Added SDL_AUDIO_FRAMESIZE

include/SDL3/SDL_audio.h

@@ -152,6 +152,9 @@ typedef struct SDL_AudioSpec
     int freq;                   /**< sample rate: sample frames per second */
 } SDL_AudioSpec;
 
+/* Calculate the size of each audio frame (in bytes) */
+#define SDL_AUDIO_FRAMESIZE(x) (SDL_AUDIO_BYTESIZE((x).format) * (x).channels)
+
 /* SDL_AudioStream is an audio conversion interface.
     - It can handle resampling data in chunks without generating
       artifacts, when it doesn't have the complete buffer available.

src/audio/SDL_audio.c

@@ -832,9 +832,9 @@ 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_BYTESIZE(device->spec.format)) / device->spec.channels;
+    const int frames = device->buffer_size / SDL_AUDIO_FRAMESIZE(device->spec);
     // Wait for the audio to drain. !!! FIXME: don't bother waiting if device is lost.
-    SDL_Delay(((samples * 1000) / device->spec.freq) * 2);
+    SDL_Delay(((frames * 1000) / device->spec.freq) * 2);
     current_audio.impl.ThreadDeinit(device);
     SDL_AudioThreadFinalize(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_BYTESIZE(device->spec.format) * device->spec.channels;
+    device->buffer_size = device->sample_frames * SDL_AUDIO_FRAMESIZE(device->spec);
     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

@@ -1056,11 +1056,6 @@ static int CalculateMaxFrameSize(SDL_AudioFormat src_format, int src_channels, S
     return max_format_size * max_channels;
 }
 
-static int GetAudioSpecFrameSize(const SDL_AudioSpec* spec)
-{
-    return SDL_AUDIO_BYTESIZE(spec->format) * spec->channels;
-}
-
 static Sint64 GetStreamResampleRate(SDL_AudioStream* stream, int src_freq)
 {
     src_freq = (int)((float)src_freq * stream->freq_ratio);
@@ -1070,7 +1065,7 @@ static Sint64 GetStreamResampleRate(SDL_AudioStream* stream, int src_freq)
 
 static int ResetHistoryBuffer(SDL_AudioStream *stream, const SDL_AudioSpec *spec)
 {
-    const size_t history_buffer_allocation = GetHistoryBufferSampleFrames() * GetAudioSpecFrameSize(spec);
+    const size_t history_buffer_allocation = GetHistoryBufferSampleFrames() * SDL_AUDIO_FRAMESIZE(*spec);
     Uint8 *history_buffer = stream->history_buffer;
 
     if (stream->history_buffer_allocation < history_buffer_allocation) {
@@ -1313,7 +1308,7 @@ int SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len)
         return -1;
     }
 
-    if ((len % GetAudioSpecFrameSize(&stream->src_spec)) != 0) {
+    if ((len % SDL_AUDIO_FRAMESIZE(stream->src_spec)) != 0) {
         SDL_UnlockMutex(stream->lock);
         return SDL_SetError("Can't add partial sample frames");
     }
@@ -1398,7 +1393,7 @@ static void UpdateStreamHistoryBuffer(SDL_AudioStream* stream, const SDL_AudioSp
 
     // Even if we aren't currently resampling, we always need to update the history buffer
     Uint8 *history_buffer = stream->history_buffer;
-    int history_bytes = history_buffer_frames * GetAudioSpecFrameSize(spec);
+    int history_bytes = history_buffer_frames * SDL_AUDIO_FRAMESIZE(*spec);
 
     if (left_padding != NULL) {
         // Fill in the left padding using the history buffer
@@ -1418,7 +1413,7 @@ static void UpdateStreamHistoryBuffer(SDL_AudioStream* stream, const SDL_AudioSp
 
 static Sint64 GetAudioStreamTrackAvailableFrames(SDL_AudioStream* stream, SDL_AudioTrack* track, Sint64 resample_offset)
 {
-    size_t input_frames = track->queued_bytes / GetAudioSpecFrameSize(&track->spec);
+    size_t input_frames = track->queued_bytes / SDL_AUDIO_FRAMESIZE(track->spec);
     Sint64 resample_rate = GetStreamResampleRate(stream, track->spec.freq);
     Sint64 output_frames = (Sint64) input_frames;
 
@@ -1460,7 +1455,7 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int ou
 
     const SDL_AudioFormat src_format = src_spec->format;
     const int src_channels = src_spec->channels;
-    const int src_frame_size = GetAudioSpecFrameSize(src_spec);
+    const int src_frame_size = SDL_AUDIO_FRAMESIZE(*src_spec);
 
     const SDL_AudioFormat dst_format = dst_spec->format;
     const int dst_channels = dst_spec->channels;
@@ -1646,7 +1641,7 @@ int SDL_GetAudioStreamData(SDL_AudioStream *stream, void *voidbuf, int len)
         return -1;
     }
 
-    const int dst_frame_size = GetAudioSpecFrameSize(&stream->dst_spec);
+    const int dst_frame_size = SDL_AUDIO_FRAMESIZE(stream->dst_spec);
 
     len -= len % dst_frame_size;  // chop off any fractional sample frame.
 
