Added SDL_AUDIO_BYTESIZE

include/SDL3/SDL_audio.h

@@ -84,6 +84,7 @@ typedef Uint16 SDL_AudioFormat;
 #define SDL_AUDIO_MASK_BIG_ENDIAN    (1<<12)
 #define SDL_AUDIO_MASK_SIGNED        (1<<15)
 #define SDL_AUDIO_BITSIZE(x)         ((x) & SDL_AUDIO_MASK_BITSIZE)
+#define SDL_AUDIO_BYTESIZE(x)        (SDL_AUDIO_BITSIZE(x) / 8)
 #define SDL_AUDIO_ISFLOAT(x)         ((x) & SDL_AUDIO_MASK_FLOAT)
 #define SDL_AUDIO_ISBIGENDIAN(x)     ((x) & SDL_AUDIO_MASK_BIG_ENDIAN)
 #define SDL_AUDIO_ISLITTLEENDIAN(x)  (!SDL_AUDIO_ISBIGENDIAN(x))

src/audio/SDL_audio.c

@@ -766,7 +766,7 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device)
             case MIXSTRATEGY_MIX: {
                 //SDL_Log("MIX STRATEGY: MIX");
                 float *mix_buffer = (float *) ((device->spec.format == SDL_AUDIO_F32) ? device_buffer : device->mix_buffer);
-                const int needed_samples = buffer_size / (SDL_AUDIO_BITSIZE(device->spec.format) / 8);
+                const int needed_samples = buffer_size / SDL_AUDIO_BYTESIZE(device->spec.format);
                 const int work_buffer_size = needed_samples * sizeof (float);
                 SDL_AudioSpec outspec;
 
@@ -832,7 +832,7 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device)
 void SDL_OutputAudioThreadShutdown(SDL_AudioDevice *device)
 {
     SDL_assert(!device->iscapture);
-    const int samples = (device->buffer_size / (SDL_AUDIO_BITSIZE(device->spec.format) / 8)) / device->spec.channels;
+    const int samples = (device->buffer_size / SDL_AUDIO_BYTESIZE(device->spec.format)) / device->spec.channels;
     // Wait for the audio to drain. !!! FIXME: don't bother waiting if device is lost.
     SDL_Delay(((samples * 1000) / device->spec.freq) * 2);
     current_audio.impl.ThreadDeinit(device);
@@ -1261,7 +1261,7 @@ static int GetDefaultSampleFramesFromFreq(int freq)
 void SDL_UpdatedAudioDeviceFormat(SDL_AudioDevice *device)
 {
     device->silence_value = SDL_GetSilenceValueForFormat(device->spec.format);
-    device->buffer_size = device->sample_frames * (SDL_AUDIO_BITSIZE(device->spec.format) / 8) * device->spec.channels;
+    device->buffer_size = device->sample_frames * SDL_AUDIO_BYTESIZE(device->spec.format) * device->spec.channels;
     device->work_buffer_size = device->sample_frames * sizeof (float) * device->spec.channels;
     device->work_buffer_size = SDL_max(device->buffer_size, device->work_buffer_size);  // just in case we end up with a 64-bit audio format at some point.
 }

src/audio/SDL_audiocvt.c

@@ -1048,8 +1048,8 @@ void ConvertAudio(int num_frames, const void *src, SDL_AudioFormat src_format, i
 // Calculate the largest frame size needed to convert between the two formats.
 static int CalculateMaxFrameSize(SDL_AudioFormat src_format, int src_channels, SDL_AudioFormat dst_format, int dst_channels)
 {
-    const int src_format_size = SDL_AUDIO_BITSIZE(src_format) / 8;
-    const int dst_format_size = SDL_AUDIO_BITSIZE(dst_format) / 8;
+    const int src_format_size = SDL_AUDIO_BYTESIZE(src_format);
+    const int dst_format_size = SDL_AUDIO_BYTESIZE(dst_format);
     const int max_app_format_size = SDL_max(src_format_size, dst_format_size);
     const int max_format_size = SDL_max(max_app_format_size, sizeof (float));  // ConvertAudio and ResampleAudio use floats.
     const int max_channels = SDL_max(src_channels, dst_channels);
@@ -1058,7 +1058,7 @@ static int CalculateMaxFrameSize(SDL_AudioFormat src_format, int src_channels, S
 
 static int GetAudioSpecFrameSize(const SDL_AudioSpec* spec)
 {
-    return (SDL_AUDIO_BITSIZE(spec->format) / 8) * spec->channels;
+    return SDL_AUDIO_BYTESIZE(spec->format) * spec->channels;
 }
 
 static Sint64 GetStreamResampleRate(SDL_AudioStream* stream, int src_freq)

src/audio/alsa/SDL_alsa_audio.c

@@ -355,7 +355,7 @@ static int ALSA_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buf
 {
     SDL_assert(buffer == device->hidden->mixbuf);
     Uint8 *sample_buf = device->hidden->mixbuf;
-    const int frame_size = ((SDL_AUDIO_BITSIZE(device->spec.format)) / 8) *
+    const int frame_size = SDL_AUDIO_BYTESIZE(device->spec.format) *
                            device->spec.channels;
     snd_pcm_uframes_t frames_left = (snd_pcm_uframes_t) (buflen / frame_size);
 
