ENABLE_RESAMPLING
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#define FULL_FRM_LOOKAHEAD // on: encoder delay = zero or frame length
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static const int16_t usfc2x[32] = { // 2x upsampling filter coefficients
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@ -247,9 +245,7 @@ static void eaApplyDownsampler (int32_t* const pcmBuffer, int32_t* const resampl
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}
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}
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}
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#endif // ENABLE_RESAMPLING
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#if ENABLE_STDOUT_LOAS
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static uint16_t eaApplyLevelNorm (int32_t* /*o*/ pcmBuffer, uint16_t* oldLoudness, const uint16_t currLoudness,
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const uint16_t frameSize, const uint16_t numChannels)
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{
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@ -273,6 +269,7 @@ static uint16_t eaApplyLevelNorm (int32_t* /*o*/ pcmBuffer, uint16_t* oldLoudnes
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return (*oldLoudness = currLoudness);
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}
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#if ENABLE_STDOUT_LOAS
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static uint16_t eaInitLoasHeader (uint8_t* const loasHeader, // sets up LATM/LOAS header, returns payload offset
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const uint8_t* const ascUcBuf, const uint32_t ascUcSize)
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{
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@ -325,9 +322,7 @@ int main (const int argc, char* argv[])
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const bool readStdin = (argc == 3 || argc == 5);
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BasicWavReader wavReader;
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int32_t* inPcmData = nullptr; // 24-bit WAVE audio input buffer
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#if ENABLE_RESAMPLING
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int32_t* inPcmRsmp = nullptr; // temporary buffer for resampler
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#endif
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uint8_t* outAuData = nullptr; // access unit (AU) output buffer
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int inFileHandle = -1, outFileHandle = -1;
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uint32_t loudStats = EA_LOUD_INIT; // valid empty loudness data
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@ -342,18 +337,17 @@ int main (const int argc, char* argv[])
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# endif
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const char* exePath = argv[0];
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#endif
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uint16_t i, exePathEnd = 0;
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uint16_t i, exePathEnd = 0, loudMemory = 0; // for LUFS leveling
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uint16_t compatibleExtensionFlag = 0; // 0: disabled, 1: enabled
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uint16_t coreSbrFrameLengthIndex = 1; // 0: 768, 1: 1024 samples
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uint16_t variableCoreBitRateMode = 3; // 0: lowest... 9: highest
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#if ENABLE_RESAMPLING
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uint8_t zeroDelayForSbrEncoding = (argc >= 5 && (argv[2][0] == 's' || argv[2][0] == 'S') && argv[2][1] == 0 ? 1 : 0);
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#endif
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#if ENABLE_STDOUT_LOAS
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const bool writeStdout = (zeroDelayForSbrEncoding != 0 && argv[1][0] >= 'a' && argv[argc - 1][0] == '-' && argv[argc - 1][1] == 0);
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uint16_t loasMuxOffset = 0, loudMemory = 0;
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uint16_t loasMuxOffset = 0;
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uint8_t loasHeader[64] = {0};
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#endif
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bool enableLufsLevel = (argc >= 5 && (argv[2][0] == 'l' || argv[2][0] == 'L') && argv[2][1] == 0);
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#ifdef EXHALE_APP_WIN
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const HANDLE hConsole = GetStdHandle (STD_OUTPUT_HANDLE);
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CONSOLE_SCREEN_BUFFER_INFO csbi;
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@ -704,10 +698,7 @@ int main (const int argc, char* argv[])
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# endif
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if ((wavReader.getSampleRate () > 32100 + (unsigned) variableCoreBitRateMode * 12000 + (variableCoreBitRateMode >> 2) * 3900) &&
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# if ENABLE_RESAMPLING
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(variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) &&
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# endif
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!enableSbrCoding)
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(variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) && !enableSbrCoding)
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{
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i = (variableCoreBitRateMode > 4 ? 96 : __min (64, 32 + variableCoreBitRateMode * 12));
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# ifdef EXHALE_APP_WCHAR
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@ -721,14 +712,14 @@ int main (const int argc, char* argv[])
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}
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if ((wavReader.getSampleRate () > 32000) && !enableSbrCoding && (variableCoreBitRateMode <= 1))
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{
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# if ENABLE_RESAMPLING
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if (wavReader.getSampleRate () == 48000)
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{
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_ERROR2 (" NOTE: Downsampling the input audio from 48 kHz to 32 kHz with preset mode %d\n\n", variableCoreBitRateMode);
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}
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else
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# endif
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_ERROR2 (" WARNING: The input sampling rate should be 32 kHz or less for preset mode %d!\n\n", variableCoreBitRateMode);
