geemili
/
SDL
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

Added SDL_AUDIO_FRAMESIZE

main
Brick 2023-09-03 22:07:05 +01:00 committed by Sam Lantinga
parent 53122593f8
commit f2ca9a615b
11 changed files with 29 additions and 33 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

3
include/SDL3/SDL_audio.h
View File

@ -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.

6
src/audio/SDL_audio.c
View File

@ -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.
}

21
src/audio/SDL_audiocvt.c
View File

@ -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);

6
src/audio/alsa/SDL_alsa_audio.c
View File

@ -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) *
device->spec.channels;
const int frame_size = SDL_AUDIO_FRAMESIZE(device->spec);
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) *
device->spec.channels;
const int frame_size = SDL_AUDIO_FRAMESIZE(device->spec);
const int total_frames = buflen / frame_size;
snd_pcm_uframes_t frames_left = total_frames;

2
src/audio/emscripten/SDL_emscriptenaudio.c
View File

@ -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'];

2
src/audio/n3ds/SDL_n3dsaudio.c
View File

@ -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;

2
src/audio/pipewire/SDL_pipewire.c
View File

@ -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;

2
src/audio/wasapi/SDL_wasapi.c
View File

@ -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;

6
test/testaudio.c
View File

@ -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;

2
test/testaudiostreamdynamicresample.c
View File

@ -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)) {

10
test/testautomation_audio.c
View File

@ -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;
dst_samplesize = SDL_AUDIO_BYTESIZE(spec2.format) * spec2.channels;
src_framesize = SDL_AUDIO_FRAMESIZE(spec1);
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) {