Redesigned to support disabling features on compilation with `-DSUPPORT_FEATURE=0`
REMOVED: `SUPPORT_DEFAULT_FONT`, always supported
REMOVED: `SUPPORT_IMAGE_MANIPULATION `, always supported
REMOVED: `SUPPORT_TEXT_MANIPULATION`, always supported
REDESIGNED: `SUPPORT_FONT_ATLAS_WHITE_REC` to `FONT_ATLAS_CORNER_REC_SIZE`
REVIEWED: Config values (other than 0-1) are already defined on respective modules
Other config tweaks here and there
* ANDROID: Fix broken LoadMusicStream due to missing fopen macro override in raudio
* ANDROID: Add missing forward declaration before fopen macro override in raudio
* renamed SHADER_LOC_VERTEX_INSTANCETRANSFORMS to SHADER_LOC_VERTEX_INSTANCETRANSFORM
* rlparser: update raylib_api.* by CI
---------
Co-authored-by: github-actions[bot] <41898282+github-actions[bot]@users.noreply.github.com>
* win32 clipbaord: fix for BI_ALPHABITFIELDS narrow support
* Define BI_ALPHABITFIELDS even if wingdi headers are already included
since BI_ALPHABITFIELDS is not always defined there
REVIEWED: Reorganized structures for a clearer distinction between "skeleton", "skin" and "skinning" data
ADDED: New structures: `ModelSkeleton`, `ModelAnimPose` (alias `Transform*`)
ADDED: Runtime data `currentPose` and `boneMatrices` to `Model` structure
ADDED: Support animation frames-blending, for timing control
ADDED: Support animations blending, between two animations
REVIEWED: All models animation loading functions
ADDED: `UpdateModelAnimationEx()` for two animations blending
REMOVED: `UpdateModelAnimationBones*()`, simplified API
REVIEWED: Shader attributes/uniforms names for animations, for consistency
REVIEWED: Multiple tweaks on animations loading for consistency between formats
ADDED: example: `models_animation_timing`
ADDED: example: `models_animation_blending`
REVIEWED: example: `models_animation_gpu_skinning`
REVIEWED: example: `models_animation_blend_custom`
REVIEWED: All animated models loading examples
This is happening because the processing function keeps reading audio
data from the AudioBuffer even after it has been marked as stopped.
There is also an error in ReadAudioBufferFramesInInternalFormat() where
if it is called on a stopped sound, it'll still return audio frames.
This has also been addressed with this commit.
* Audio: Stop setting capture config options.
Since the device is being configured as a playback device, all capture
config options are unused and therefore need to not be set.
* Audio: Stop pre-silencing the miniaudio output buffer.
raylib already manually silences the output buffer prior to mixing so
there is no reason to have miniaudio also do it. It can therefore be
disabled via the device config to make data processing slightly more
efficient.
* Audio: Stop forcing fixed sized processing callbacks.
There is no requirement for raylib to have guaranteed fixed sized
audio processing. By disabling it, audio processing can be made more
efficient by not having to run the data through an internal intermediary
buffer.
* Audio: Make the period size (latency) configurable.
The default period size is 10ms, but this is inappropriate for certain
platforms so it is useful to be able to allow those platforms to
configure the period size as required.
* Audio: Fix documentation for pan.
The pan if -1..1, not 0..1.
* Audio: Remove use of ma_data_converter_get_required_input_frame_count().
This function is being removed from miniaudio. To make this work with
the current architecture of raylib it requires the use of a cache.
This commit implements a generic solution that works across all
AudioBuffer types (static, streams and callback based), but the static
case could be optimized to avoid the cache by incorporating the
functionality of ReadAudioBufferFramesInInternalFormat() into
ReadAudioBufferFramesInMixingFormat(). It would be unpractical to avoid
the cache with streams and callback-based AudioBuffers however so this
commit sticks with a generic solution.
* Audio: Correct usage of miniaudio's dynamic rate adjustment.
This affects pitch shifting. The output rate is being modified with
ma_data_converter_set_rate(), but then that value is being used in the
computation of the output rate the next time SetAudioBufferPitch() which
results in a cascade. The correct way to do this is to use an anchored
output rate as the basis for the calculation after pitch shifting. In
this case, it's the device's sample rate that acts as the anchor.
* Audio: Optimize memory usage for data conversion.
This reduces the per-AudioBuffer conversion cache from 256 PCM frames
down to 8.
