Move revision history out of miniaudio.h and into CHANGES.md

cc274c67 · David Reid · a548ffba · cc274c67 · cc274c67
Commit cc274c67 authored Jan 11, 2022 by David Reid
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+v0.11.4 - 2022-01-12
+  - AAudio: Add initial support for automatic stream routing.
+  - Add support for initializing an encoder from a VFS.
+  - Fix a bug when initializing an encoder from a file where the file handle does not get closed in
+    the event of an error.
+v0.11.3 - 2022-01-07
+  - Add a new flag for nodes called MA_NODE_FLAG_SILENT_OUTPUT which tells miniaudio that the
+    output of the node should always be treated as silence. This gives miniaudio an optimization
+    opportunity by skipping mixing of those nodes. Useful for special nodes that need to have
+    their outputs wired up to the graph so they're processed, but don't want the output to
+    contribute to the final mix.
+  - Add support for fixed sized callbacks. With this change, the data callback will be fired with
+    a consistent frame count based on the periodSizeInFrames or periodSizeInMilliseconds config
+    variable (depending on which one is used). If the period size is not specified, the backend's
+    internal period size will be used. Under the hood this uses an intermediary buffer which
+    introduces a small inefficiency. To avoid this you can use the `noFixedSizedCallback` config
+    variable and set it to true. This will make the callback equivalent to the way it was before
+    this change and will avoid the intermediary buffer, but the data callback could get fired with
+    an inconsistent frame count which might cause problems where certain operations need to operate
+    on fixed sized chunks.
+  - Change the logging system to always process debug log messages. This is useful for allowing
+    debug and test builds of applications to output debug information that can later be passed on
+    for debugging in miniaudio. To filter out these messages, just filter against the log level
+    which will be MA_LOG_LEVEL_DEBUG.
+  - Change the wav decoder to pick the closest format to the source file by default if no preferred
+    format is specified.
+  - Fix a bug where ma_device_get_info() and ma_device_get_name() return an error.
+  - Fix a bug where ma_data_source_read_pcm_frames() can return MA_AT_END even when some data has
+    been read. MA_AT_END should only be returned when nothing has been read.
+  - PulseAudio: Fix some bugs where starting and stopping a device can result in a deadlock.
+v0.11.2 - 2021-12-31
+  - Add a new device notification system to replace the stop callback. The stop callback is still
+    in place, but will be removed in version 0.12. New code should use the notificationCallback
+    member in the device config instead of stopCallback.
+  - Fix a bug where the stopped notification doesn't get fired.
+  - iOS: The IO buffer size is now configured based on the device's configured period size.
+  - WebAudio: Optimizations to some JavaScript code.
+v0.11.1 - 2021-12-27
+  - Result codes are now declared as an enum rather than #defines.
+  - Channel positions (MA_CHANNEL_*) are now declared as an enum rather than #defines.
+  - Add ma_device_get_info() for retrieving device information from an initialized device.
+  - Add ma_device_get_name() for retrieving the name of an initialized device.
+  - Add support for setting the directional attenuation factor to sounds and groups.
+  - Fix a crash when passing in NULL for the pEngine parameter of ma_engine_init().
+  - Fix a bug where the node graph will output silence if a node has zero input connections.
+  - Fix a bug in the engine where sounds in front of the listener are too loud.
+  - AAudio: Fix an incorrect assert.
+  - AAudio: Fix a bug that resulted in exclusive mode always resulting in initialization failure.
+  - AAudio: Fix a bug that resulted in a capture device incorrectly being detected as disconnected.
+  - OpenSL: Fix an error when initializing a device with a non-NULL device ID.
+  - OpenSL: Fix some bugs with device initialization.
+v0.11.0 - 2021-12-18
+  - Add a node graph system for advanced mixing and effect processing.
+  - Add a resource manager for loading and streaming sounds.
+  - Add a high level engine API for sound management and mixing. This wraps around the node graph
+    and resource manager.
+  - Add support for custom resmplers.
+  - Add ma_decoder_get_data_format().
+  - Add support for disabling denormals on the audio thread.
+  - Add a delay/echo effect called ma_delay.
+  - Add a stereo pan effect called ma_panner.
+  - Add a spataializer effect called ma_spatializer.
+  - Add support for amplification for device master volume.
+  - Remove dependency on MA_MAX_CHANNELS from filters and data conversion.
+  - Increase MA_MAX_CHANNELS from 32 to 254.
+  - API CHANGE: Changes have been made to the way custom data sources are made. See documentation
+    on how to implement custom data sources.
+  - API CHANGE: Remove ma_data_source_map() and ma_data_source_unmap()
+  - API CHANGE: Remove the `loop` parameter from ma_data_source_read_pcm_frames(). Use
+    ma_data_source_set_looping() to enable or disable looping.
+  - API CHANGE: Remove ma_channel_mix_mode_planar_blend. Use ma_channel_mix_mode_rectangular instead.
+  - API CHANGE: Remove MA_MIN_SAMPLE_RATE and MA_MAX_SAMPLE_RATE. Use ma_standard_sample_rate_min
+    and ma_standard_sample_rate_max instead.
+  - API CHANGE: Changes have been made to the ma_device_info structure. See documentation for
+    details of these changes.
+  - API CHANGE: Remove the `shareMode` parameter from ma_context_get_device_info().
+  - API CHANGE: Rename noPreZeroedOutputBuffer to noPreSilencedOutputBuffer in the device config.
+  - API CHANGE: Remove pBufferOut parameter from ring buffer commit functions.
+  - API CHANGE: Remove ma_zero_pcm_frames(). Use ma_silence_pcm_frames() instead.
+  - API CHANGE: Change ma_clip_samples_f32() to take input and output buffers rather than working
+    exclusively in-place.
+  - API CHANGE: Remove ma_clip_pcm_frames_f32(). Use ma_clip_samples_f32() or ma_clip_pcm_frames()
+    instead.
+  - API CHANGE: Remove the onLog callback from the context config and replaced with a more
+    flexible system. See the documentation for how to use logging.
+  - API CHANGE: Remove MA_LOG_LEVEL_VERBOSE and add MA_LOG_LEVEL_DEBUG. Logs using the
+    MA_LOG_LEVEL_DEBUG logging level will only be output when miniaudio is compiled with the
+    MA_DEBUG_OUTPUT option.
+  - API CHANGE: MA_LOG_LEVEL has been removed. All log levels will be posted, except for
+    MA_LOG_LEVEL_DEBUG which will only be output when MA_DEBUG_OUTPUT is enabled.
+  - API CHANGE: Rename ma_resource_format to ma_encoding_format.
+  - API CHANGE: Remove all encoding-specific initialization routines for decoders. Use the
+    encodingFormat properties in the decoder config instead.
+  - API CHANGE: Change ma_decoder_get_length_in_pcm_frames() to return a result code and output the
+    number of frames read via an output parameter.
+  - API CHANGE: Allocation callbacks must now implement the onRealloc() callback.
+  - API CHANGE: Remove ma_get_standard_channel_map() and add ma_channel_map_init_standard().
+  - API CHANGE: Rename ma_channel_map_valid() to ma_channel_map_is_valid().
+  - API CHANGE: Rename ma_channel_map_equal() to ma_channel_map_is_equal().
+  - API CHANGE: Rename ma_channel_map_blank() to ma_channel_map_is_blank().
+  - API CHANGE: Remove the Speex resampler. Use a custom resampler instead.
+  - API CHANGE: Change the following resampler APIs to return a result code and output their result
+    via an output parameter:
+    - ma_linear_resampler_get_required_input_frame_count()
+    - ma_linear_resampler_get_expected_output_frame_count()
+    - ma_resampler_get_required_input_frame_count()
+    - ma_resampler_get_expected_output_frame_count()
+  - API CHANGE: Update relevant init/uninit functions to take a pointer to allocation callbacks.
+  - API CHANGE: Remove ma_scale_buffer_size()
+  - API CHANGE: Update ma_encoder_write_pcm_frames() to return a result code and output the number
+    of frames written via an output parameter.
+  - API CHANGE: Update ma_noise_read_pcm_frames() to return a result code and output the number of
+    frames read via an output parameter.
+  - API CHANGE: Update ma_waveform_read_pcm_frames() to return a result code and output the number
+    of frames read via an output parameter.
+  - API CHANGE: Remove The MA_STATE_* and add ma_device_state_* enums.
+  - API CHANGE: Rename ma_factor_to_gain_db() to ma_volume_linear_to_db().
+  - API CHANGE: Rename ma_gain_db_to_factor() to ma_volume_db_to_linear().
+  - API CHANGE: Rename ma_device_set_master_gain_db() to ma_device_set_master_volume_db().
+  - API CHANGE: Rename ma_device_get_master_gain_db() to ma_device_get_master_volume_db()
+v0.10.43 - 2021-12-10
+  - ALSA: Fix use of uninitialized variables.
+  - ALSA: Fix enumeration of devices that support both playback and capture.
+  - PulseAudio: Fix a possible division by zero.
+  - WebAudio: Fix errors in strict mode.
+v0.10.42 - 2021-08-22
+  - Fix a possible deadlock when stopping devices.
+v0.10.41 - 2021-08-15
+  - Core Audio: Fix some deadlock errors.
+v0.10.40 - 2021-07-23
+  - Fix a bug when converting from stereo to mono.
+  - PulseAudio: Fix a glitch when pausing and resuming a device.
+v0.10.39 - 2021-07-20
+  - Core Audio: Fix a deadlock when the default device is changed.
+  - Core Audio: Fix compilation errors on macOS and iOS.
+  - PulseAudio: Fix a bug where the stop callback is not fired when a device is unplugged.
+  - PulseAudio: Fix a null pointer dereference.
+v0.10.38 - 2021-07-14
+  - Fix a linking error when MA_DEBUG_OUTPUT is not enabled.
+  - Fix an error where ma_log_postv() does not return a value.
+  - OpenSL: Fix a bug with setting of stream types and recording presets.
+0.10.37 - 2021-07-06
+  - Fix a bug with log message formatting.
+  - Fix build when compiling with MA_NO_THREADING.
+  - Minor updates to channel mapping.
+0.10.36 - 2021-07-03
+  - Add support for custom decoding backends.
+  - Fix some bugs with the Vorbis decoder.
+  - PulseAudio: Fix a bug with channel mapping.
+  - PulseAudio: Fix a bug where miniaudio does not fall back to a supported format when PulseAudio
+    defaults to a format not known to miniaudio.
+  - OpenSL: Fix a crash when initializing a capture device when a recording preset other than the
+    default is specified.
+  - Silence some warnings when compiling with MA_DEBUG_OUTPUT
+  - Improvements to logging. See the `ma_log` API for details. The logCallback variable used by
+    ma_context has been deprecated and will be replaced with the new system in version 0.11.
+    - Initialize an `ma_log` object with `ma_log_init()`.
+    - Register a callback with `ma_log_register_callback()`.
+    - In the context config, set `pLog` to your `ma_log` object and stop using `logCallback`.
+  - Prep work for some upcoming changes to data sources. These changes are still compatible with
+    existing code, however code will need to be updated in preparation for version 0.11 which will
+    be breaking. You should make these changes now for any custom data sources:
+    - Change your base data source object from `ma_data_source_callbacks` to `ma_data_source_base`.
+    - Call `ma_data_source_init()` for your base object in your custom data source's initialization
+      routine. This takes a config object which includes a pointer to a vtable which is now where
+      your custom callbacks are defined.
+    - Call `ma_data_source_uninit()` in your custom data source's uninitialization routine. This
+      doesn't currently do anything, but it placeholder in case some future uninitialization code
+      is required to be added at a later date.
+v0.10.35 - 2021-04-27
+  - Fix the C++ build.
+v0.10.34 - 2021-04-26
+  - WASAPI: Fix a bug where a result code is not getting checked at initialization time.
+  - WASAPI: Bug fixes for loopback mode.
+  - ALSA: Fix a possible deadlock when stopping devices.
+  - Mark devices as default on the null backend.
+v0.10.33 - 2021-04-04
+  - Core Audio: Fix a memory leak.
+  - Core Audio: Fix a bug where the performance profile is not being used by playback devices.
+  - JACK: Fix loading of 64-bit JACK on Windows.
+  - Fix a calculation error and add a safety check to the following APIs to prevent a division by zero:
+    - ma_calculate_buffer_size_in_milliseconds_from_frames()
+    - ma_calculate_buffer_size_in_frames_from_milliseconds()
+  - Fix compilation errors relating to c89atomic.
+  - Update FLAC decoder.
+v0.10.32 - 2021-02-23
+  - WASAPI: Fix a deadlock in exclusive mode.
+  - WASAPI: No longer return an error from ma_context_get_device_info() when an exclusive mode format
+    cannot be retrieved.
+  - WASAPI: Attempt to fix some bugs with device uninitialization.
+  - PulseAudio: Yet another refactor, this time to remove the dependency on `pa_threaded_mainloop`.
+  - Web Audio: Fix a bug on Chrome and any other browser using the same engine.
+  - Web Audio: Automatically start the device on some user input if the device has been started. This
+    is to work around Google's policy of not starting audio if no user input has yet been performed.
+  - Fix a bug where thread handles are not being freed.
+  - Fix some static analysis warnings in FLAC, WAV and MP3 decoders.
+  - Fix a warning due to referencing _MSC_VER when it is undefined.
+  - Update to latest version of c89atomic.
+  - Internal refactoring to migrate over to the new backend callback system for the following backends:
+    - PulseAudio
+    - ALSA
+    - Core Audio
+    - AAudio
+    - OpenSL|ES
+    - OSS
+    - audio(4)
+    - sndio
+v0.10.31 - 2021-01-17
+  - Make some functions const correct.
+  - Update ma_data_source_read_pcm_frames() to initialize pFramesRead to 0 for safety.
+  - Add the MA_ATOMIC annotation for use with variables that should be used atomically and remove unnecessary volatile qualifiers.
+  - Add support for enabling only specific backends at compile time. This is the reverse of the pre-existing system. With the new
+    system, all backends are first disabled with `MA_ENABLE_ONLY_SPECIFIC_BACKENDS`, which is then followed with `MA_ENABLE_*`. The
+    old system where you disable backends with `MA_NO_*` still exists and is still the default.
+v0.10.30 - 2021-01-10
+  - Fix a crash in ma_audio_buffer_read_pcm_frames().
+  - Update spinlock APIs to take a volatile parameter as input.
+  - Silence some unused parameter warnings.