@@ -402,7 +402,7 @@ static Uint8 *ALSA_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
 static int ALSA_CaptureFromDevice(SDL_AudioDevice *device, void *buffer, int buflen)
 {
     Uint8 *sample_buf = (Uint8 *)buffer;
-    const int frame_size = ((SDL_AUDIO_BITSIZE(device->spec.format)) / 8) *
+    const int frame_size = SDL_AUDIO_BYTESIZE(device->spec.format) *
                            device->spec.channels;
     const int total_frames = buflen / frame_size;
     snd_pcm_uframes_t frames_left = total_frames;

src/audio/emscripten/SDL_emscriptenaudio.c

@@ -38,7 +38,7 @@ static Uint8 *EMSCRIPTENAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_
 
 static int EMSCRIPTENAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buffer_size)
 {
-    const int framelen = (SDL_AUDIO_BITSIZE(device->spec.format) / 8) * device->spec.channels;
+    const int framelen = SDL_AUDIO_BYTESIZE(device->spec.format) * device->spec.channels;
     MAIN_THREAD_EM_ASM({
         var SDL3 = Module['SDL3'];
         var numChannels = SDL3.audio.currentOutputBuffer['numberOfChannels'];

src/audio/n3ds/SDL_n3dsaudio.c

@@ -161,7 +161,7 @@ static int N3DSAUDIO_OpenDevice(SDL_AudioDevice *device)
 
     SDL_memset(device->hidden->waveBuf, 0, sizeof(ndspWaveBuf) * NUM_BUFFERS);
 
-    const int sample_frame_size = device->spec.channels * (SDL_AUDIO_BITSIZE(device->spec.format) / 8);
+    const int sample_frame_size = device->spec.channels * SDL_AUDIO_BYTESIZE(device->spec.format);
     for (unsigned i = 0; i < NUM_BUFFERS; i++) {
         device->hidden->waveBuf[i].data_vaddr = data_vaddr;
         device->hidden->waveBuf[i].nsamples = device->buffer_size / sample_frame_size;

src/audio/netbsd/SDL_netbsdaudio.c

@@ -130,7 +130,7 @@ static void NETBSDAUDIO_WaitDevice(SDL_AudioDevice *device)
             SDL_AudioDeviceDisconnected(device);
             return;
         }
-        const size_t remain = (size_t)((iscapture ? info.record.seek : info.play.seek) * (SDL_AUDIO_BITSIZE(device->spec.format) / 8));
+        const size_t remain = (size_t)((iscapture ? info.record.seek : info.play.seek) * SDL_AUDIO_BYTESIZE(device->spec.format));
         if (!iscapture && (remain >= device->buffer_size)) {
             SDL_Delay(10);
         } else if (iscapture && (remain < device->buffer_size)) {
@@ -181,7 +181,7 @@ static void NETBSDAUDIO_FlushCapture(SDL_AudioDevice *device)
     struct SDL_PrivateAudioData *h = device->hidden;
     audio_info_t info;
     if (ioctl(device->hidden->audio_fd, AUDIO_GETINFO, &info) == 0) {
-        size_t remain = (size_t)(info.record.seek * (SDL_AUDIO_BITSIZE(device->spec.format) / 8));
+        size_t remain = (size_t)(info.record.seek * SDL_AUDIO_BYTESIZE(device->spec.format));
         while (remain > 0) {
             char buf[512];
             const size_t len = SDL_min(sizeof(buf), remain);

src/audio/pipewire/SDL_pipewire.c

@@ -1108,7 +1108,7 @@ static int PIPEWIRE_OpenDevice(SDL_AudioDevice *device)
     }
 
     /* Size of a single audio frame in bytes */
-    priv->stride = (SDL_AUDIO_BITSIZE(device->spec.format) / 8) * device->spec.channels;
+    priv->stride = SDL_AUDIO_BYTESIZE(device->spec.format) * device->spec.channels;
 
     if (device->sample_frames < min_period) {
         device->sample_frames = min_period;

src/audio/wasapi/SDL_wasapi.c

@@ -621,7 +621,7 @@ static int mgmtthrtask_PrepDevice(void *userdata)
         return -1;
     }
 
