Switch out minimp3 with the dr_mp3 implementation

This commit is contained in:
Lukas Dürrenberger 2024-05-13 00:16:24 +02:00 committed by Lukas Dürrenberger
parent e39f48742b
commit 5646e05e8a
9 changed files with 4869 additions and 3333 deletions

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@ -180,13 +180,13 @@ if(SFML_USE_SYSTEM_DEPS)
foreach(DEP_FILE ${DEP_LIBS} ${DEP_BINS} ${DEP_HEADERS})
get_filename_component(DEP_DIR ${DEP_FILE} PATH)
if(NOT DEP_DIR MATCHES "/(stb_image|minimp3)(/|$)")
if(NOT DEP_DIR MATCHES "/(stb_image|dr_mp3)(/|$)")
list(APPEND CMAKE_IGNORE_PATH "${DEP_DIR}")
endif()
get_filename_component(DEP_PARENT_DIR ${DEP_DIR} PATH)
while(NOT DEP_PARENT_DIR STREQUAL "${CMAKE_SOURCE_DIR}/extlibs")
if(NOT DEP_DIR MATCHES "/(stb_image|minimp3)(/|$)")
if(NOT DEP_DIR MATCHES "/(stb_image|dr_mp3)(/|$)")
list(APPEND CMAKE_IGNORE_PATH "${DEP_PARENT_DIR}")
endif()

4837
extlibs/headers/dr_mp3/dr_mp3.h vendored Normal file

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@ -66,11 +66,6 @@ public:
/// The supported audio formats are: WAV (PCM only), OGG/Vorbis, FLAC, MP3.
/// The supported sample sizes for FLAC and WAV are 8, 16, 24 and 32 bit.
///
/// Because of minimp3_ex limitation, for MP3 files with big (>16kb) APEv2 tag,
/// it may not be properly removed, tag data will be treated as MP3 data
/// and there is a low chance of garbage decoded at the end of file.
/// See also: https://github.com/lieff/minimp3
///
/// \param filename Path of the sound file to load
///
/// \return True if the file was successfully opened

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@ -18,4 +18,4 @@ Permission is granted to anyone to use this software for any purpose, including
* _libogg_ is under the BSD license
* _libvorbis_ is under the BSD license
* _libflac_ is under the BSD license
* _minimp3_ is under the CC0 license
* _dr\_mp3_ is public domain

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@ -83,8 +83,8 @@ sfml_add_library(sfml-audio
# setup dependencies
target_link_libraries(sfml-audio PRIVATE OpenAL)
# minimp3 sources
target_include_directories(sfml-audio SYSTEM PRIVATE "${PROJECT_SOURCE_DIR}/extlibs/headers/minimp3")
# dr_mp3 sources
target_include_directories(sfml-audio SYSTEM PRIVATE "${PROJECT_SOURCE_DIR}/extlibs/headers/dr_mp3")
if(SFML_OS_ANDROID)
target_link_libraries(sfml-audio PRIVATE android OpenSLES)