+  - Fix a warning on GCC when compiling as C++.
+v0.10.29 - 2020-12-26
+  - Fix some subtle multi-threading bugs on non-x86 platforms.
+  - Fix a bug resulting in superfluous memory allocations when enumerating devices.
+  - Core Audio: Fix a compilation error when compiling for iOS.
+v0.10.28 - 2020-12-16
+  - Fix a crash when initializing a POSIX thread.
+  - OpenSL|ES: Respect the MA_NO_RUNTIME_LINKING option.
+v0.10.27 - 2020-12-04
+  - Add support for dynamically configuring some properties of `ma_noise` objects post-initialization.
+  - Add support for configuring the channel mixing mode in the device config.
+  - Fix a bug with simple channel mixing mode (drop or silence excess channels).
+  - Fix some bugs with trying to access uninitialized variables.
+  - Fix some errors with stopping devices for synchronous backends where the backend's stop callback would get fired twice.
+  - Fix a bug in the decoder due to using an uninitialized variable.
+  - Fix some data race errors.
+v0.10.26 - 2020-11-24
+  - WASAPI: Fix a bug where the exclusive mode format may not be retrieved correctly due to accessing freed memory.
+  - Fix a bug with ma_waveform where glitching occurs after changing frequency.
+  - Fix compilation with OpenWatcom.
+  - Fix compilation with TCC.
+  - Fix compilation with Digital Mars.
+  - Fix compilation warnings.
+  - Remove bitfields from public structures to aid in binding maintenance.
+v0.10.25 - 2020-11-15
+  - PulseAudio: Fix a bug where the stop callback isn't fired.
+  - WebAudio: Fix an error that occurs when Emscripten increases the size of it's heap.
+  - Custom Backends: Change the onContextInit and onDeviceInit callbacks to take a parameter which is a pointer to the config that was
+    passed into ma_context_init() and ma_device_init(). This replaces the deviceType parameter of onDeviceInit.
+  - Fix compilation warnings on older versions of GCC.
+v0.10.24 - 2020-11-10
+  - Fix a bug where initialization of a backend can fail due to some bad state being set from a prior failed attempt at initializing a
+    lower priority backend.
+v0.10.23 - 2020-11-09
+  - AAudio: Add support for configuring a playback stream's usage.
+  - Fix a compilation error when all built-in asynchronous backends are disabled at compile time.
+  - Fix compilation errors when compiling as C++.
+v0.10.22 - 2020-11-08
+  - Add support for custom backends.
+  - Add support for detecting default devices during device enumeration and with `ma_context_get_device_info()`.
+  - Refactor to the PulseAudio backend. This simplifies the implementation and fixes a capture bug.
+  - ALSA: Fix a bug in `ma_context_get_device_info()` where the PCM handle is left open in the event of an error.
+  - Core Audio: Further improvements to sample rate selection.
+  - Core Audio: Fix some bugs with capture mode.
+  - OpenSL: Add support for configuring stream types and recording presets.
+  - AAudio: Add support for configuring content types and input presets.
+  - Fix bugs in `ma_decoder_init_file*()` where the file handle is not closed after a decoding error.
+  - Fix some compilation warnings on GCC and Clang relating to the Speex resampler.
+  - Fix a compilation error for the Linux build when the ALSA and JACK backends are both disabled.
+  - Fix a compilation error for the BSD build.
+  - Fix some compilation errors on older versions of GCC.
+  - Add documentation for `MA_NO_RUNTIME_LINKING`.
+v0.10.21 - 2020-10-30
+  - Add ma_is_backend_enabled() and ma_get_enabled_backends() for retrieving enabled backends at run-time.
+  - WASAPI: Fix a copy and paste bug relating to loopback mode.
+  - Core Audio: Fix a bug when using multiple contexts.
+  - Core Audio: Fix a compilation warning.
+  - Core Audio: Improvements to sample rate selection.
+  - Core Audio: Improvements to format/channels/rate selection when requesting defaults.
+  - Core Audio: Add notes regarding the Apple notarization process.
+  - Fix some bugs due to null pointer dereferences.
+v0.10.20 - 2020-10-06
+  - Fix build errors with UWP.
+  - Minor documentation updates.
+v0.10.19 - 2020-09-22
+  - WASAPI: Return an error when exclusive mode is requested, but the native format is not supported by miniaudio.
+  - Fix a bug where ma_decoder_seek_to_pcm_frames() never returns MA_SUCCESS even though it was successful.
+  - Store the sample rate in the `ma_lpf` and `ma_hpf` structures.
+v0.10.18 - 2020-08-30
+  - Fix build errors with VC6.
+  - Fix a bug in channel converter for s32 format.
+  - Change channel converter configs to use the default channel map instead of a blank channel map when no channel map is specified when initializing the
+    config. This fixes an issue where the optimized mono expansion path would never get used.
+  - Use a more appropriate default format for FLAC decoders. This will now use ma_format_s16 when the FLAC is encoded as 16-bit.
+  - Update FLAC decoder.
+  - Update links to point to the new repository location (https://github.com/mackron/miniaudio).
+v0.10.17 - 2020-08-28
+  - Fix an error where the WAV codec is incorrectly excluded from the build depending on which compile time options are set.
+  - Fix a bug in ma_audio_buffer_read_pcm_frames() where it isn't returning the correct number of frames processed.
+  - Fix compilation error on Android.
+  - Core Audio: Fix a bug with full-duplex mode.
+  - Add ma_decoder_get_cursor_in_pcm_frames().
+  - Update WAV codec.
+v0.10.16 - 2020-08-14
+  - WASAPI: Fix a potential crash due to using an uninitialized variable.
+  - OpenSL: Enable runtime linking.
+  - OpenSL: Fix a multithreading bug when initializing and uninitializing multiple contexts at the same time.
+  - iOS: Improvements to device enumeration.
+  - Fix a crash in ma_data_source_read_pcm_frames() when the output frame count parameter is NULL.
+  - Fix a bug in ma_data_source_read_pcm_frames() where looping doesn't work.
+  - Fix some compilation warnings on Windows when both DirectSound and WinMM are disabled.
+  - Fix some compilation warnings when no decoders are enabled.
+  - Add ma_audio_buffer_get_available_frames().
+  - Add ma_decoder_get_available_frames().
+  - Add sample rate to ma_data_source_get_data_format().
+  - Change volume APIs to take 64-bit frame counts.
+  - Updates to documentation.
+v0.10.15 - 2020-07-15
+  - Fix a bug when converting bit-masked channel maps to miniaudio channel maps. This affects the WASAPI and OpenSL backends.
+v0.10.14 - 2020-07-14
+  - Fix compilation errors on Android.
+  - Fix compilation errors with -march=armv6.
+  - Updates to the documentation.
+v0.10.13 - 2020-07-11
+  - Fix some potential buffer overflow errors with channel maps when channel counts are greater than MA_MAX_CHANNELS.
+  - Fix compilation error on Emscripten.
+  - Silence some unused function warnings.
+  - Increase the default buffer size on the Web Audio backend. This fixes glitching issues on some browsers.
+  - Bring FLAC decoder up-to-date with dr_flac.
+  - Bring MP3 decoder up-to-date with dr_mp3.
+v0.10.12 - 2020-07-04
+  - Fix compilation errors on the iOS build.
+v0.10.11 - 2020-06-28
+  - Fix some bugs with device tracking on Core Audio.
+  - Updates to documentation.
+v0.10.10 - 2020-06-26
+  - Add include guard for the implementation section.
+  - Mark ma_device_sink_info_callback() as static.
+  - Fix compilation errors with MA_NO_DECODING and MA_NO_ENCODING.
+  - Fix compilation errors with MA_NO_DEVICE_IO
+v0.10.9 - 2020-06-24
+  - Amalgamation of dr_wav, dr_flac and dr_mp3. With this change, including the header section of these libraries before the implementation of miniaudio is no
+    longer required. Decoding of WAV, FLAC and MP3 should be supported seamlessly without any additional libraries. Decoders can be excluded from the build
+    with the following options:
+    - MA_NO_WAV
+    - MA_NO_FLAC
+    - MA_NO_MP3
+    If you get errors about multiple definitions you need to either enable the options above, move the implementation of dr_wav, dr_flac and/or dr_mp3 to before
+    the implementation of miniaudio, or update dr_wav, dr_flac and/or dr_mp3.
+  - Changes to the internal atomics library. This has been replaced with c89atomic.h which is embedded within this file.
+  - Fix a bug when a decoding backend reports configurations outside the limits of miniaudio's decoder abstraction.
+  - Fix the UWP build.
+  - Fix the Core Audio build.
+  - Fix the -std=c89 build on GCC.
+v0.10.8 - 2020-06-22
+  - Remove dependency on ma_context from mutexes.
+  - Change ma_data_source_read_pcm_frames() to return a result code and output the frames read as an output parameter.
+  - Change ma_data_source_seek_pcm_frames() to return a result code and output the frames seeked as an output parameter.
+  - Change ma_audio_buffer_unmap() to return MA_AT_END when the end has been reached. This should be considered successful.
+  - Change playback.pDeviceID and capture.pDeviceID to constant pointers in ma_device_config.
+  - Add support for initializing decoders from a virtual file system object. This is achieved via the ma_vfs API and allows the application to customize file
+    IO for the loading and reading of raw audio data. Passing in NULL for the VFS will use defaults. New APIs:
+    - ma_decoder_init_vfs()
+    - ma_decoder_init_vfs_wav()
+    - ma_decoder_init_vfs_flac()
+    - ma_decoder_init_vfs_mp3()
+    - ma_decoder_init_vfs_vorbis()
+    - ma_decoder_init_vfs_w()
+    - ma_decoder_init_vfs_wav_w()
+    - ma_decoder_init_vfs_flac_w()
+    - ma_decoder_init_vfs_mp3_w()
+    - ma_decoder_init_vfs_vorbis_w()
+  - Add support for memory mapping to ma_data_source.
+    - ma_data_source_map()
+    - ma_data_source_unmap()
+  - Add ma_offset_pcm_frames_ptr() and ma_offset_pcm_frames_const_ptr() which can be used for offsetting a pointer by a specified number of PCM frames.
+  - Add initial implementation of ma_yield() which is useful for spin locks which will be used in some upcoming work.
+  - Add documentation for log levels.
+  - The ma_event API has been made public in preparation for some uncoming work.
+  - Fix a bug in ma_decoder_seek_to_pcm_frame() where the internal sample rate is not being taken into account for determining the seek location.
+  - Fix some bugs with the linear resampler when dynamically changing the sample rate.
+  - Fix compilation errors with MA_NO_DEVICE_IO.
+  - Fix some warnings with GCC and -std=c89.
+  - Fix some formatting warnings with GCC and -Wall and -Wpedantic.
+  - Fix some warnings with VC6.
+  - Minor optimization to ma_copy_pcm_frames(). This is now a no-op when the input and output buffers are the same.
+v0.10.7 - 2020-05-25
+  - Fix a compilation error in the C++ build.
+  - Silence a warning.
+v0.10.6 - 2020-05-24
+  - Change ma_clip_samples_f32() and ma_clip_pcm_frames_f32() to take a 64-bit sample/frame count.
+  - Change ma_zero_pcm_frames() to clear to 128 for ma_format_u8.
+  - Add ma_silence_pcm_frames() which replaces ma_zero_pcm_frames(). ma_zero_pcm_frames() will be removed in version 0.11.
+  - Add support for u8, s24 and s32 formats to ma_channel_converter.
+  - Add compile-time and run-time version querying.
+    - MA_VERSION_MINOR
+    - MA_VERSION_MAJOR
+    - MA_VERSION_REVISION
+    - MA_VERSION_STRING
+    - ma_version()
+    - ma_version_string()
+  - Add ma_audio_buffer for reading raw audio data directly from memory.
+  - Fix a bug in shuffle mode in ma_channel_converter.
+  - Fix compilation errors in certain configurations for ALSA and PulseAudio.
+  - The data callback now initializes the output buffer to 128 when the playback sample format is ma_format_u8.
+v0.10.5 - 2020-05-05
+  - Change ma_zero_pcm_frames() to take a 64-bit frame count.
+  - Add ma_copy_pcm_frames().
+  - Add MA_NO_GENERATION build option to exclude the `ma_waveform` and `ma_noise` APIs from the build.
+  - Add support for formatted logging to the VC6 build.
+  - Fix a crash in the linear resampler when LPF order is 0.
+  - Fix compilation errors and warnings with older versions of Visual Studio.
+  - Minor documentation updates.
+v0.10.4 - 2020-04-12
+  - Fix a data conversion bug when converting from the client format to the native device format.
+v0.10.3 - 2020-04-07
+  - Bring up to date with breaking changes to dr_mp3.
+  - Remove MA_NO_STDIO. This was causing compilation errors and the maintenance cost versus practical benefit is no longer worthwhile.
+  - Fix a bug with data conversion where it was unnecessarily converting to s16 or f32 and then straight back to the original format.
+  - Fix compilation errors and warnings with Visual Studio 2005.
+  - ALSA: Disable ALSA's automatic data conversion by default and add configuration options to the device config:
+    - alsa.noAutoFormat
+    - alsa.noAutoChannels
+    - alsa.noAutoResample
+  - WASAPI: Add some overrun recovery for ma_device_type_capture devices.
+v0.10.2 - 2020-03-22
+  - Decorate some APIs with MA_API which were missed in the previous version.
+  - Fix a bug in ma_linear_resampler_set_rate() and ma_linear_resampler_set_rate_ratio().
+v0.10.1 - 2020-03-17
+  - Add MA_API decoration. This can be customized by defining it before including miniaudio.h.
+  - Fix a bug where opening a file would return a success code when in fact it failed.
+  - Fix compilation errors with Visual Studio 6 and 2003.
+  - Fix warnings on macOS.
+v0.10.0 - 2020-03-07
+  - API CHANGE: Refactor data conversion APIs
+    - ma_format_converter has been removed. Use ma_convert_pcm_frames_format() instead.
+    - ma_channel_router has been replaced with ma_channel_converter.
