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authorBorislav Stanimirov <b.stanimirov@abv.bg>2018-11-30 01:46:50 +0200
committerBorislav Stanimirov <b.stanimirov@abv.bg>2018-11-30 01:51:11 +0200
commita6a63322a841b3fd4406e70fb56b2b07cea3f417 (patch)
tree1dfee8c375fb844c967b25f7a38b861c44dd517c /sokol_audio.h
parent487822d82ca79dba7b67718d962e1ba6beef01b2 (diff)
removed trailing spaces
Diffstat (limited to 'sokol_audio.h')
-rw-r--r--sokol_audio.h40
1 files changed, 20 insertions, 20 deletions
diff --git a/sokol_audio.h b/sokol_audio.h
index 2d5aaf0b..3a390594 100644
--- a/sokol_audio.h
+++ b/sokol_audio.h
@@ -4,7 +4,7 @@
Do this:
#define SOKOL_IMPL
- before you include this file in *one* C or C++ file to create the
+ before you include this file in *one* C or C++ file to create the
implementation.
Optionally provide the following defines with your own implementations:
@@ -43,8 +43,8 @@
The callback model is preferred because it is the most direct way to
feed sample data into the audio backends and also has less moving parts
- (there is no ring buffer between your code and the audio backend).
-
+ (there is no ring buffer between your code and the audio backend).
+
Sometimes it is not possible to generate the audio stream directly in a
callback function running in a separate thread, for such cases Sokol Audio
provides the push-model as a convenience.
@@ -72,7 +72,7 @@
a separate thread, on WebAudio, this is called per-frame in the
browser thread.
- - channel:
+ - channel:
A discrete track of audio data, currently 1-channel (mono) and
2-channel (stereo) is supported and tested.
@@ -204,7 +204,7 @@
calling saudio_setup().
To provide sample data with the push model, call the saudio_push()
- function at regular intervals (for instance once per frame). You can
+ function at regular intervals (for instance once per frame). You can
call the saudio_expect() function to ask Sokol Audio how much room is
in the ring buffer, but if you provide a continuous stream of data
at the right sample rate, saudio_expect() isn't required (it's a simple
@@ -213,7 +213,7 @@
With saudio_push() you may need to maintain your own intermediate sample
buffer, since pushing individual sample values isn't very efficient.
- The following example is from the MOD player sample in
+ The following example is from the MOD player sample in
sokol-samples (https://github.com/floooh/sokol-samples):
const int num_frames = saudio_expect();
@@ -257,7 +257,7 @@
The WebAudio backend is automatically selected when compiling for
emscripten (__EMSCRIPTEN__ define exists).
-
+
https://developers.google.com/web/updates/2017/12/audio-worklet
https://developers.google.com/web/updates/2018/06/audio-worklet-design-pattern
@@ -402,7 +402,7 @@ SOKOL_API_DECL int saudio_push(const float* frames, int num_frames);
#define SOKOL_FREE(p) free(p)
#endif
#ifndef SOKOL_LOG
- #ifdef SOKOL_DEBUG
+ #ifdef SOKOL_DEBUG
#include <stdio.h>
#define SOKOL_LOG(s) { SOKOL_ASSERT(s); puts(s); }
#else
@@ -745,7 +745,7 @@ _SOKOL_PRIVATE bool _saudio_backend_init(void) {
/* init or modify actual playback parameters */
_saudio.bytes_per_frame = fmt.mBytesPerFrame;
-
+
/* ...and start playback */
res = AudioQueueStart(_saudio_ca_audio_queue, NULL);
SOKOL_ASSERT(0 == res);
@@ -913,7 +913,7 @@ typedef struct {
} _saudio_wasapi_state;
static _saudio_wasapi_state _saudio_wasapi;
-/* fill intermediate buffer with new data and reset buffer_pos */
+/* fill intermediate buffer with new data and reset buffer_pos */
_SOKOL_PRIVATE void _saudio_wasapi_fill_buffer(void) {
if (_saudio.stream_cb) {
_saudio.stream_cb(_saudio_wasapi.thread.src_buffer, _saudio_wasapi.thread.src_buffer_frames, _saudio.num_channels);
@@ -1011,25 +1011,25 @@ _SOKOL_PRIVATE bool _saudio_backend_init(void) {
SOKOL_LOG("sokol_audio wasapi: failed to create buffer_end_event");
goto error;
}
- if (FAILED(CoCreateInstance(&_saudio_CLSID_IMMDeviceEnumerator,
- 0, CLSCTX_ALL,
- &_saudio_IID_IMMDeviceEnumerator,
- (void**)&_saudio_wasapi.device_enumerator)))
+ if (FAILED(CoCreateInstance(&_saudio_CLSID_IMMDeviceEnumerator,
+ 0, CLSCTX_ALL,
+ &_saudio_IID_IMMDeviceEnumerator,
+ (void**)&_saudio_wasapi.device_enumerator)))
{
SOKOL_LOG("sokol_audio wasapi: failed to create device enumerator");
goto error;
}
if (FAILED(IMMDeviceEnumerator_GetDefaultAudioEndpoint(_saudio_wasapi.device_enumerator,
- eRender, eConsole,
- &_saudio_wasapi.device)))
+ eRender, eConsole,
+ &_saudio_wasapi.device)))
{
SOKOL_LOG("sokol_audio wasapi: GetDefaultAudioEndPoint failed");
goto error;
- }
+ }
if (FAILED(IMMDevice_Activate(_saudio_wasapi.device,
&_saudio_IID_IAudioClient,
CLSCTX_ALL, 0,
- (void**)&_saudio_wasapi.audio_client)))
+ (void**)&_saudio_wasapi.audio_client)))
{
SOKOL_LOG("sokol_audio wasapi: device activate failed");
goto error;
@@ -1056,7 +1056,7 @@ _SOKOL_PRIVATE bool _saudio_backend_init(void) {
SOKOL_LOG("sokol_audio wasapi: audio client get buffer size failed");
goto error;
}
- if (FAILED(IAudioClient_GetService(_saudio_wasapi.audio_client,
+ if (FAILED(IAudioClient_GetService(_saudio_wasapi.audio_client,
&_saudio_IID_IAudioRenderClient,
(void**)&_saudio_wasapi.render_client)))
{
@@ -1067,7 +1067,7 @@ _SOKOL_PRIVATE bool _saudio_backend_init(void) {
SOKOL_LOG("sokol_audio wasapi: audio client SetEventHandle failed");
goto error;
}
- _saudio_wasapi.si16_bytes_per_frame = _saudio.num_channels * sizeof(int16_t);
+ _saudio_wasapi.si16_bytes_per_frame = _saudio.num_channels * sizeof(int16_t);
_saudio.bytes_per_frame = _saudio.num_channels * sizeof(float);
_saudio_wasapi.thread.src_buffer_frames = _saudio.buffer_frames;
_saudio_wasapi.thread.src_buffer_byte_size = _saudio_wasapi.thread.src_buffer_frames * _saudio.bytes_per_frame;