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@ -25,30 +25,10 @@
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#define MINIMP3_IMPLEMENTATION // Minimp3 control define, places implementation in this file.
#ifndef NOMINMAX
#define NOMINMAX // To avoid windows.h and std::min issue
#endif
#define MINIMP3_NO_STDIO // Minimp3 control define, eliminate file manipulation code which is useless here
#ifdef _MSC_VER
#pragma warning(push)
#pragma warning(disable : 4242 4244 4267 4456 4706)
#else
#pragma GCC diagnostic push
#pragma GCC diagnostic ignored "-Wstringop-overflow"
#endif
#include <minimp3_ex.h>
#ifdef _MSC_VER
#pragma warning(pop)
#else
#pragma GCC diagnostic pop
#endif
#undef NOMINMAX
#undef MINIMP3_NO_STDIO
#define DR_MP3_IMPLEMENTATION
#define DR_MP3_NO_STDIO
#include <dr_mp3.h>
#undef DR_MP3_NO_STDIO
#include <SFML/Audio/SoundFileReaderMp3.hpp>
#include <SFML/System/InputStream.hpp>
@ -60,17 +40,17 @@
namespace
{
std::size_t readCallback(void* ptr, std::size_t size, void* data)
std::size_t readCallback(void* userData, void* bufferOut, std::size_t bytesToRead)
{
sf::InputStream* stream = static_cast<sf::InputStream*>(data);
return static_cast<std::size_t>(stream->read(ptr, static_cast<sf::Int64>(size)));
sf::InputStream* stream = static_cast<sf::InputStream*>(userData);
return static_cast<std::size_t>(stream->read(bufferOut, static_cast<sf::Int64>(bytesToRead)));
}
int seekCallback(uint64_t offset, void* data)
drmp3_bool32 seekCallback(void* userData, int offset, drmp3_seek_origin /* origin */)
{
sf::InputStream* stream = static_cast<sf::InputStream*>(data);
sf::Int64 position = stream->seek(static_cast<sf::Int64>(offset));
return position < 0 ? -1 : 0;
sf::InputStream* stream = static_cast<sf::InputStream*>(userData);
const sf::Int64 position = stream->seek(static_cast<sf::Int64>(offset));
return position >= 0;
}
bool hasValidId3Tag(const sf::Uint8* header)
@ -94,7 +74,7 @@ bool SoundFileReaderMp3::check(InputStream& stream)
if (hasValidId3Tag(header))
return true;
if (hdr_valid(header))
if (drmp3_hdr_valid(header))
return true;
return false;
@ -106,36 +86,27 @@ SoundFileReaderMp3::SoundFileReaderMp3() :
m_numSamples(0),
m_position(0)
{
std::memset(&m_io, 0, sizeof(m_io));
std::memset(&m_decoder, 0, sizeof(m_decoder));
m_io.read = readCallback;
m_io.seek = seekCallback;
}
////////////////////////////////////////////////////////////
SoundFileReaderMp3::~SoundFileReaderMp3()
{
mp3dec_ex_close(&m_decoder);
drmp3_uninit(&m_decoder);
}
////////////////////////////////////////////////////////////
bool SoundFileReaderMp3::open(InputStream& stream, Info& info)
{
// Init IO callbacks
m_io.read_data = &stream;
m_io.seek_data = &stream;
// Init mp3 decoder
mp3dec_ex_open_cb(&m_decoder, &m_io, MP3D_SEEK_TO_SAMPLE);
if (!m_decoder.samples)
if (drmp3_init(&m_decoder, readCallback, seekCallback, &stream, NULL) != 1)
return false;
// Retrieve the music attributes
info.channelCount = static_cast<unsigned int>(m_decoder.info.channels);
info.sampleRate = static_cast<unsigned int>(m_decoder.info.hz);
info.sampleCount = m_decoder.samples;
info.channelCount = static_cast<unsigned int>(m_decoder.channels);
info.sampleRate = static_cast<unsigned int>(m_decoder.sampleRate);
info.sampleCount = drmp3_get_pcm_frame_count(&m_decoder) * m_decoder.channels;
m_numSamples = info.sampleCount;
return true;
@ -146,15 +117,16 @@ bool SoundFileReaderMp3::open(InputStream& stream, Info& info)
void SoundFileReaderMp3::seek(Uint64 sampleOffset)
{
m_position = std::min(sampleOffset, m_numSamples);
mp3dec_ex_seek(&m_decoder, m_position);
drmp3_seek_to_pcm_frame(&m_decoder, m_position);
}
////////////////////////////////////////////////////////////
Uint64 SoundFileReaderMp3::read(Int16* samples, Uint64 maxCount)
{
Uint64 toRead = std::min(maxCount, m_numSamples - m_position);
toRead = static_cast<Uint64>(mp3dec_ex_read(&m_decoder, samples, static_cast<std::size_t>(toRead)));
// dr_mp3's PCM frame contains one sample per channel
Uint64 toRead = std::min(maxCount, m_numSamples - m_position) / m_decoder.channels;
toRead = static_cast<Uint64>(drmp3_read_pcm_frames_s16(&m_decoder, toRead, samples)) * m_decoder.channels;
m_position += toRead;
return toRead;
}

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@ -28,24 +28,9 @@
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#ifndef NOMINMAX
#define NOMINMAX // To avoid windows.h and std::min issue
#endif
#define MINIMP3_NO_STDIO // Minimp3 control define, eliminate file manipulation code which is useless here
#ifdef _MSC_VER
#pragma warning(push)
#pragma warning(disable : 4242 4244 4267 4456 4706)
#endif
#include <minimp3_ex.h>
#ifdef _MSC_VER
#pragma warning(pop)
#endif
#undef NOMINMAX
#undef MINIMP3_NO_STDIO
#define DR_MP3_NO_STDIO
#include <dr_mp3.h>
#undef DR_MP3_NO_STDIO
#include <SFML/Audio/SoundFileReader.hpp>
#include <vector>
@ -129,10 +114,9 @@ private:
////////////////////////////////////////////////////////////
// Member data
////////////////////////////////////////////////////////////
mp3dec_io_t m_io;
mp3dec_ex_t m_decoder;
Uint64 m_numSamples; // Decompressed audio storage size
Uint64 m_position; // Position in decompressed audio buffer
drmp3 m_decoder;
Uint64 m_numSamples; // Decompressed audio storage size
Uint64 m_position; // Position in decompressed audio buffer
};
} // namespace priv