+    - ma_src has been replaced with ma_resampler
+    - ma_pcm_converter has been replaced with ma_data_converter
+  - API CHANGE: Add support for custom memory allocation callbacks. The following APIs have been updated to take an extra parameter for the allocation
+    callbacks:
+    - ma_malloc()
+    - ma_realloc()
+    - ma_free()
+    - ma_aligned_malloc()
+    - ma_aligned_free()
+    - ma_rb_init() / ma_rb_init_ex()
+    - ma_pcm_rb_init() / ma_pcm_rb_init_ex()
+  - API CHANGE: Simplify latency specification in device configurations. The bufferSizeInFrames and bufferSizeInMilliseconds parameters have been replaced with
+    periodSizeInFrames and periodSizeInMilliseconds respectively. The previous variables defined the size of the entire buffer, whereas the new ones define the
+    size of a period. The following APIs have been removed since they are no longer relevant:
+    - ma_get_default_buffer_size_in_milliseconds()
+    - ma_get_default_buffer_size_in_frames()
+  - API CHANGE: ma_device_set_stop_callback() has been removed. If you require a stop callback, you must now set it via the device config just like the data
+    callback.
+  - API CHANGE: The ma_sine_wave API has been replaced with ma_waveform. The following APIs have been removed:
+    - ma_sine_wave_init()
+    - ma_sine_wave_read_f32()
+    - ma_sine_wave_read_f32_ex()
+  - API CHANGE: ma_convert_frames() has been updated to take an extra parameter which is the size of the output buffer in PCM frames. Parameters have also been
+    reordered.
+  - API CHANGE: ma_convert_frames_ex() has been changed to take a pointer to a ma_data_converter_config object to specify the input and output formats to
+    convert between.
+  - API CHANGE: ma_calculate_frame_count_after_src() has been renamed to ma_calculate_frame_count_after_resampling().
+  - Add support for the following filters:
+    - Biquad (ma_biquad)
+    - First order low-pass (ma_lpf1)
+    - Second order low-pass (ma_lpf2)
+    - Low-pass with configurable order (ma_lpf)
+    - First order high-pass (ma_hpf1)
+    - Second order high-pass (ma_hpf2)
+    - High-pass with configurable order (ma_hpf)
+    - Second order band-pass (ma_bpf2)
+    - Band-pass with configurable order (ma_bpf)
+    - Second order peaking EQ (ma_peak2)
+    - Second order notching (ma_notch2)
+    - Second order low shelf (ma_loshelf2)
+    - Second order high shelf (ma_hishelf2)
+  - Add waveform generation API (ma_waveform) with support for the following:
+    - Sine
+    - Square
+    - Triangle
+    - Sawtooth
+  - Add noise generation API (ma_noise) with support for the following:
+    - White
+    - Pink
+    - Brownian
+  - Add encoding API (ma_encoder). This only supports outputting to WAV files via dr_wav.
+  - Add ma_result_description() which is used to retrieve a human readable description of a given result code.
+  - Result codes have been changed. Binding maintainers will need to update their result code constants.
+  - More meaningful result codes are now returned when a file fails to open.
+  - Internal functions have all been made static where possible.
+  - Fix potential crash when ma_device object's are not aligned to MA_SIMD_ALIGNMENT.
+  - Fix a bug in ma_decoder_get_length_in_pcm_frames() where it was returning the length based on the internal sample rate rather than the output sample rate.
+  - Fix bugs in some backends where the device is not drained properly in ma_device_stop().
+  - Improvements to documentation.
+v0.9.10 - 2020-01-15
+  - Fix compilation errors due to #if/#endif mismatches.
+  - WASAPI: Fix a bug where automatic stream routing is being performed for devices that are initialized with an explicit device ID.
+  - iOS: Fix a crash on device uninitialization.
+v0.9.9 - 2020-01-09
+  - Fix compilation errors with MinGW.
+  - Fix compilation errors when compiling on Apple platforms.
+  - WASAPI: Add support for disabling hardware offloading.
+  - WASAPI: Add support for disabling automatic stream routing.
+  - Core Audio: Fix bugs in the case where the internal device uses deinterleaved buffers.
+  - Core Audio: Add support for controlling the session category (AVAudioSessionCategory) and options (AVAudioSessionCategoryOptions).
+  - JACK: Fix bug where incorrect ports are connected.
+v0.9.8 - 2019-10-07
+  - WASAPI: Fix a potential deadlock when starting a full-duplex device.
+  - WASAPI: Enable automatic resampling by default. Disable with config.wasapi.noAutoConvertSRC.
+  - Core Audio: Fix bugs with automatic stream routing.
+  - Add support for controlling whether or not the content of the output buffer passed in to the data callback is pre-initialized
+    to zero. By default it will be initialized to zero, but this can be changed by setting noPreZeroedOutputBuffer in the device
+    config. Setting noPreZeroedOutputBuffer to true will leave the contents undefined.
+  - Add support for clipping samples after the data callback has returned. This only applies when the playback sample format is
+    configured as ma_format_f32. If you are doing clipping yourself, you can disable this overhead by setting noClip to true in
+    the device config.
+  - Add support for master volume control for devices.
+    - Use ma_device_set_master_volume() to set the volume to a factor between 0 and 1, where 0 is silence and 1 is full volume.
+    - Use ma_device_set_master_volume_db() to set the volume in decibels where 0 is full volume and < 0 reduces the volume.
+  - Fix warnings emitted by GCC when `__inline__` is undefined or defined as nothing.
+v0.9.7 - 2019-08-28
+  - Add support for loopback mode (WASAPI only).
+    - To use this, set the device type to ma_device_type_loopback, and then fill out the capture section of the device config.
+    - If you need to capture from a specific output device, set the capture device ID to that of a playback device.
+  - Fix a crash when an error is posted in ma_device_init().
+  - Fix a compilation error when compiling for ARM architectures.
+  - Fix a bug with the audio(4) backend where the device is incorrectly being opened in non-blocking mode.
+  - Fix memory leaks in the Core Audio backend.
+  - Minor refactoring to the WinMM, ALSA, PulseAudio, OSS, audio(4), sndio and null backends.
+v0.9.6 - 2019-08-04
+  - Add support for loading decoders using a wchar_t string for file paths.
+  - Don't trigger an assert when ma_device_start() is called on a device that is already started. This will now log a warning
+    and return MA_INVALID_OPERATION. The same applies for ma_device_stop().
+  - Try fixing an issue with PulseAudio taking a long time to start playback.
+  - Fix a bug in ma_convert_frames() and ma_convert_frames_ex().
+  - Fix memory leaks in the WASAPI backend.
+  - Fix a compilation error with Visual Studio 2010.
+v0.9.5 - 2019-05-21
+  - Add logging to ma_dlopen() and ma_dlsym().
+  - Add ma_decoder_get_length_in_pcm_frames().
+  - Fix a bug with capture on the OpenSL|ES backend.
+  - Fix a bug with the ALSA backend where a device would not restart after being stopped.
+v0.9.4 - 2019-05-06
+  - Add support for C89. With this change, miniaudio should compile clean with GCC/Clang with "-std=c89 -ansi -pedantic" and
+    Microsoft compilers back to VC6. Other compilers should also work, but have not been tested.
+v0.9.3 - 2019-04-19
+  - Fix compiler errors on GCC when compiling with -std=c99.
+v0.9.2 - 2019-04-08
+  - Add support for per-context user data.
+  - Fix a potential bug with context configs.
+  - Fix some bugs with PulseAudio.
+v0.9.1 - 2019-03-17
+  - Fix a bug where the output buffer is not getting zeroed out before calling the data callback. This happens when
+    the device is running in passthrough mode (not doing any data conversion).
+  - Fix an issue where the data callback is getting called too frequently on the WASAPI and DirectSound backends.
+  - Fix error on the UWP build.
+  - Fix a build error on Apple platforms.
+v0.9 - 2019-03-06
+  - Rebranded to "miniaudio". All namespaces have been renamed from "mal" to "ma".
+  - API CHANGE: ma_device_init() and ma_device_config_init() have changed significantly:
+    - The device type, device ID and user data pointer have moved from ma_device_init() to the config.
+    - All variations of ma_device_config_init_*() have been removed in favor of just ma_device_config_init().
+    - ma_device_config_init() now takes only one parameter which is the device type. All other properties need
+      to be set on the returned object directly.
+    - The onDataCallback and onStopCallback members of ma_device_config have been renamed to "dataCallback"
+      and "stopCallback".
+    - The ID of the physical device is now split into two: one for the playback device and the other for the
+      capture device. This is required for full-duplex. These are named "pPlaybackDeviceID" and "pCaptureDeviceID".
+  - API CHANGE: The data callback has changed. It now uses a unified callback for all device types rather than
+    being separate for each. It now takes two pointers - one containing input data and the other output data. This
+    design in required for full-duplex. The return value is now void instead of the number of frames written. The
+    new callback looks like the following:
+        void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
+  - API CHANGE: Remove the log callback parameter from ma_context_config_init(). With this change,
+    ma_context_config_init() now takes no parameters and the log callback is set via the structure directly. The
+    new policy for config initialization is that only mandatory settings are passed in to *_config_init(). The
+    "onLog" member of ma_context_config has been renamed to "logCallback".
+  - API CHANGE: Remove ma_device_get_buffer_size_in_bytes().
+  - API CHANGE: Rename decoding APIs to "pcm_frames" convention.
+    - mal_decoder_read()          -> ma_decoder_read_pcm_frames()
+    - mal_decoder_seek_to_frame() -> ma_decoder_seek_to_pcm_frame()
+  - API CHANGE: Rename sine wave reading APIs to f32 convention.
+    - mal_sine_wave_read()    -> ma_sine_wave_read_f32()
+    - mal_sine_wave_read_ex() -> ma_sine_wave_read_f32_ex()
+  - API CHANGE: Remove some deprecated APIs
+    - mal_device_set_recv_callback()
+    - mal_device_set_send_callback()
+    - mal_src_set_input_sample_rate()
+    - mal_src_set_output_sample_rate()
+  - API CHANGE: Add log level to the log callback. New signature:
+    - void on_log(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message)
+  - API CHANGE: Changes to result codes. Constants have changed and unused codes have been removed. If you're
+    a binding mainainer you will need to update your result code constants.
+  - API CHANGE: Change the order of the ma_backend enums to priority order. If you are a binding maintainer, you
+    will need to update.
+  - API CHANGE: Rename mal_dsp to ma_pcm_converter. All functions have been renamed from mal_dsp_*() to
+    ma_pcm_converter_*(). All structures have been renamed from mal_dsp* to ma_pcm_converter*.
+  - API CHANGE: Reorder parameters of ma_decoder_read_pcm_frames() to be consistent with the new parameter order scheme.
+  - The resampling algorithm has been changed from sinc to linear. The rationale for this is that the sinc implementation
+    is too inefficient right now. This will hopefully be improved at a later date.
+  - Device initialization will no longer fall back to shared mode if exclusive mode is requested but is unusable.
+    With this change, if you request an device in exclusive mode, but exclusive mode is not supported, it will not
+    automatically fall back to shared mode. The client will need to reinitialize the device in shared mode if that's
+    what they want.
+  - Add ring buffer API. This is ma_rb and ma_pcm_rb, the difference being that ma_rb operates on bytes and
+    ma_pcm_rb operates on PCM frames.
+  - Add Web Audio backend. This is used when compiling with Emscripten. The SDL backend, which was previously
+    used for web support, will be removed in a future version.
+  - Add AAudio backend (Android Audio). This is the new priority backend for Android. Support for AAudio starts
+    with Android 8. OpenSL|ES is used as a fallback for older versions of Android.
+  - Remove OpenAL and SDL backends.
+  - Fix a possible deadlock when rapidly stopping the device after it has started.
+  - Update documentation.
+  - Change licensing to a choice of public domain _or_ MIT-0 (No Attribution).
+v0.8.14 - 2018-12-16
+  - Core Audio: Fix a bug where the device state is not set correctly after stopping.
+  - Add support for custom weights to the channel router.
+  - Update decoders to use updated APIs in dr_flac, dr_mp3 and dr_wav.
+v0.8.13 - 2018-12-04
+  - Core Audio: Fix a bug with channel mapping.
+  - Fix a bug with channel routing where the back/left and back/right channels have the wrong weight.
+v0.8.12 - 2018-11-27
+  - Drop support for SDL 1.2. The Emscripten build now requires "-s USE_SDL=2".
+  - Fix a linking error with ALSA.
+  - Fix a bug on iOS where the device name is not set correctly.
+v0.8.11 - 2018-11-21
+  - iOS bug fixes.
+  - Minor tweaks to PulseAudio.
+v0.8.10 - 2018-10-21
+  - Core Audio: Fix a hang when uninitializing a device.
+  - Fix a bug where an incorrect value is returned from mal_device_stop().
+v0.8.9 - 2018-09-28
+  - Fix a bug with the SDL backend where device initialization fails.
+v0.8.8 - 2018-09-14
+  - Fix Linux build with the ALSA backend.
+  - Minor documentation fix.
+v0.8.7 - 2018-09-12
+  - Fix a bug with UWP detection.
+v0.8.6 - 2018-08-26
+  - Automatically switch the internal device when the default device is unplugged. Note that this is still in the
+    early stages and not all backends handle this the same way. As of this version, this will not detect a default
+    device switch when changed from the operating system's audio preferences (unless the backend itself handles
+    this automatically). This is not supported in exclusive mode.
+  - WASAPI and Core Audio: Add support for stream routing. When the application is using a default device and the
+    user switches the default device via the operating system's audio preferences, miniaudio will automatically switch
+    the internal device to the new default. This is not supported in exclusive mode.
+  - WASAPI: Add support for hardware offloading via IAudioClient2. Only supported on Windows 8 and newer.
+  - WASAPI: Add support for low-latency shared mode via IAudioClient3. Only supported on Windows 10 and newer.
+  - Add support for compiling the UWP build as C.
+  - mal_device_set_recv_callback() and mal_device_set_send_callback() have been deprecated. You must now set this
+    when the device is initialized with mal_device_init*(). These will be removed in version 0.9.0.
+v0.8.5 - 2018-08-12
+  - Add support for specifying the size of a device's buffer in milliseconds. You can still set the buffer size in
+    frames if that suits you. When bufferSizeInFrames is 0, bufferSizeInMilliseconds will be used. If both are non-0
+    then bufferSizeInFrames will take priority. If both are set to 0 the default buffer size is used.
+  - Add support for the audio(4) backend to OpenBSD.
+  - Fix a bug with the ALSA backend that was causing problems on Raspberry Pi. This significantly improves the
+    Raspberry Pi experience.
+  - Fix a bug where an incorrect number of samples is returned from sinc resampling.
+  - Add support for setting the value to be passed to internal calls to CoInitializeEx().
+  - WASAPI and WinMM: Stop the device when it is unplugged.