@@ -1663,7 +1658,7 @@ int SDL_GetAudioStreamData(SDL_AudioStream *stream, void *voidbuf, int len)
             approx_request = GetResamplerNeededInputFrames((int) approx_request, resample_rate, 0);
         }
 
-        approx_request *= GetAudioSpecFrameSize(&stream->src_spec);  // convert sample frames to bytes.
+        approx_request *= SDL_AUDIO_FRAMESIZE(stream->src_spec);  // convert sample frames to bytes.
 
         if (approx_request > 0) {  // don't call the callback if we can satisfy this request with existing data.
             stream->get_callback(stream->get_callback_userdata, stream, (int) SDL_min(approx_request, SDL_INT_MAX));
@@ -1749,7 +1744,7 @@ int SDL_GetAudioStreamAvailable(SDL_AudioStream *stream)
     Sint64 count = GetAudioStreamAvailableFrames(stream);
 
     // convert from sample frames to bytes in destination format.
-    count *= GetAudioSpecFrameSize(&stream->dst_spec);
+    count *= SDL_AUDIO_FRAMESIZE(stream->dst_spec);
 
     SDL_UnlockMutex(stream->lock);
 

src/audio/alsa/SDL_alsa_audio.c

@@ -355,8 +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_BYTESIZE(device->spec.format) *
+    const int frame_size = SDL_AUDIO_FRAMESIZE(device->spec);
-                           device->spec.channels;
     snd_pcm_uframes_t frames_left = (snd_pcm_uframes_t) (buflen / frame_size);
 
     device->hidden->swizzle_func(device, sample_buf, frames_left);
@@ -402,8 +401,7 @@ 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_BYTESIZE(device->spec.format) *
+    const int frame_size = SDL_AUDIO_FRAMESIZE(device->spec);
-                           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_BYTESIZE(device->spec.format) * device->spec.channels;
+    const int framelen = SDL_AUDIO_FRAMESIZE(device->spec);
     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_BYTESIZE(device->spec.format);
+    const int sample_frame_size = SDL_AUDIO_FRAMESIZE(device->spec);
     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/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_BYTESIZE(device->spec.format) * device->spec.channels;
+    priv->stride = SDL_AUDIO_FRAMESIZE(device->spec);
 
     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_BYTESIZE(device->spec.format) * device->spec.channels;
+    device->hidden->framesize = SDL_AUDIO_FRAMESIZE(device->spec);
 
     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_BYTESIZE(spec.format)) / spec.channels)) * 1000) / spec.freq);
+            const int ticksleft = (int) ((((Uint64) (available / SDL_AUDIO_FRAMESIZE(spec))) * 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_BYTESIZE(spec->format))) / spec->channels) * 1000) / spec->freq);
+                    thing->data.stream.total_ticks = (int) ((((Uint64) (SDL_GetAudioStreamAvailable(thing->data.stream.stream) / SDL_AUDIO_FRAMESIZE(*spec))) * 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_BYTESIZE(spec->format))) / spec->channels) * 1000) / spec->freq);
+        thing->data.stream.total_ticks = (int) ((((Uint64) (SDL_GetAudioStreamAvailable(thing->data.stream.stream) / SDL_AUDIO_FRAMESIZE(*spec))) * 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_BYTESIZE(dst_spec.format) * dst_spec.freq * dst_spec.channels);
+        available_seconds = (float)available_bytes / (float)(SDL_AUDIO_FRAMESIZE(dst_spec) * dst_spec.freq);
 
         /* keep it looping. */
         if (auto_loop && (available_seconds < 10.0f)) {

test/testautomation_audio.c

@@ -709,13 +709,13 @@ static int audio_convertAudio(void *arg)
                         Uint8 *dst_buf = NULL, *src_buf = NULL;
                         int dst_len = 0, src_len = 0, real_dst_len = 0;
                         int l = 64, m;
-                        int src_samplesize, dst_samplesize;
+                        int src_framesize, dst_framesize;
                         int src_silence, dst_silence;
 
-                        src_samplesize = SDL_AUDIO_BYTESIZE(spec1.format) * spec1.channels;
+                        src_framesize = SDL_AUDIO_FRAMESIZE(spec1);
-                        dst_samplesize = SDL_AUDIO_BYTESIZE(spec2.format) * spec2.channels;
+                        dst_framesize = SDL_AUDIO_FRAMESIZE(spec2);
 
-                        src_len = l * src_samplesize;
+                        src_len = l * src_framesize;
                         SDLTest_Log("Creating dummy sample buffer of %i length (%i bytes)", l, src_len);
                         src_buf = (Uint8 *)SDL_malloc(src_len);
                         SDLTest_AssertCheck(src_buf != NULL, "Check src data buffer to convert is not NULL");
@@ -726,7 +726,7 @@ static int audio_convertAudio(void *arg)
                         src_silence = SDL_GetSilenceValueForFormat(spec1.format);
                         SDL_memset(src_buf, src_silence, src_len);
 
-                        dst_len = ((int)((((Sint64)l * spec2.freq) - 1) / spec1.freq) + 1) * dst_samplesize;
+                        dst_len = ((int)((((Sint64)l * spec2.freq) - 1) / spec1.freq) + 1) * dst_framesize;
                         dst_buf = (Uint8 *)SDL_malloc(dst_len);
                         SDLTest_AssertCheck(dst_buf != NULL, "Check dst data buffer to convert is not NULL");
                         if (dst_buf == NULL) {