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{
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_ERROR2 (" WARNING: The input sampling rate should be 32 kHz or less for preset mode %d!\n\n", variableCoreBitRateMode);
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}
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}
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}
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#endif
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@ -757,10 +748,7 @@ int main (const int argc, char* argv[])
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}
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if ((wavReader.getSampleRate () > 32100 + (unsigned) variableCoreBitRateMode * 12000 + (variableCoreBitRateMode >> 2) * 3900) &&
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#if ENABLE_RESAMPLING
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(variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) &&
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#endif
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!enableSbrCoding)
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(variableCoreBitRateMode > 1 || wavReader.getSampleRate () != 48000) && !enableSbrCoding)
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{
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i = (variableCoreBitRateMode > 4 ? 96 : __min (64, 32 + variableCoreBitRateMode * 12));
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#ifdef EXHALE_APP_WCHAR
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@ -774,14 +762,14 @@ int main (const int argc, char* argv[])
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}
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if ((wavReader.getSampleRate () > 32000) && !enableSbrCoding && (variableCoreBitRateMode <= 1))
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{
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#if ENABLE_RESAMPLING
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if (wavReader.getSampleRate () == 48000)
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{
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fprintf_s (stdout, " NOTE: Downsampling the input audio from 48 kHz to 32 kHz with preset mode %d\n\n", variableCoreBitRateMode);
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}
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else
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#endif
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_ERROR2 (" WARNING: The input sampling rate should be 32 kHz or less for preset mode %d!\n\n", variableCoreBitRateMode);
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{
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_ERROR2 (" WARNING: The input sampling rate should be 32 kHz or less for preset mode %d!\n\n", variableCoreBitRateMode);
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}
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}
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|
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if (outPathEnd == 0) // name has no path
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@ -847,7 +835,6 @@ int main (const int argc, char* argv[])
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const unsigned numChannels = wavReader.getNumChannels ();
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const unsigned inSampDepth = wavReader.getBitDepth ();
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const unsigned sbrEncDelay = (enableSbrCoding ? 962 : 0);
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#if ENABLE_RESAMPLING
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const bool enableUpsampler = eaInitUpsampler2x (&inPcmRsmp, variableCoreBitRateMode, i, frameLength, numChannels);
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const bool enableResampler = (enableSbrCoding ? false : // no 3:2 downsampling required when encoding in SBR mode
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eaInitDownsampler (&inPcmRsmp, variableCoreBitRateMode, i, frameLength, numChannels));
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@ -855,32 +842,29 @@ int main (const int argc, char* argv[])
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const unsigned inFrameSize = (enableResampler ? startLength : frameLength) * sizeof (int32_t); // max buffer size
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const unsigned resampRatio = (enableResampler ? 3 : 1); // for resampling ratio
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const unsigned resampShift = (enableResampler || enableUpsampler ? 1 : 0);
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# ifdef FULL_FRM_LOOKAHEAD
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#ifdef FULL_FRM_LOOKAHEAD
|
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const uint16_t inPadLength = uint16_t ((((frameLength << 1) - startLength) * resampRatio) >> resampShift) +
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(enableSbrCoding ? firstLength - (sbrEncDelay >> resampShift) - (enableUpsampler ? 32 : 0) : 0);
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# endif
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#endif
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const int64_t expectLength = (wavReader.getDataBytesLeft () << resampShift) / int64_t ((numChannels * inSampDepth * resampRatio) >> 3);
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||||
if (enableUpsampler) // notify by printf
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{
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||||
# if ENABLE_STDOUT_LOAS
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#if ENABLE_STDOUT_LOAS
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if (writeStdout) // relocate to stderr
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||||
{
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# ifdef EXHALE_APP_WCHAR
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# ifdef EXHALE_APP_WCHAR
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fwprintf_s (stderr, L" NOTE: Upsampling the input audio from %d kHz to %d kHz with preset mode %d\n\n",
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# else
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# else
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fprintf_s (stderr, " NOTE: Upsampling the input audio from %d kHz to %d kHz with preset mode %d\n\n",
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||||
# 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)
|
||||
{
|
||||
|
||||
@ -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
|
||||
|
||||
@ -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"
|
||||
|
||||
@ -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_
|
||||
|
||||
@ -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"
|
||||
|
||||
@ -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_
|
||||
|
||||
@ -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
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user