+v0.8.4 - 2018-08-06
+  - Add sndio backend for OpenBSD.
+  - Add audio(4) backend for NetBSD.
+  - Drop support for the OSS backend on everything except FreeBSD and DragonFly BSD.
+  - Formats are now native-endian (were previously little-endian).
+  - Mark some APIs as deprecated:
+    - mal_src_set_input_sample_rate() and mal_src_set_output_sample_rate() are replaced with mal_src_set_sample_rate().
+    - mal_dsp_set_input_sample_rate() and mal_dsp_set_output_sample_rate() are replaced with mal_dsp_set_sample_rate().
+  - Fix a bug when capturing using the WASAPI backend.
+  - Fix some aliasing issues with resampling, specifically when increasing the sample rate.
+  - Fix warnings.
+v0.8.3 - 2018-07-15
+  - Fix a crackling bug when resampling in capture mode.
+  - Core Audio: Fix a bug where capture does not work.
+  - ALSA: Fix a bug where the worker thread can get stuck in an infinite loop.
+  - PulseAudio: Fix a bug where mal_context_init() succeeds when PulseAudio is unusable.
+  - JACK: Fix a bug where mal_context_init() succeeds when JACK is unusable.
+v0.8.2 - 2018-07-07
+  - Fix a bug on macOS with Core Audio where the internal callback is not called.
+v0.8.1 - 2018-07-06
+  - Fix compilation errors and warnings.
+v0.8 - 2018-07-05
+  - Changed MAL_IMPLEMENTATION to MINI_AL_IMPLEMENTATION for consistency with other libraries. The old
+    way is still supported for now, but you should update as it may be removed in the future.
+  - API CHANGE: Replace device enumeration APIs. mal_enumerate_devices() has been replaced with
+    mal_context_get_devices(). An additional low-level device enumration API has been introduced called
+    mal_context_enumerate_devices() which uses a callback to report devices.
+  - API CHANGE: Rename mal_get_sample_size_in_bytes() to mal_get_bytes_per_sample() and add
+    mal_get_bytes_per_frame().
+  - API CHANGE: Replace mal_device_config.preferExclusiveMode with mal_device_config.shareMode.
+    - This new config can be set to mal_share_mode_shared (default) or mal_share_mode_exclusive.
+  - API CHANGE: Remove excludeNullDevice from mal_context_config.alsa.
+  - API CHANGE: Rename MAL_MAX_SAMPLE_SIZE_IN_BYTES to MAL_MAX_PCM_SAMPLE_SIZE_IN_BYTES.
+  - API CHANGE: Change the default channel mapping to the standard Microsoft mapping.
+  - API CHANGE: Remove backend-specific result codes.
+  - API CHANGE: Changes to the format conversion APIs (mal_pcm_f32_to_s16(), etc.)
+  - Add support for Core Audio (Apple).
+  - Add support for PulseAudio.
+    - This is the highest priority backend on Linux (higher priority than ALSA) since it is commonly
+      installed by default on many of the popular distros and offer's more seamless integration on
+      platforms where PulseAudio is used. In addition, if PulseAudio is installed and running (which
+      is extremely common), it's better to just use PulseAudio directly rather than going through the
+      "pulse" ALSA plugin (which is what the "default" ALSA device is likely set to).
+  - Add support for JACK.
+  - Remove dependency on asound.h for the ALSA backend. This means the ALSA development packages are no
+    longer required to build miniaudio.
+  - Remove dependency on dsound.h for the DirectSound backend. This fixes build issues with some
+    distributions of MinGW.
+  - Remove dependency on audioclient.h for the WASAPI backend. This fixes build issues with some
+    distributions of MinGW.
+  - Add support for dithering to format conversion.
+  - Add support for configuring the priority of the worker thread.
+  - Add a sine wave generator.
+  - Improve efficiency of sample rate conversion.
+  - Introduce the notion of standard channel maps. Use mal_get_standard_channel_map().
+  - Introduce the notion of default device configurations. A default config uses the same configuration
+    as the backend's internal device, and as such results in a pass-through data transmission pipeline.
+  - Add support for passing in NULL for the device config in mal_device_init(), which uses a default
+    config. This requires manually calling mal_device_set_send/recv_callback().
+  - Add support for decoding from raw PCM data (mal_decoder_init_raw(), etc.)
+  - Make mal_device_init_ex() more robust.
+  - Make some APIs more const-correct.
+  - Fix errors with SDL detection on Apple platforms.
+  - Fix errors with OpenAL detection.
+  - Fix some memory leaks.
+  - Fix a bug with opening decoders from memory.
+  - Early work on SSE2, AVX2 and NEON optimizations.
+  - Miscellaneous bug fixes.
+  - Documentation updates.
+v0.7 - 2018-02-25
+  - API CHANGE: Change mal_src_read_frames() and mal_dsp_read_frames() to use 64-bit sample counts.
+  - Add decoder APIs for loading WAV, FLAC, Vorbis and MP3 files.
+  - Allow opening of devices without a context.
+    - In this case the context is created and managed internally by the device.
+  - Change the default channel mapping to the same as that used by FLAC.
+  - Fix build errors with macOS.
+v0.6c - 2018-02-12
+  - Fix build errors with BSD/OSS.
+v0.6b - 2018-02-03
+  - Fix some warnings when compiling with Visual C++.
+v0.6a - 2018-01-26
+  - Fix errors with channel mixing when increasing the channel count.
+  - Improvements to the build system for the OpenAL backend.
+  - Documentation fixes.
+v0.6 - 2017-12-08
+  - API CHANGE: Expose and improve mutex APIs. If you were using the mutex APIs before this version you'll
+    need to update.
+  - API CHANGE: SRC and DSP callbacks now take a pointer to a mal_src and mal_dsp object respectively.
+  - API CHANGE: Improvements to event and thread APIs. These changes make these APIs more consistent.
+  - Add support for SDL and Emscripten.
+  - Simplify the build system further for when development packages for various backends are not installed.
+    With this change, when the compiler supports __has_include, backends without the relevant development
+    packages installed will be ignored. This fixes the build for old versions of MinGW.
+  - Fixes to the Android build.
+  - Add mal_convert_frames(). This is a high-level helper API for performing a one-time, bulk conversion of
+    audio data to a different format.
+  - Improvements to f32 -> u8/s16/s24/s32 conversion routines.
+  - Fix a bug where the wrong value is returned from mal_device_start() for the OpenSL backend.
+  - Fixes and improvements for Raspberry Pi.
+  - Warning fixes.
+v0.5 - 2017-11-11
+  - API CHANGE: The mal_context_init() function now takes a pointer to a mal_context_config object for
+    configuring the context. The works in the same kind of way as the device config. The rationale for this
+    change is to give applications better control over context-level properties, add support for backend-
+    specific configurations, and support extensibility without breaking the API.
+  - API CHANGE: The alsa.preferPlugHW device config variable has been removed since it's not really useful for
+    anything anymore.
+  - ALSA: By default, device enumeration will now only enumerate over unique card/device pairs. Applications
+    can enable verbose device enumeration by setting the alsa.useVerboseDeviceEnumeration context config
+    variable.
+  - ALSA: When opening a device in shared mode (the default), the dmix/dsnoop plugin will be prioritized. If
+    this fails it will fall back to the hw plugin. With this change the preferExclusiveMode config is now
+    honored. Note that this does not happen when alsa.useVerboseDeviceEnumeration is set to true (see above)
+    which is by design.
+  - ALSA: Add support for excluding the "null" device using the alsa.excludeNullDevice context config variable.
+  - ALSA: Fix a bug with channel mapping which causes an assertion to fail.
+  - Fix errors with enumeration when pInfo is set to NULL.
+  - OSS: Fix a bug when starting a device when the client sends 0 samples for the initial buffer fill.
+v0.4 - 2017-11-05
+  - API CHANGE: The log callback is now per-context rather than per-device and as is thus now passed to
+    mal_context_init(). The rationale for this change is that it allows applications to capture diagnostic
+    messages at the context level. Previously this was only available at the device level.
+  - API CHANGE: The device config passed to mal_device_init() is now const.
+  - Added support for OSS which enables support on BSD platforms.
+  - Added support for WinMM (waveOut/waveIn).
+  - Added support for UWP (Universal Windows Platform) applications. Currently C++ only.
+  - Added support for exclusive mode for selected backends. Currently supported on WASAPI.
+  - POSIX builds no longer require explicit linking to libpthread (-lpthread).
+  - ALSA: Explicit linking to libasound (-lasound) is no longer required.
+  - ALSA: Latency improvements.
+  - ALSA: Use MMAP mode where available. This can be disabled with the alsa.noMMap config.
+  - ALSA: Use "hw" devices instead of "plughw" devices by default. This can be disabled with the
+    alsa.preferPlugHW config.
+  - WASAPI is now the highest priority backend on Windows platforms.
+  - Fixed an error with sample rate conversion which was causing crackling when capturing.
+  - Improved error handling.
+  - Improved compiler support.
+  - Miscellaneous bug fixes.
+v0.3 - 2017-06-19
+  - API CHANGE: Introduced the notion of a context. The context is the highest level object and is required for
+    enumerating and creating devices. Now, applications must first create a context, and then use that to
+    enumerate and create devices. The reason for this change is to ensure device enumeration and creation is
+    tied to the same backend. In addition, some backends are better suited to this design.
+  - API CHANGE: Removed the rewinding APIs because they're too inconsistent across the different backends, hard
+    to test and maintain, and just generally unreliable.
+  - Added helper APIs for initializing mal_device_config objects.
+  - Null Backend: Fixed a crash when recording.
+  - Fixed build for UWP.
+  - Added support for f32 formats to the OpenSL|ES backend.
+  - Added initial implementation of the WASAPI backend.
+  - Added initial implementation of the OpenAL backend.
+  - Added support for low quality linear sample rate conversion.
+  - Added early support for basic channel mapping.
+v0.2 - 2016-10-28
+  - API CHANGE: Add user data pointer as the last parameter to mal_device_init(). The rationale for this
+    change is to ensure the logging callback has access to the user data during initialization.
+  - API CHANGE: Have device configuration properties be passed to mal_device_init() via a structure. Rationale:
+    1) The number of parameters is just getting too much.
+    2) It makes it a bit easier to add new configuration properties in the future. In particular, there's a
+       chance there will be support added for backend-specific properties.
+  - Dropped support for f64, A-law and Mu-law formats since they just aren't common enough to justify the
+    added maintenance cost.
+  - DirectSound: Increased the default buffer size for capture devices.
+  - Added initial implementation of the OpenSL|ES backend.
+v0.1 - 2016-10-21
+  - Initial versioned release.
\ No newline at end of file
--- a/miniaudio.h
+++ b/miniaudio.h
@@ -90098,1062 +90098,6 @@ DRMP3_API void drmp3_free(void* p, const drmp3_allocation_callbacks* pAllocation
 #endif  /* miniaudio_c */
 #endif  /* MINIAUDIO_IMPLEMENTATION */
-/*
-RELEASE NOTES - VERSION 0.11.x
-==============================
-Version 0.11 adds some high level APIs and includes some breaking API changes.
-High Level Engine API
---------------------
-The engine API provides sound management, mixing and effect processing. It can be used as both an
-alternative to the low level device API or as a supplementary feature depending on your needs. See
-the documentation for information on how to use this API.
-Resource Management
-------------------
-Support for management of sound files has been added via the new `ma_resource_manager` API. This
-supports loading of sounds entirely in memory with reference counting and also streaming. See the
-documentation for more information on the resource manager.
-Node Graphs
-----------
-A node graph system has been added to support complex effect chains and mixing. The node graph is
-made up of a series of connected nodes, with the output of one node being fed into the input of
-another node. By connecting nodes together you can create complex effect and mixing graphs. A
-number of different types of nodes are included, but custom nodes are also supported. See the
-documentation for more information on the node graph system.
-Breaking API Changes
--------------------
-There are many breaking API changes in this release.
-  - The "loop" parameter has been removed from ma_data_source_read_pcm_frames(). To enabled looping
-    you now need to call ma_data_source_set_looping(). The looping state can be retrieved with
-    ma_data_source_is_looping().
-  - ma_channel_mix_mode_planar_blend has been removed. Use ma_channel_mix_mode_rectangular instead.
-  - MA_MIN_SAMPLE_RATE and MA_MAX_SAMPLE_RATE have been removed. Use ma_standard_sample_rate_min
-    and ma_standard_sample_rate_max instead.
-  - The ma_device_info structure has changed. Now, an array of supported format/channels/rate
-    combinations are stored in an array and the old minChannels, maxChannels, etc. have been
-    removed.
-  - The shareMode parameter has been removed from ma_context_get_device_info(). Whether or not the
-    format relates to a shared or exclusive device is now set in a flags variables.
-  - The noPreZeroedOutputBuffer variable in the device config has been renamed to
-    noPreSilencedOutputBuffer.
-  - The pBufferOut parameter has been removed from the ring buffer commit functions.
-  - The ma_zero_pcm_frames() function has been removed. Use ma_silence_pcm_frames() instead.
-  - ma_clip_samples_f32() has been updated to take an input and output buffer rather than working
-    exclusively in-place.
-  - ma_clip_pcm_frames_f32() has been removed. Use ma_clip_samples_f32() or ma_clip_pcm_frames()
-    instead.
-  - ma_data_source_callbacks has been removed. If you're building a custom data source you need to
-    use ma_data_source_vtable instead.
-  - Support for mapping has been removed from data sources since it had very limited use and has
-    proven to not be worth the maintenance cost.
-  - The onLog callback has been removed from the context config. This has been replaced with a much
-    more flexible logging system. The context will have a `ma_log` object which can be retrieved
-    with `ma_context_get_log()`. From there you can register a callback with the `ma_log` object
-    with `ma_log_register_callback()`. An pre-allocated log can be specified in the context config.
-  - MA_LOG_LEVEL_VERBOSE has been removed and replaced with MA_LOG_LEVEL_DEBUG. With this change,
-    logs posted with MA_LOG_LEVEL_DEBUG will only be output if MA_DEBUG_OUTPUT is enabled.
-  - MA_LOG_LEVEL has been removed. Now all log levels are posted, except for MA_LOG_LEVEL_DEBUG
-    which will only be posted if MA_DEBUG_OUTPUT is enabled.
-  - ma_resource_format has been replaced with ma_encoding_format.
-  - The encoding-specific initialization functions for decoders, such as ma_decoder_init_wav(),
-    have been removed. Instead you should set the encodingFormat variable in the decoder config.