-    device->hidden->framesize = (SDL_AUDIO_BITSIZE(device->spec.format) / 8) * device->spec.channels;
+    device->hidden->framesize = SDL_AUDIO_BYTESIZE(device->spec.format) * device->spec.channels;
 
     if (device->iscapture) {
         IAudioCaptureClient *capture = NULL;

test/testaudio.c

@@ -513,7 +513,7 @@ static void StreamThing_ontick(Thing *thing, Uint64 now)
         if (!available || (SDL_GetAudioStreamFormat(thing->data.stream.stream, NULL, &spec) < 0)) {
             DestroyThingInPoof(thing);
         } else {
-            const int ticksleft = (int) ((((Uint64) ((available / (SDL_AUDIO_BITSIZE(spec.format) / 8)) / spec.channels)) * 1000) / spec.freq);
+            const int ticksleft = (int) ((((Uint64) ((available / SDL_AUDIO_BYTESIZE(spec.format)) / spec.channels)) * 1000) / spec.freq);
             const float pct = thing->data.stream.total_ticks ? (((float) (ticksleft)) / ((float) thing->data.stream.total_ticks)) : 0.0f;
             thing->progress = 1.0f - pct;
         }
@@ -553,7 +553,7 @@ static void StreamThing_ondrop(Thing *thing, int button, float x, float y)
                 SDL_UnbindAudioStream(thing->data.stream.stream); /* unbind from current device */
                 if (thing->line_connected_to->what == THING_LOGDEV_CAPTURE) {
                     SDL_FlushAudioStream(thing->data.stream.stream);
-                    thing->data.stream.total_ticks = (int) (((((Uint64) (SDL_GetAudioStreamAvailable(thing->data.stream.stream) / (SDL_AUDIO_BITSIZE(spec->format) / 8))) / spec->channels) * 1000) / spec->freq);
+                    thing->data.stream.total_ticks = (int) (((((Uint64) (SDL_GetAudioStreamAvailable(thing->data.stream.stream) / SDL_AUDIO_BYTESIZE(spec->format))) / spec->channels) * 1000) / spec->freq);
                 }
             }
 
@@ -596,7 +596,7 @@ static Thing *CreateStreamThing(const SDL_AudioSpec *spec, const Uint8 *buf, con
     if (buf && buflen) {
         SDL_PutAudioStreamData(thing->data.stream.stream, buf, (int) buflen);
         SDL_FlushAudioStream(thing->data.stream.stream);
-        thing->data.stream.total_ticks = (int) (((((Uint64) (SDL_GetAudioStreamAvailable(thing->data.stream.stream) / (SDL_AUDIO_BITSIZE(spec->format) / 8))) / spec->channels) * 1000) / spec->freq);
+        thing->data.stream.total_ticks = (int) (((((Uint64) (SDL_GetAudioStreamAvailable(thing->data.stream.stream) / SDL_AUDIO_BYTESIZE(spec->format))) / spec->channels) * 1000) / spec->freq);
     }
     thing->ontick = StreamThing_ontick;
     thing->ondrag = StreamThing_ondrag;

test/testaudiostreamdynamicresample.c

@@ -292,7 +292,7 @@ static void loop(void)
 
     if (SDL_GetAudioStreamFormat(stream, &src_spec, &dst_spec) == 0) {
         available_bytes = SDL_GetAudioStreamAvailable(stream);
-        available_seconds = (float)available_bytes / (float)(SDL_AUDIO_BITSIZE(dst_spec.format) / 8 * dst_spec.freq * dst_spec.channels);
+        available_seconds = (float)available_bytes / (float)(SDL_AUDIO_BYTESIZE(dst_spec.format) * dst_spec.freq * dst_spec.channels);
 
         /* keep it looping. */
         if (auto_loop && (available_seconds < 10.0f)) {

test/testautomation_audio.c

@@ -712,8 +712,8 @@ static int audio_convertAudio(void *arg)
                         int src_samplesize, dst_samplesize;
                         int src_silence, dst_silence;
 
-                        src_samplesize = (SDL_AUDIO_BITSIZE(spec1.format) / 8) * spec1.channels;
-                        dst_samplesize = (SDL_AUDIO_BITSIZE(spec2.format) / 8) * spec2.channels;
+                        src_samplesize = SDL_AUDIO_BYTESIZE(spec1.format) * spec1.channels;
+                        dst_samplesize = SDL_AUDIO_BYTESIZE(spec2.format) * spec2.channels;
 
                         src_len = l * src_samplesize;
                         SDLTest_Log("Creating dummy sample buffer of %i length (%i bytes)", l, src_len);