-  - ma_decoder_get_length_in_pcm_frames() has been updated to return a result code and output the
-    length via an output parameter. This makes it consistent with data sources.
-  - ma_decoder_read_pcm_frames() has been updated to return a result code and output the number of
-    frames read via an output parameter.
-  - Allocation callbacks must now implement the onRealloc() callback. Previously miniaudio would
-    emulate this in terms of malloc and free, but for simplicity it is now required that allocation
-    callbacks handle this theselves.
-  - ma_get_standard_channel_map() has been renamed to ma_channel_map_init_standard() and now
-    includes a parameter specifying the capacity of the output channel map. This is useful for
-    initializing a channel map with a fixed capacity.
-  - ma_channel_map_valid() has been renamed to ma_channel_map_is_valid().
-  - ma_channel_map_equal() has been renamed to ma_channel_map_is_equal().
-  - ma_channel_map_blank() has been renamed to ma_channel_map_is_blank().
-  - The Speex resampler has been removed. Support for custom resamplers has been added. If you need
-    a Speex resampler you will need to implement a custom resampler.
-  - The following functions from the resampler have changed to return a result code and output
-    their result via an output parameter:
-    - ma_linear_resampler_get_required_input_frame_count()
-    - ma_linear_resampler_get_expected_output_frame_count()
-    - ma_resampler_get_required_input_frame_count()
-    - ma_resampler_get_expected_output_frame_count()
-  - Many init/uninit functions have been changed to take a pointer to allocation callbacks.
-  - ma_scale_buffer_size() has been removed.
-  - ma_encoder_write_pcm_frames() has been updated to return a result code and output the number of
-    frames written via an output parameter.
-  - ma_noise_read_pcm_frames() has been updated to return a result code and output the number of
-    frames read via an output parameter.
-  - ma_waveform_read_pcm_frames() has been updated to return a result code and output the number of
-    frames read via an output parameter.
-  - The MA_STATE_* tokens have been reanmed to ma_device_state_* and turned into an enum.
-  - ma_factor_to_gain_db() has been renamed to ma_volume_linear_to_db()
-  - ma_gain_db_to_factor() has been renamed to ma_volume_db_to_linear()
-Changes to Custom Data Sources
------------------------------
-The implementation of data sources has seen some changes. Previously, you needed to only implement
-the `ma_data_source_callbacks` structure and make it the first member of the data source's
-structure. This has now changed. You now need to make the first member of the data source's
-structure `ma_data_source_base` and initialize and uninitialize it with `ma_data_source_init()` and
-`ma_data_source_uninit()` from your data source's init/uninit routines.
-When you call `ma_data_source_init()` you will need to specify a config. The config includes a
-pointer to a `ma_data_source_vtable` object which needs to be filled out.
-The onGetDataFormat callback has been changed to output the channel map. With this change,
-ma_data_source_get_data_format() has been update to accept a channel map.
-The onMap and onUnmap callbacks have been removed, as has ma_data_source_map() and
-ma_data_source_unmap().
-Other Changes
-------------
-There have also been some other smaller changes added to this release.
-  - Support for custom resmplers has been added. See the implementation of ma_linear_resampler for
-    an example on how to implement a custom resmpler.
-  - ma_decoder_get_data_format() has been added.
-  - Support has been added for disabling denormals on the audio thread. This is configured on a
-    per-device basis via the device config and are disabled by default. Use the noDisableDenormals
-    config variable to control this.
-  - A delay/echo effect has been added called `ma_delay`.
-  - A stereo pan effect has been added called `ma_panner`.
-  - A spataializer effect has been added called `ma_spatializer`. This is used by the engine to
-    achieve it's 3D spatialization effect for sounds.
-  - ma_free() now performs a null pointer check which means NULL should never be passed into
-    allocation callbacks.
-  - Filters, effects and data conversion no longer have a dependency on MA_MAX_CHANNELS.
-  - MA_MAX_CHANNELS has been increased from 32 to 254 and currently only affects devices. This may
-    be relaxed in a future version.
-*/
-/*
-REVISION HISTORY
-================
-v0.11.4 - 2022-01-12
-  - AAudio: Add initial support for automatic stream routing.
-  - Add support for initializing an encoder from a VFS.
-  - Fix a bug when initializing an encoder from a file where the file handle does not get closed in
-    the event of an error.
-v0.11.3 - 2022-01-07
-  - Add a new flag for nodes called MA_NODE_FLAG_SILENT_OUTPUT which tells miniaudio that the
-    output of the node should always be treated as silence. This gives miniaudio an optimization
-    opportunity by skipping mixing of those nodes. Useful for special nodes that need to have
-    their outputs wired up to the graph so they're processed, but don't want the output to
-    contribute to the final mix.
-  - Add support for fixed sized callbacks. With this change, the data callback will be fired with
-    a consistent frame count based on the periodSizeInFrames or periodSizeInMilliseconds config
-    variable (depending on which one is used). If the period size is not specified, the backend's
-    internal period size will be used. Under the hood this uses an intermediary buffer which
-    introduces a small inefficiency. To avoid this you can use the `noFixedSizedCallback` config
-    variable and set it to true. This will make the callback equivalent to the way it was before
-    this change and will avoid the intermediary buffer, but the data callback could get fired with
-    an inconsistent frame count which might cause problems where certain operations need to operate
-    on fixed sized chunks.
-  - Change the logging system to always process debug log messages. This is useful for allowing
-    debug and test builds of applications to output debug information that can later be passed on
-    for debugging in miniaudio. To filter out these messages, just filter against the log level
-    which will be MA_LOG_LEVEL_DEBUG.
-  - Change the wav decoder to pick the closest format to the source file by default if no preferred
-    format is specified.
-  - Fix a bug where ma_device_get_info() and ma_device_get_name() return an error.
-  - Fix a bug where ma_data_source_read_pcm_frames() can return MA_AT_END even when some data has
-    been read. MA_AT_END should only be returned when nothing has been read.
-  - PulseAudio: Fix some bugs where starting and stopping a device can result in a deadlock.
-v0.11.2 - 2021-12-31
-  - Add a new device notification system to replace the stop callback. The stop callback is still
-    in place, but will be removed in version 0.12. New code should use the notificationCallback
-    member in the device config instead of stopCallback.
-  - Fix a bug where the stopped notification doesn't get fired.
-  - iOS: The IO buffer size is now configured based on the device's configured period size.
-  - WebAudio: Optimizations to some JavaScript code.
-v0.11.1 - 2021-12-27
-  - Result codes are now declared as an enum rather than #defines.
-  - Channel positions (MA_CHANNEL_*) are now declared as an enum rather than #defines.
-  - Add ma_device_get_info() for retrieving device information from an initialized device.
-  - Add ma_device_get_name() for retrieving the name of an initialized device.
-  - Add support for setting the directional attenuation factor to sounds and groups.
-  - Fix a crash when passing in NULL for the pEngine parameter of ma_engine_init().
-  - Fix a bug where the node graph will output silence if a node has zero input connections.
-  - Fix a bug in the engine where sounds in front of the listener are too loud.
-  - AAudio: Fix an incorrect assert.
-  - AAudio: Fix a bug that resulted in exclusive mode always resulting in initialization failure.
-  - AAudio: Fix a bug that resulted in a capture device incorrectly being detected as disconnected.
-  - OpenSL: Fix an error when initializing a device with a non-NULL device ID.
-  - OpenSL: Fix some bugs with device initialization.
-v0.11.0 - 2021-12-18
-  - Add a node graph system for advanced mixing and effect processing.
-  - Add a resource manager for loading and streaming sounds.
-  - Add a high level engine API for sound management and mixing. This wraps around the node graph
-    and resource manager.
-  - Add support for custom resmplers.
-  - Add ma_decoder_get_data_format().
-  - Add support for disabling denormals on the audio thread.
-  - Add a delay/echo effect called ma_delay.
-  - Add a stereo pan effect called ma_panner.
-  - Add a spataializer effect called ma_spatializer.
-  - Add support for amplification for device master volume.
-  - Remove dependency on MA_MAX_CHANNELS from filters and data conversion.
-  - Increase MA_MAX_CHANNELS from 32 to 254.
-  - API CHANGE: Changes have been made to the way custom data sources are made. See documentation
-    on how to implement custom data sources.
-  - API CHANGE: Remove ma_data_source_map() and ma_data_source_unmap()
-  - API CHANGE: Remove the `loop` parameter from ma_data_source_read_pcm_frames(). Use
-    ma_data_source_set_looping() to enable or disable looping.
-  - API CHANGE: Remove ma_channel_mix_mode_planar_blend. Use ma_channel_mix_mode_rectangular instead.
-  - API CHANGE: Remove MA_MIN_SAMPLE_RATE and MA_MAX_SAMPLE_RATE. Use ma_standard_sample_rate_min
-    and ma_standard_sample_rate_max instead.
-  - API CHANGE: Changes have been made to the ma_device_info structure. See documentation for
-    details of these changes.
-  - API CHANGE: Remove the `shareMode` parameter from ma_context_get_device_info().
-  - API CHANGE: Rename noPreZeroedOutputBuffer to noPreSilencedOutputBuffer in the device config.
-  - API CHANGE: Remove pBufferOut parameter from ring buffer commit functions.
-  - API CHANGE: Remove ma_zero_pcm_frames(). Use ma_silence_pcm_frames() instead.
-  - API CHANGE: Change ma_clip_samples_f32() to take input and output buffers rather than working
-    exclusively in-place.
-  - API CHANGE: Remove ma_clip_pcm_frames_f32(). Use ma_clip_samples_f32() or ma_clip_pcm_frames()
-    instead.
-  - API CHANGE: Remove the onLog callback from the context config and replaced with a more
-    flexible system. See the documentation for how to use logging.
-  - API CHANGE: Remove MA_LOG_LEVEL_VERBOSE and add MA_LOG_LEVEL_DEBUG. Logs using the
-    MA_LOG_LEVEL_DEBUG logging level will only be output when miniaudio is compiled with the
-    MA_DEBUG_OUTPUT option.
-  - API CHANGE: MA_LOG_LEVEL has been removed. All log levels will be posted, except for
-    MA_LOG_LEVEL_DEBUG which will only be output when MA_DEBUG_OUTPUT is enabled.
-  - API CHANGE: Rename ma_resource_format to ma_encoding_format.
-  - API CHANGE: Remove all encoding-specific initialization routines for decoders. Use the
-    encodingFormat properties in the decoder config instead.
-  - API CHANGE: Change ma_decoder_get_length_in_pcm_frames() to return a result code and output the
-    number of frames read via an output parameter.
-  - API CHANGE: Allocation callbacks must now implement the onRealloc() callback.
-  - API CHANGE: Remove ma_get_standard_channel_map() and add ma_channel_map_init_standard().
-  - API CHANGE: Rename ma_channel_map_valid() to ma_channel_map_is_valid().
-  - API CHANGE: Rename ma_channel_map_equal() to ma_channel_map_is_equal().
-  - API CHANGE: Rename ma_channel_map_blank() to ma_channel_map_is_blank().
-  - API CHANGE: Remove the Speex resampler. Use a custom resampler instead.
-  - API CHANGE: Change the following resampler APIs to return a result code and output their result
-    via an output parameter:
-    - ma_linear_resampler_get_required_input_frame_count()
-    - ma_linear_resampler_get_expected_output_frame_count()
-    - ma_resampler_get_required_input_frame_count()
-    - ma_resampler_get_expected_output_frame_count()
-  - API CHANGE: Update relevant init/uninit functions to take a pointer to allocation callbacks.
-  - API CHANGE: Remove ma_scale_buffer_size()
-  - API CHANGE: Update ma_encoder_write_pcm_frames() to return a result code and output the number
-    of frames written via an output parameter.
-  - API CHANGE: Update ma_noise_read_pcm_frames() to return a result code and output the number of
-    frames read via an output parameter.
-  - API CHANGE: Update ma_waveform_read_pcm_frames() to return a result code and output the number
-    of frames read via an output parameter.
-  - API CHANGE: Remove The MA_STATE_* and add ma_device_state_* enums.
-  - API CHANGE: Rename ma_factor_to_gain_db() to ma_volume_linear_to_db().
-  - API CHANGE: Rename ma_gain_db_to_factor() to ma_volume_db_to_linear().
-  - API CHANGE: Rename ma_device_set_master_gain_db() to ma_device_set_master_volume_db().
-  - API CHANGE: Rename ma_device_get_master_gain_db() to ma_device_get_master_volume_db()
-v0.10.43 - 2021-12-10
-  - ALSA: Fix use of uninitialized variables.
-  - ALSA: Fix enumeration of devices that support both playback and capture.
-  - PulseAudio: Fix a possible division by zero.
-  - WebAudio: Fix errors in strict mode.
-v0.10.42 - 2021-08-22
-  - Fix a possible deadlock when stopping devices.
-v0.10.41 - 2021-08-15
-  - Core Audio: Fix some deadlock errors.
-v0.10.40 - 2021-07-23
-  - Fix a bug when converting from stereo to mono.
-  - PulseAudio: Fix a glitch when pausing and resuming a device.
-v0.10.39 - 2021-07-20
-  - Core Audio: Fix a deadlock when the default device is changed.
-  - Core Audio: Fix compilation errors on macOS and iOS.
-  - PulseAudio: Fix a bug where the stop callback is not fired when a device is unplugged.
-  - PulseAudio: Fix a null pointer dereference.
-v0.10.38 - 2021-07-14
-  - Fix a linking error when MA_DEBUG_OUTPUT is not enabled.
-  - Fix an error where ma_log_postv() does not return a value.
-  - OpenSL: Fix a bug with setting of stream types and recording presets.
-0.10.37 - 2021-07-06
-  - Fix a bug with log message formatting.
-  - Fix build when compiling with MA_NO_THREADING.
-  - Minor updates to channel mapping.
-0.10.36 - 2021-07-03
-  - Add support for custom decoding backends.
-  - Fix some bugs with the Vorbis decoder.
-  - PulseAudio: Fix a bug with channel mapping.
-  - PulseAudio: Fix a bug where miniaudio does not fall back to a supported format when PulseAudio
-    defaults to a format not known to miniaudio.
-  - OpenSL: Fix a crash when initializing a capture device when a recording preset other than the
-    default is specified.
-  - Silence some warnings when compiling with MA_DEBUG_OUTPUT
-  - Improvements to logging. See the `ma_log` API for details. The logCallback variable used by
-    ma_context has been deprecated and will be replaced with the new system in version 0.11.
-    - Initialize an `ma_log` object with `ma_log_init()`.
-    - Register a callback with `ma_log_register_callback()`.
-    - In the context config, set `pLog` to your `ma_log` object and stop using `logCallback`.
-  - Prep work for some upcoming changes to data sources. These changes are still compatible with
-    existing code, however code will need to be updated in preparation for version 0.11 which will
-    be breaking. You should make these changes now for any custom data sources:
-    - Change your base data source object from `ma_data_source_callbacks` to `ma_data_source_base`.
-    - Call `ma_data_source_init()` for your base object in your custom data source's initialization
-      routine. This takes a config object which includes a pointer to a vtable which is now where
-      your custom callbacks are defined.
-    - Call `ma_data_source_uninit()` in your custom data source's uninitialization routine. This
-      doesn't currently do anything, but it placeholder in case some future uninitialization code
-      is required to be added at a later date.
-v0.10.35 - 2021-04-27
-  - Fix the C++ build.
-v0.10.34 - 2021-04-26
-  - WASAPI: Fix a bug where a result code is not getting checked at initialization time.
-  - WASAPI: Bug fixes for loopback mode.
-  - ALSA: Fix a possible deadlock when stopping devices.
-  - Mark devices as default on the null backend.
-v0.10.33 - 2021-04-04
-  - Core Audio: Fix a memory leak.
-  - Core Audio: Fix a bug where the performance profile is not being used by playback devices.
-  - JACK: Fix loading of 64-bit JACK on Windows.
-  - Fix a calculation error and add a safety check to the following APIs to prevent a division by zero:
-    - ma_calculate_buffer_size_in_milliseconds_from_frames()
-    - ma_calculate_buffer_size_in_frames_from_milliseconds()
-  - Fix compilation errors relating to c89atomic.
-  - Update FLAC decoder.
-v0.10.32 - 2021-02-23
-  - WASAPI: Fix a deadlock in exclusive mode.
-  - WASAPI: No longer return an error from ma_context_get_device_info() when an exclusive mode format
-    cannot be retrieved.
-  - WASAPI: Attempt to fix some bugs with device uninitialization.
-  - PulseAudio: Yet another refactor, this time to remove the dependency on `pa_threaded_mainloop`.
-  - Web Audio: Fix a bug on Chrome and any other browser using the same engine.
-  - Web Audio: Automatically start the device on some user input if the device has been started. This
-    is to work around Google's policy of not starting audio if no user input has yet been performed.
-  - Fix a bug where thread handles are not being freed.
-  - Fix some static analysis warnings in FLAC, WAV and MP3 decoders.
-  - Fix a warning due to referencing _MSC_VER when it is undefined.
-  - Update to latest version of c89atomic.
-  - Internal refactoring to migrate over to the new backend callback system for the following backends:
-    - PulseAudio
-    - ALSA
-    - Core Audio
-    - AAudio
-    - OpenSL|ES
-    - OSS
-    - audio(4)
-    - sndio
-v0.10.31 - 2021-01-17
-  - Make some functions const correct.
-  - Update ma_data_source_read_pcm_frames() to initialize pFramesRead to 0 for safety.
-  - Add the MA_ATOMIC annotation for use with variables that should be used atomically and remove unnecessary volatile qualifiers.
-  - Add support for enabling only specific backends at compile time. This is the reverse of the pre-existing system. With the new
-    system, all backends are first disabled with `MA_ENABLE_ONLY_SPECIFIC_BACKENDS`, which is then followed with `MA_ENABLE_*`. The
-    old system where you disable backends with `MA_NO_*` still exists and is still the default.
-v0.10.30 - 2021-01-10
-  - Fix a crash in ma_audio_buffer_read_pcm_frames().
-  - Update spinlock APIs to take a volatile parameter as input.
-  - Silence some unused parameter warnings.
-  - Fix a warning on GCC when compiling as C++.
-v0.10.29 - 2020-12-26
-  - Fix some subtle multi-threading bugs on non-x86 platforms.
-  - Fix a bug resulting in superfluous memory allocations when enumerating devices.
-  - Core Audio: Fix a compilation error when compiling for iOS.
-v0.10.28 - 2020-12-16
-  - Fix a crash when initializing a POSIX thread.
-  - OpenSL|ES: Respect the MA_NO_RUNTIME_LINKING option.
-v0.10.27 - 2020-12-04
-  - Add support for dynamically configuring some properties of `ma_noise` objects post-initialization.
-  - Add support for configuring the channel mixing mode in the device config.
-  - Fix a bug with simple channel mixing mode (drop or silence excess channels).
-  - Fix some bugs with trying to access uninitialized variables.
-  - Fix some errors with stopping devices for synchronous backends where the backend's stop callback would get fired twice.
-  - Fix a bug in the decoder due to using an uninitialized variable.
-  - Fix some data race errors.
-v0.10.26 - 2020-11-24
-  - WASAPI: Fix a bug where the exclusive mode format may not be retrieved correctly due to accessing freed memory.
-  - Fix a bug with ma_waveform where glitching occurs after changing frequency.
-  - Fix compilation with OpenWatcom.
-  - Fix compilation with TCC.
-  - Fix compilation with Digital Mars.
-  - Fix compilation warnings.
-  - Remove bitfields from public structures to aid in binding maintenance.
-v0.10.25 - 2020-11-15
-  - PulseAudio: Fix a bug where the stop callback isn't fired.
-  - WebAudio: Fix an error that occurs when Emscripten increases the size of it's heap.
-  - Custom Backends: Change the onContextInit and onDeviceInit callbacks to take a parameter which is a pointer to the config that was
-    passed into ma_context_init() and ma_device_init(). This replaces the deviceType parameter of onDeviceInit.
-  - Fix compilation warnings on older versions of GCC.
-v0.10.24 - 2020-11-10
-  - Fix a bug where initialization of a backend can fail due to some bad state being set from a prior failed attempt at initializing a
-    lower priority backend.
-v0.10.23 - 2020-11-09
-  - AAudio: Add support for configuring a playback stream's usage.
-  - Fix a compilation error when all built-in asynchronous backends are disabled at compile time.
-  - Fix compilation errors when compiling as C++.
-v0.10.22 - 2020-11-08
-  - Add support for custom backends.
-  - Add support for detecting default devices during device enumeration and with `ma_context_get_device_info()`.
-  - Refactor to the PulseAudio backend. This simplifies the implementation and fixes a capture bug.
-  - ALSA: Fix a bug in `ma_context_get_device_info()` where the PCM handle is left open in the event of an error.
-  - Core Audio: Further improvements to sample rate selection.
-  - Core Audio: Fix some bugs with capture mode.
-  - OpenSL: Add support for configuring stream types and recording presets.
-  - AAudio: Add support for configuring content types and input presets.
-  - Fix bugs in `ma_decoder_init_file*()` where the file handle is not closed after a decoding error.
-  - Fix some compilation warnings on GCC and Clang relating to the Speex resampler.
-  - Fix a compilation error for the Linux build when the ALSA and JACK backends are both disabled.
-  - Fix a compilation error for the BSD build.
-  - Fix some compilation errors on older versions of GCC.
-  - Add documentation for `MA_NO_RUNTIME_LINKING`.
-v0.10.21 - 2020-10-30
-  - Add ma_is_backend_enabled() and ma_get_enabled_backends() for retrieving enabled backends at run-time.
-  - WASAPI: Fix a copy and paste bug relating to loopback mode.
-  - Core Audio: Fix a bug when using multiple contexts.
-  - Core Audio: Fix a compilation warning.
-  - Core Audio: Improvements to sample rate selection.
-  - Core Audio: Improvements to format/channels/rate selection when requesting defaults.
-  - Core Audio: Add notes regarding the Apple notarization process.
-  - Fix some bugs due to null pointer dereferences.
-v0.10.20 - 2020-10-06
-  - Fix build errors with UWP.
-  - Minor documentation updates.
-v0.10.19 - 2020-09-22
-  - WASAPI: Return an error when exclusive mode is requested, but the native format is not supported by miniaudio.
-  - Fix a bug where ma_decoder_seek_to_pcm_frames() never returns MA_SUCCESS even though it was successful.
-  - Store the sample rate in the `ma_lpf` and `ma_hpf` structures.
-v0.10.18 - 2020-08-30
-  - Fix build errors with VC6.
-  - Fix a bug in channel converter for s32 format.
-  - Change channel converter configs to use the default channel map instead of a blank channel map when no channel map is specified when initializing the
-    config. This fixes an issue where the optimized mono expansion path would never get used.
-  - Use a more appropriate default format for FLAC decoders. This will now use ma_format_s16 when the FLAC is encoded as 16-bit.
-  - Update FLAC decoder.
-  - Update links to point to the new repository location (https://github.com/mackron/miniaudio).
-v0.10.17 - 2020-08-28
-  - Fix an error where the WAV codec is incorrectly excluded from the build depending on which compile time options are set.
-  - Fix a bug in ma_audio_buffer_read_pcm_frames() where it isn't returning the correct number of frames processed.
-  - Fix compilation error on Android.
-  - Core Audio: Fix a bug with full-duplex mode.
-  - Add ma_decoder_get_cursor_in_pcm_frames().
-  - Update WAV codec.
-v0.10.16 - 2020-08-14
-  - WASAPI: Fix a potential crash due to using an uninitialized variable.
-  - OpenSL: Enable runtime linking.
-  - OpenSL: Fix a multithreading bug when initializing and uninitializing multiple contexts at the same time.
-  - iOS: Improvements to device enumeration.
-  - Fix a crash in ma_data_source_read_pcm_frames() when the output frame count parameter is NULL.
-  - Fix a bug in ma_data_source_read_pcm_frames() where looping doesn't work.
-  - Fix some compilation warnings on Windows when both DirectSound and WinMM are disabled.
-  - Fix some compilation warnings when no decoders are enabled.
-  - Add ma_audio_buffer_get_available_frames().
-  - Add ma_decoder_get_available_frames().
-  - Add sample rate to ma_data_source_get_data_format().
-  - Change volume APIs to take 64-bit frame counts.
-  - Updates to documentation.
-v0.10.15 - 2020-07-15
-  - Fix a bug when converting bit-masked channel maps to miniaudio channel maps. This affects the WASAPI and OpenSL backends.
-v0.10.14 - 2020-07-14
-  - Fix compilation errors on Android.
-  - Fix compilation errors with -march=armv6.
-  - Updates to the documentation.
-v0.10.13 - 2020-07-11
-  - Fix some potential buffer overflow errors with channel maps when channel counts are greater than MA_MAX_CHANNELS.
-  - Fix compilation error on Emscripten.
-  - Silence some unused function warnings.
-  - Increase the default buffer size on the Web Audio backend. This fixes glitching issues on some browsers.
-  - Bring FLAC decoder up-to-date with dr_flac.
-  - Bring MP3 decoder up-to-date with dr_mp3.
-v0.10.12 - 2020-07-04
-  - Fix compilation errors on the iOS build.
-v0.10.11 - 2020-06-28
-  - Fix some bugs with device tracking on Core Audio.
-  - Updates to documentation.
-v0.10.10 - 2020-06-26
-  - Add include guard for the implementation section.
-  - Mark ma_device_sink_info_callback() as static.
-  - Fix compilation errors with MA_NO_DECODING and MA_NO_ENCODING.
-  - Fix compilation errors with MA_NO_DEVICE_IO
-v0.10.9 - 2020-06-24
-  - Amalgamation of dr_wav, dr_flac and dr_mp3. With this change, including the header section of these libraries before the implementation of miniaudio is no
-    longer required. Decoding of WAV, FLAC and MP3 should be supported seamlessly without any additional libraries. Decoders can be excluded from the build
-    with the following options:
-    - MA_NO_WAV
-    - MA_NO_FLAC
-    - MA_NO_MP3
-    If you get errors about multiple definitions you need to either enable the options above, move the implementation of dr_wav, dr_flac and/or dr_mp3 to before
-    the implementation of miniaudio, or update dr_wav, dr_flac and/or dr_mp3.
-  - Changes to the internal atomics library. This has been replaced with c89atomic.h which is embedded within this file.
-  - Fix a bug when a decoding backend reports configurations outside the limits of miniaudio's decoder abstraction.
-  - Fix the UWP build.
-  - Fix the Core Audio build.
-  - Fix the -std=c89 build on GCC.
-v0.10.8 - 2020-06-22
-  - Remove dependency on ma_context from mutexes.
-  - Change ma_data_source_read_pcm_frames() to return a result code and output the frames read as an output parameter.
-  - Change ma_data_source_seek_pcm_frames() to return a result code and output the frames seeked as an output parameter.
-  - Change ma_audio_buffer_unmap() to return MA_AT_END when the end has been reached. This should be considered successful.
-  - Change playback.pDeviceID and capture.pDeviceID to constant pointers in ma_device_config.
-  - Add support for initializing decoders from a virtual file system object. This is achieved via the ma_vfs API and allows the application to customize file
-    IO for the loading and reading of raw audio data. Passing in NULL for the VFS will use defaults. New APIs:
-    - ma_decoder_init_vfs()
-    - ma_decoder_init_vfs_wav()
-    - ma_decoder_init_vfs_flac()
-    - ma_decoder_init_vfs_mp3()
-    - ma_decoder_init_vfs_vorbis()
-    - ma_decoder_init_vfs_w()
-    - ma_decoder_init_vfs_wav_w()
-    - ma_decoder_init_vfs_flac_w()
-    - ma_decoder_init_vfs_mp3_w()
-    - ma_decoder_init_vfs_vorbis_w()
-  - Add support for memory mapping to ma_data_source.
-    - ma_data_source_map()
-    - ma_data_source_unmap()
-  - Add ma_offset_pcm_frames_ptr() and ma_offset_pcm_frames_const_ptr() which can be used for offsetting a pointer by a specified number of PCM frames.
-  - Add initial implementation of ma_yield() which is useful for spin locks which will be used in some upcoming work.
-  - Add documentation for log levels.
-  - The ma_event API has been made public in preparation for some uncoming work.
-  - Fix a bug in ma_decoder_seek_to_pcm_frame() where the internal sample rate is not being taken into account for determining the seek location.
-  - Fix some bugs with the linear resampler when dynamically changing the sample rate.
-  - Fix compilation errors with MA_NO_DEVICE_IO.
-  - Fix some warnings with GCC and -std=c89.
-  - Fix some formatting warnings with GCC and -Wall and -Wpedantic.
-  - Fix some warnings with VC6.
-  - Minor optimization to ma_copy_pcm_frames(). This is now a no-op when the input and output buffers are the same.
-v0.10.7 - 2020-05-25
-  - Fix a compilation error in the C++ build.
-  - Silence a warning.
-v0.10.6 - 2020-05-24
-  - Change ma_clip_samples_f32() and ma_clip_pcm_frames_f32() to take a 64-bit sample/frame count.
-  - Change ma_zero_pcm_frames() to clear to 128 for ma_format_u8.
-  - Add ma_silence_pcm_frames() which replaces ma_zero_pcm_frames(). ma_zero_pcm_frames() will be removed in version 0.11.
-  - Add support for u8, s24 and s32 formats to ma_channel_converter.
-  - Add compile-time and run-time version querying.
-    - MA_VERSION_MINOR
-    - MA_VERSION_MAJOR
-    - MA_VERSION_REVISION
-    - MA_VERSION_STRING
-    - ma_version()
-    - ma_version_string()
-  - Add ma_audio_buffer for reading raw audio data directly from memory.
-  - Fix a bug in shuffle mode in ma_channel_converter.
-  - Fix compilation errors in certain configurations for ALSA and PulseAudio.
-  - The data callback now initializes the output buffer to 128 when the playback sample format is ma_format_u8.
-v0.10.5 - 2020-05-05
-  - Change ma_zero_pcm_frames() to take a 64-bit frame count.
-  - Add ma_copy_pcm_frames().
-  - Add MA_NO_GENERATION build option to exclude the `ma_waveform` and `ma_noise` APIs from the build.
-  - Add support for formatted logging to the VC6 build.
-  - Fix a crash in the linear resampler when LPF order is 0.
-  - Fix compilation errors and warnings with older versions of Visual Studio.
-  - Minor documentation updates.
-v0.10.4 - 2020-04-12
-  - Fix a data conversion bug when converting from the client format to the native device format.
-v0.10.3 - 2020-04-07
-  - Bring up to date with breaking changes to dr_mp3.
-  - Remove MA_NO_STDIO. This was causing compilation errors and the maintenance cost versus practical benefit is no longer worthwhile.
-  - Fix a bug with data conversion where it was unnecessarily converting to s16 or f32 and then straight back to the original format.
-  - Fix compilation errors and warnings with Visual Studio 2005.
-  - ALSA: Disable ALSA's automatic data conversion by default and add configuration options to the device config:
-    - alsa.noAutoFormat
-    - alsa.noAutoChannels
-    - alsa.noAutoResample
-  - WASAPI: Add some overrun recovery for ma_device_type_capture devices.
-v0.10.2 - 2020-03-22
-  - Decorate some APIs with MA_API which were missed in the previous version.
-  - Fix a bug in ma_linear_resampler_set_rate() and ma_linear_resampler_set_rate_ratio().
-v0.10.1 - 2020-03-17
-  - Add MA_API decoration. This can be customized by defining it before including miniaudio.h.
-  - Fix a bug where opening a file would return a success code when in fact it failed.
-  - Fix compilation errors with Visual Studio 6 and 2003.
-  - Fix warnings on macOS.
-v0.10.0 - 2020-03-07
-  - API CHANGE: Refactor data conversion APIs
-    - ma_format_converter has been removed. Use ma_convert_pcm_frames_format() instead.
-    - ma_channel_router has been replaced with ma_channel_converter.
-    - ma_src has been replaced with ma_resampler
-    - ma_pcm_converter has been replaced with ma_data_converter
-  - API CHANGE: Add support for custom memory allocation callbacks. The following APIs have been updated to take an extra parameter for the allocation
-    callbacks:
-    - ma_malloc()
-    - ma_realloc()
-    - ma_free()
-    - ma_aligned_malloc()
-    - ma_aligned_free()
-    - ma_rb_init() / ma_rb_init_ex()
-    - ma_pcm_rb_init() / ma_pcm_rb_init_ex()
-  - API CHANGE: Simplify latency specification in device configurations. The bufferSizeInFrames and bufferSizeInMilliseconds parameters have been replaced with
-    periodSizeInFrames and periodSizeInMilliseconds respectively. The previous variables defined the size of the entire buffer, whereas the new ones define the
-    size of a period. The following APIs have been removed since they are no longer relevant:
-    - ma_get_default_buffer_size_in_milliseconds()
-    - ma_get_default_buffer_size_in_frames()
-  - API CHANGE: ma_device_set_stop_callback() has been removed. If you require a stop callback, you must now set it via the device config just like the data
-    callback.
-  - API CHANGE: The ma_sine_wave API has been replaced with ma_waveform. The following APIs have been removed:
-    - ma_sine_wave_init()
-    - ma_sine_wave_read_f32()
-    - ma_sine_wave_read_f32_ex()
-  - API CHANGE: ma_convert_frames() has been updated to take an extra parameter which is the size of the output buffer in PCM frames. Parameters have also been
-    reordered.
-  - API CHANGE: ma_convert_frames_ex() has been changed to take a pointer to a ma_data_converter_config object to specify the input and output formats to
-    convert between.
-  - API CHANGE: ma_calculate_frame_count_after_src() has been renamed to ma_calculate_frame_count_after_resampling().
-  - Add support for the following filters:
-    - Biquad (ma_biquad)
-    - First order low-pass (ma_lpf1)
-    - Second order low-pass (ma_lpf2)
-    - Low-pass with configurable order (ma_lpf)
-    - First order high-pass (ma_hpf1)
-    - Second order high-pass (ma_hpf2)
-    - High-pass with configurable order (ma_hpf)
-    - Second order band-pass (ma_bpf2)
-    - Band-pass with configurable order (ma_bpf)
-    - Second order peaking EQ (ma_peak2)
-    - Second order notching (ma_notch2)
-    - Second order low shelf (ma_loshelf2)
-    - Second order high shelf (ma_hishelf2)
-  - Add waveform generation API (ma_waveform) with support for the following:
-    - Sine
-    - Square
-    - Triangle
-    - Sawtooth
-  - Add noise generation API (ma_noise) with support for the following:
-    - White
-    - Pink
-    - Brownian
-  - Add encoding API (ma_encoder). This only supports outputting to WAV files via dr_wav.
-  - Add ma_result_description() which is used to retrieve a human readable description of a given result code.
-  - Result codes have been changed. Binding maintainers will need to update their result code constants.
-  - More meaningful result codes are now returned when a file fails to open.
-  - Internal functions have all been made static where possible.
-  - Fix potential crash when ma_device object's are not aligned to MA_SIMD_ALIGNMENT.
-  - Fix a bug in ma_decoder_get_length_in_pcm_frames() where it was returning the length based on the internal sample rate rather than the output sample rate.
-  - Fix bugs in some backends where the device is not drained properly in ma_device_stop().
-  - Improvements to documentation.
-v0.9.10 - 2020-01-15
-  - Fix compilation errors due to #if/#endif mismatches.
-  - WASAPI: Fix a bug where automatic stream routing is being performed for devices that are initialized with an explicit device ID.
-  - iOS: Fix a crash on device uninitialization.
-v0.9.9 - 2020-01-09
-  - Fix compilation errors with MinGW.
-  - Fix compilation errors when compiling on Apple platforms.
-  - WASAPI: Add support for disabling hardware offloading.
-  - WASAPI: Add support for disabling automatic stream routing.
-  - Core Audio: Fix bugs in the case where the internal device uses deinterleaved buffers.
-  - Core Audio: Add support for controlling the session category (AVAudioSessionCategory) and options (AVAudioSessionCategoryOptions).
-  - JACK: Fix bug where incorrect ports are connected.
-v0.9.8 - 2019-10-07
-  - WASAPI: Fix a potential deadlock when starting a full-duplex device.
-  - WASAPI: Enable automatic resampling by default. Disable with config.wasapi.noAutoConvertSRC.
-  - Core Audio: Fix bugs with automatic stream routing.
-  - Add support for controlling whether or not the content of the output buffer passed in to the data callback is pre-initialized
-    to zero. By default it will be initialized to zero, but this can be changed by setting noPreZeroedOutputBuffer in the device
-    config. Setting noPreZeroedOutputBuffer to true will leave the contents undefined.
-  - Add support for clipping samples after the data callback has returned. This only applies when the playback sample format is
-    configured as ma_format_f32. If you are doing clipping yourself, you can disable this overhead by setting noClip to true in
-    the device config.
-  - Add support for master volume control for devices.
-    - Use ma_device_set_master_volume() to set the volume to a factor between 0 and 1, where 0 is silence and 1 is full volume.
-    - Use ma_device_set_master_volume_db() to set the volume in decibels where 0 is full volume and < 0 reduces the volume.
-  - Fix warnings emitted by GCC when `__inline__` is undefined or defined as nothing.
-v0.9.7 - 2019-08-28
-  - Add support for loopback mode (WASAPI only).
-    - To use this, set the device type to ma_device_type_loopback, and then fill out the capture section of the device config.
-    - If you need to capture from a specific output device, set the capture device ID to that of a playback device.
-  - Fix a crash when an error is posted in ma_device_init().
-  - Fix a compilation error when compiling for ARM architectures.
-  - Fix a bug with the audio(4) backend where the device is incorrectly being opened in non-blocking mode.
-  - Fix memory leaks in the Core Audio backend.
-  - Minor refactoring to the WinMM, ALSA, PulseAudio, OSS, audio(4), sndio and null backends.
-v0.9.6 - 2019-08-04
-  - Add support for loading decoders using a wchar_t string for file paths.
-  - Don't trigger an assert when ma_device_start() is called on a device that is already started. This will now log a warning
-    and return MA_INVALID_OPERATION. The same applies for ma_device_stop().
-  - Try fixing an issue with PulseAudio taking a long time to start playback.
-  - Fix a bug in ma_convert_frames() and ma_convert_frames_ex().
-  - Fix memory leaks in the WASAPI backend.
-  - Fix a compilation error with Visual Studio 2010.
-v0.9.5 - 2019-05-21
-  - Add logging to ma_dlopen() and ma_dlsym().
-  - Add ma_decoder_get_length_in_pcm_frames().
-  - Fix a bug with capture on the OpenSL|ES backend.
-  - Fix a bug with the ALSA backend where a device would not restart after being stopped.
-v0.9.4 - 2019-05-06
-  - Add support for C89. With this change, miniaudio should compile clean with GCC/Clang with "-std=c89 -ansi -pedantic" and
-    Microsoft compilers back to VC6. Other compilers should also work, but have not been tested.
-v0.9.3 - 2019-04-19
-  - Fix compiler errors on GCC when compiling with -std=c99.
-v0.9.2 - 2019-04-08
-  - Add support for per-context user data.
-  - Fix a potential bug with context configs.
-  - Fix some bugs with PulseAudio.
-v0.9.1 - 2019-03-17
-  - Fix a bug where the output buffer is not getting zeroed out before calling the data callback. This happens when
-    the device is running in passthrough mode (not doing any data conversion).
-  - Fix an issue where the data callback is getting called too frequently on the WASAPI and DirectSound backends.
-  - Fix error on the UWP build.
-  - Fix a build error on Apple platforms.
-v0.9 - 2019-03-06
-  - Rebranded to "miniaudio". All namespaces have been renamed from "mal" to "ma".
-  - API CHANGE: ma_device_init() and ma_device_config_init() have changed significantly:
-    - The device type, device ID and user data pointer have moved from ma_device_init() to the config.
-    - All variations of ma_device_config_init_*() have been removed in favor of just ma_device_config_init().
-    - ma_device_config_init() now takes only one parameter which is the device type. All other properties need
-      to be set on the returned object directly.
-    - The onDataCallback and onStopCallback members of ma_device_config have been renamed to "dataCallback"
-      and "stopCallback".
-    - The ID of the physical device is now split into two: one for the playback device and the other for the
-      capture device. This is required for full-duplex. These are named "pPlaybackDeviceID" and "pCaptureDeviceID".
-  - API CHANGE: The data callback has changed. It now uses a unified callback for all device types rather than
-    being separate for each. It now takes two pointers - one containing input data and the other output data. This
-    design in required for full-duplex. The return value is now void instead of the number of frames written. The
-    new callback looks like the following:
-        void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
-  - API CHANGE: Remove the log callback parameter from ma_context_config_init(). With this change,
-    ma_context_config_init() now takes no parameters and the log callback is set via the structure directly. The
-    new policy for config initialization is that only mandatory settings are passed in to *_config_init(). The
-    "onLog" member of ma_context_config has been renamed to "logCallback".
-  - API CHANGE: Remove ma_device_get_buffer_size_in_bytes().
-  - API CHANGE: Rename decoding APIs to "pcm_frames" convention.
-    - mal_decoder_read()          -> ma_decoder_read_pcm_frames()
-    - mal_decoder_seek_to_frame() -> ma_decoder_seek_to_pcm_frame()
-  - API CHANGE: Rename sine wave reading APIs to f32 convention.
-    - mal_sine_wave_read()    -> ma_sine_wave_read_f32()
-    - mal_sine_wave_read_ex() -> ma_sine_wave_read_f32_ex()
-  - API CHANGE: Remove some deprecated APIs
-    - mal_device_set_recv_callback()
-    - mal_device_set_send_callback()
-    - mal_src_set_input_sample_rate()
-    - mal_src_set_output_sample_rate()
-  - API CHANGE: Add log level to the log callback. New signature:
-    - void on_log(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message)
-  - API CHANGE: Changes to result codes. Constants have changed and unused codes have been removed. If you're
-    a binding mainainer you will need to update your result code constants.
-  - API CHANGE: Change the order of the ma_backend enums to priority order. If you are a binding maintainer, you
-    will need to update.
-  - API CHANGE: Rename mal_dsp to ma_pcm_converter. All functions have been renamed from mal_dsp_*() to
-    ma_pcm_converter_*(). All structures have been renamed from mal_dsp* to ma_pcm_converter*.
-  - API CHANGE: Reorder parameters of ma_decoder_read_pcm_frames() to be consistent with the new parameter order scheme.
-  - The resampling algorithm has been changed from sinc to linear. The rationale for this is that the sinc implementation
-    is too inefficient right now. This will hopefully be improved at a later date.
-  - Device initialization will no longer fall back to shared mode if exclusive mode is requested but is unusable.
-    With this change, if you request an device in exclusive mode, but exclusive mode is not supported, it will not
-    automatically fall back to shared mode. The client will need to reinitialize the device in shared mode if that's
-    what they want.
-  - Add ring buffer API. This is ma_rb and ma_pcm_rb, the difference being that ma_rb operates on bytes and
-    ma_pcm_rb operates on PCM frames.
-  - Add Web Audio backend. This is used when compiling with Emscripten. The SDL backend, which was previously
-    used for web support, will be removed in a future version.
-  - Add AAudio backend (Android Audio). This is the new priority backend for Android. Support for AAudio starts
-    with Android 8. OpenSL|ES is used as a fallback for older versions of Android.
-  - Remove OpenAL and SDL backends.
-  - Fix a possible deadlock when rapidly stopping the device after it has started.
-  - Update documentation.
-  - Change licensing to a choice of public domain _or_ MIT-0 (No Attribution).
-v0.8.14 - 2018-12-16
-  - Core Audio: Fix a bug where the device state is not set correctly after stopping.
-  - Add support for custom weights to the channel router.
-  - Update decoders to use updated APIs in dr_flac, dr_mp3 and dr_wav.
-v0.8.13 - 2018-12-04
-  - Core Audio: Fix a bug with channel mapping.
-  - Fix a bug with channel routing where the back/left and back/right channels have the wrong weight.
-v0.8.12 - 2018-11-27
-  - Drop support for SDL 1.2. The Emscripten build now requires "-s USE_SDL=2".
-  - Fix a linking error with ALSA.
-  - Fix a bug on iOS where the device name is not set correctly.
-v0.8.11 - 2018-11-21
-  - iOS bug fixes.
-  - Minor tweaks to PulseAudio.
-v0.8.10 - 2018-10-21
-  - Core Audio: Fix a hang when uninitializing a device.
-  - Fix a bug where an incorrect value is returned from mal_device_stop().
-v0.8.9 - 2018-09-28
-  - Fix a bug with the SDL backend where device initialization fails.
-v0.8.8 - 2018-09-14
-  - Fix Linux build with the ALSA backend.
-  - Minor documentation fix.
-v0.8.7 - 2018-09-12
-  - Fix a bug with UWP detection.
-v0.8.6 - 2018-08-26
-  - Automatically switch the internal device when the default device is unplugged. Note that this is still in the
-    early stages and not all backends handle this the same way. As of this version, this will not detect a default
-    device switch when changed from the operating system's audio preferences (unless the backend itself handles
-    this automatically). This is not supported in exclusive mode.
-  - WASAPI and Core Audio: Add support for stream routing. When the application is using a default device and the
-    user switches the default device via the operating system's audio preferences, miniaudio will automatically switch
-    the internal device to the new default. This is not supported in exclusive mode.
-  - WASAPI: Add support for hardware offloading via IAudioClient2. Only supported on Windows 8 and newer.
-  - WASAPI: Add support for low-latency shared mode via IAudioClient3. Only supported on Windows 10 and newer.
-  - Add support for compiling the UWP build as C.
-  - mal_device_set_recv_callback() and mal_device_set_send_callback() have been deprecated. You must now set this
-    when the device is initialized with mal_device_init*(). These will be removed in version 0.9.0.
-v0.8.5 - 2018-08-12
-  - Add support for specifying the size of a device's buffer in milliseconds. You can still set the buffer size in
-    frames if that suits you. When bufferSizeInFrames is 0, bufferSizeInMilliseconds will be used. If both are non-0
-    then bufferSizeInFrames will take priority. If both are set to 0 the default buffer size is used.
-  - Add support for the audio(4) backend to OpenBSD.
-  - Fix a bug with the ALSA backend that was causing problems on Raspberry Pi. This significantly improves the
-    Raspberry Pi experience.
-  - Fix a bug where an incorrect number of samples is returned from sinc resampling.
-  - Add support for setting the value to be passed to internal calls to CoInitializeEx().
-  - WASAPI and WinMM: Stop the device when it is unplugged.
-v0.8.4 - 2018-08-06
-  - Add sndio backend for OpenBSD.
-  - Add audio(4) backend for NetBSD.
-  - Drop support for the OSS backend on everything except FreeBSD and DragonFly BSD.
-  - Formats are now native-endian (were previously little-endian).
-  - Mark some APIs as deprecated:
-    - mal_src_set_input_sample_rate() and mal_src_set_output_sample_rate() are replaced with mal_src_set_sample_rate().
-    - mal_dsp_set_input_sample_rate() and mal_dsp_set_output_sample_rate() are replaced with mal_dsp_set_sample_rate().
-  - Fix a bug when capturing using the WASAPI backend.
-  - Fix some aliasing issues with resampling, specifically when increasing the sample rate.
-  - Fix warnings.
-v0.8.3 - 2018-07-15
-  - Fix a crackling bug when resampling in capture mode.
-  - Core Audio: Fix a bug where capture does not work.
-  - ALSA: Fix a bug where the worker thread can get stuck in an infinite loop.
-  - PulseAudio: Fix a bug where mal_context_init() succeeds when PulseAudio is unusable.
-  - JACK: Fix a bug where mal_context_init() succeeds when JACK is unusable.
-v0.8.2 - 2018-07-07
-  - Fix a bug on macOS with Core Audio where the internal callback is not called.
-v0.8.1 - 2018-07-06
-  - Fix compilation errors and warnings.
-v0.8 - 2018-07-05
-  - Changed MAL_IMPLEMENTATION to MINI_AL_IMPLEMENTATION for consistency with other libraries. The old
-    way is still supported for now, but you should update as it may be removed in the future.
-  - API CHANGE: Replace device enumeration APIs. mal_enumerate_devices() has been replaced with
-    mal_context_get_devices(). An additional low-level device enumration API has been introduced called
-    mal_context_enumerate_devices() which uses a callback to report devices.
-  - API CHANGE: Rename mal_get_sample_size_in_bytes() to mal_get_bytes_per_sample() and add
-    mal_get_bytes_per_frame().
-  - API CHANGE: Replace mal_device_config.preferExclusiveMode with mal_device_config.shareMode.
-    - This new config can be set to mal_share_mode_shared (default) or mal_share_mode_exclusive.
-  - API CHANGE: Remove excludeNullDevice from mal_context_config.alsa.
-  - API CHANGE: Rename MAL_MAX_SAMPLE_SIZE_IN_BYTES to MAL_MAX_PCM_SAMPLE_SIZE_IN_BYTES.
-  - API CHANGE: Change the default channel mapping to the standard Microsoft mapping.
-  - API CHANGE: Remove backend-specific result codes.
-  - API CHANGE: Changes to the format conversion APIs (mal_pcm_f32_to_s16(), etc.)
-  - Add support for Core Audio (Apple).
-  - Add support for PulseAudio.
-    - This is the highest priority backend on Linux (higher priority than ALSA) since it is commonly
-      installed by default on many of the popular distros and offer's more seamless integration on
-      platforms where PulseAudio is used. In addition, if PulseAudio is installed and running (which
-      is extremely common), it's better to just use PulseAudio directly rather than going through the
-      "pulse" ALSA plugin (which is what the "default" ALSA device is likely set to).
-  - Add support for JACK.
-  - Remove dependency on asound.h for the ALSA backend. This means the ALSA development packages are no
-    longer required to build miniaudio.
-  - Remove dependency on dsound.h for the DirectSound backend. This fixes build issues with some
-    distributions of MinGW.
-  - Remove dependency on audioclient.h for the WASAPI backend. This fixes build issues with some
-    distributions of MinGW.
-  - Add support for dithering to format conversion.
-  - Add support for configuring the priority of the worker thread.
-  - Add a sine wave generator.
-  - Improve efficiency of sample rate conversion.
-  - Introduce the notion of standard channel maps. Use mal_get_standard_channel_map().
-  - Introduce the notion of default device configurations. A default config uses the same configuration
-    as the backend's internal device, and as such results in a pass-through data transmission pipeline.
-  - Add support for passing in NULL for the device config in mal_device_init(), which uses a default
-    config. This requires manually calling mal_device_set_send/recv_callback().
-  - Add support for decoding from raw PCM data (mal_decoder_init_raw(), etc.)
-  - Make mal_device_init_ex() more robust.
-  - Make some APIs more const-correct.
-  - Fix errors with SDL detection on Apple platforms.
-  - Fix errors with OpenAL detection.
-  - Fix some memory leaks.
-  - Fix a bug with opening decoders from memory.
-  - Early work on SSE2, AVX2 and NEON optimizations.
-  - Miscellaneous bug fixes.
-  - Documentation updates.
-v0.7 - 2018-02-25
-  - API CHANGE: Change mal_src_read_frames() and mal_dsp_read_frames() to use 64-bit sample counts.
-  - Add decoder APIs for loading WAV, FLAC, Vorbis and MP3 files.
-  - Allow opening of devices without a context.
-    - In this case the context is created and managed internally by the device.
-  - Change the default channel mapping to the same as that used by FLAC.
-  - Fix build errors with macOS.
-v0.6c - 2018-02-12
-  - Fix build errors with BSD/OSS.
-v0.6b - 2018-02-03
-  - Fix some warnings when compiling with Visual C++.
-v0.6a - 2018-01-26
-  - Fix errors with channel mixing when increasing the channel count.
-  - Improvements to the build system for the OpenAL backend.
-  - Documentation fixes.
-v0.6 - 2017-12-08
-  - API CHANGE: Expose and improve mutex APIs. If you were using the mutex APIs before this version you'll
-    need to update.
-  - API CHANGE: SRC and DSP callbacks now take a pointer to a mal_src and mal_dsp object respectively.
-  - API CHANGE: Improvements to event and thread APIs. These changes make these APIs more consistent.
-  - Add support for SDL and Emscripten.
-  - Simplify the build system further for when development packages for various backends are not installed.
-    With this change, when the compiler supports __has_include, backends without the relevant development
-    packages installed will be ignored. This fixes the build for old versions of MinGW.
-  - Fixes to the Android build.
-  - Add mal_convert_frames(). This is a high-level helper API for performing a one-time, bulk conversion of
-    audio data to a different format.
-  - Improvements to f32 -> u8/s16/s24/s32 conversion routines.
-  - Fix a bug where the wrong value is returned from mal_device_start() for the OpenSL backend.
-  - Fixes and improvements for Raspberry Pi.
-  - Warning fixes.
-v0.5 - 2017-11-11
-  - API CHANGE: The mal_context_init() function now takes a pointer to a mal_context_config object for
-    configuring the context. The works in the same kind of way as the device config. The rationale for this
-    change is to give applications better control over context-level properties, add support for backend-
-    specific configurations, and support extensibility without breaking the API.
-  - API CHANGE: The alsa.preferPlugHW device config variable has been removed since it's not really useful for
-    anything anymore.
-  - ALSA: By default, device enumeration will now only enumerate over unique card/device pairs. Applications
-    can enable verbose device enumeration by setting the alsa.useVerboseDeviceEnumeration context config
-    variable.
-  - ALSA: When opening a device in shared mode (the default), the dmix/dsnoop plugin will be prioritized. If
-    this fails it will fall back to the hw plugin. With this change the preferExclusiveMode config is now
-    honored. Note that this does not happen when alsa.useVerboseDeviceEnumeration is set to true (see above)
-    which is by design.
-  - ALSA: Add support for excluding the "null" device using the alsa.excludeNullDevice context config variable.
-  - ALSA: Fix a bug with channel mapping which causes an assertion to fail.
-  - Fix errors with enumeration when pInfo is set to NULL.
-  - OSS: Fix a bug when starting a device when the client sends 0 samples for the initial buffer fill.
-v0.4 - 2017-11-05
-  - API CHANGE: The log callback is now per-context rather than per-device and as is thus now passed to
-    mal_context_init(). The rationale for this change is that it allows applications to capture diagnostic
-    messages at the context level. Previously this was only available at the device level.
-  - API CHANGE: The device config passed to mal_device_init() is now const.
-  - Added support for OSS which enables support on BSD platforms.
-  - Added support for WinMM (waveOut/waveIn).
-  - Added support for UWP (Universal Windows Platform) applications. Currently C++ only.
-  - Added support for exclusive mode for selected backends. Currently supported on WASAPI.
-  - POSIX builds no longer require explicit linking to libpthread (-lpthread).
-  - ALSA: Explicit linking to libasound (-lasound) is no longer required.
-  - ALSA: Latency improvements.
-  - ALSA: Use MMAP mode where available. This can be disabled with the alsa.noMMap config.
-  - ALSA: Use "hw" devices instead of "plughw" devices by default. This can be disabled with the
-    alsa.preferPlugHW config.
-  - WASAPI is now the highest priority backend on Windows platforms.
-  - Fixed an error with sample rate conversion which was causing crackling when capturing.
-  - Improved error handling.
-  - Improved compiler support.
-  - Miscellaneous bug fixes.
-v0.3 - 2017-06-19
-  - API CHANGE: Introduced the notion of a context. The context is the highest level object and is required for
-    enumerating and creating devices. Now, applications must first create a context, and then use that to
-    enumerate and create devices. The reason for this change is to ensure device enumeration and creation is
-    tied to the same backend. In addition, some backends are better suited to this design.
-  - API CHANGE: Removed the rewinding APIs because they're too inconsistent across the different backends, hard
-    to test and maintain, and just generally unreliable.
-  - Added helper APIs for initializing mal_device_config objects.
-  - Null Backend: Fixed a crash when recording.
-  - Fixed build for UWP.
-  - Added support for f32 formats to the OpenSL|ES backend.
-  - Added initial implementation of the WASAPI backend.
-  - Added initial implementation of the OpenAL backend.
-  - Added support for low quality linear sample rate conversion.
-  - Added early support for basic channel mapping.
-v0.2 - 2016-10-28
-  - API CHANGE: Add user data pointer as the last parameter to mal_device_init(). The rationale for this
-    change is to ensure the logging callback has access to the user data during initialization.
-  - API CHANGE: Have device configuration properties be passed to mal_device_init() via a structure. Rationale:
-    1) The number of parameters is just getting too much.
-    2) It makes it a bit easier to add new configuration properties in the future. In particular, there's a
-       chance there will be support added for backend-specific properties.
-  - Dropped support for f64, A-law and Mu-law formats since they just aren't common enough to justify the
-    added maintenance cost.
-  - DirectSound: Increased the default buffer size for capture devices.
-  - Added initial implementation of the OpenSL|ES backend.
-v0.1 - 2016-10-21
-  - Initial versioned release.
-*/
 /*
 This software is available as a choice of the following licenses. Choose