mirrors/SDL
1
0
Fork 0
mirror of https://github.com/libsdl-org/SDL.git synced 2025-10-16 23:06:03 +00:00
Code Issues Packages Projects Releases Wiki Activity

audio: Added a postmix callback to logical devices.

Browse Source 
You can see it in action in testaudio by mousing over a logical device; it
will show a visualizer for the current PCM (whatever is currently being
recorded on a capture device, or whatever is being mixed for output on
playback devices).

Fixes #8122.
This commit is contained in:
Ryan C. Gordon
2023-09-07 10:56:09 -04:00
parent 7207bdce5d
commit 3a992af446
7 changed files with 316 additions and 9 deletions
Download Patch File Download Diff File Expand all files Collapse all files

81
src/audio/SDL_audio.c
View File

@@ -715,6 +715,7 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device)
// can we do a basic copy without silencing/mixing the buffer? This is an extremely likely scenario, so we special-case it.
const SDL_bool simple_copy = device->logical_devices && // there's a logical device
!device->logical_devices->next && // there's only _ONE_ logical device
!device->logical_devices->postmix && // there isn't a postmix callback
!SDL_AtomicGet(&device->logical_devices->paused) && // it isn't paused
device->logical_devices->bound_streams && // there's a bound stream
!device->logical_devices->bound_streams->next_binding; // there's only _ONE_ bound stream.
@@ -731,7 +732,7 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device)
SDL_memset(device_buffer + br, device->silence_value, buffer_size - br); // silence whatever we didn't write to.
}
} else { // need to actually mix (or silence the buffer)
float *mix_buffer = (float *) ((device->spec.format == SDL_AUDIO_F32) ? device_buffer : device->mix_buffer);
float *final_mix_buffer = (float *) ((device->spec.format == SDL_AUDIO_F32) ? device_buffer : device->mix_buffer);
const int needed_samples = buffer_size / SDL_AUDIO_BYTESIZE(device->spec.format);
const int work_buffer_size = needed_samples * sizeof (float);
SDL_AudioSpec outspec;
@@ -742,13 +743,20 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device)
outspec.channels = device->spec.channels;
outspec.freq = device->spec.freq;
SDL_memset(mix_buffer, '\0', buffer_size); // start with silence.
SDL_memset(final_mix_buffer, '\0', work_buffer_size); // start with silence.
for (SDL_LogicalAudioDevice *logdev = device->logical_devices; logdev != NULL; logdev = logdev->next) {
if (SDL_AtomicGet(&logdev->paused)) {
continue; // paused? Skip this logical device.
}
const SDL_AudioPostmixCallback postmix = logdev->postmix;
float *mix_buffer = final_mix_buffer;
if (postmix) {
mix_buffer = device->postmix_buffer;
SDL_memset(mix_buffer, '\0', work_buffer_size); // start with silence.
}
for (SDL_AudioStream *stream = logdev->bound_streams; stream != NULL; stream = stream->next_binding) {
SDL_SetAudioStreamFormat(stream, NULL, &outspec);
@@ -764,12 +772,18 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device)
MixFloat32Audio(mix_buffer, (float *) device->work_buffer, br);
}
}
if (postmix) {
SDL_assert(mix_buffer == device->postmix_buffer);
postmix(logdev->postmix_userdata, &outspec, mix_buffer, work_buffer_size);
MixFloat32Audio(final_mix_buffer, mix_buffer, work_buffer_size);
}
}
if (((Uint8 *) mix_buffer) != device_buffer) {
if (((Uint8 *) final_mix_buffer) != device_buffer) {
// !!! FIXME: we can't promise the device buf is aligned/padded for SIMD.
//ConvertAudio(needed_samples * device->spec.channels, mix_buffer, SDL_AUDIO_F32, device->spec.channels, device_buffer, device->spec.format, device->spec.channels, device->work_buffer);
ConvertAudio(needed_samples / device->spec.channels, mix_buffer, SDL_AUDIO_F32, device->spec.channels, device->work_buffer, device->spec.format, device->spec.channels, NULL);
//ConvertAudio(needed_samples * device->spec.channels, final_mix_buffer, SDL_AUDIO_F32, device->spec.channels, device_buffer, device->spec.format, device->spec.channels, device->work_buffer);
ConvertAudio(needed_samples / device->spec.channels, final_mix_buffer, SDL_AUDIO_F32, device->spec.channels, device->work_buffer, device->spec.format, device->spec.channels, NULL);
SDL_memcpy(device_buffer, device->work_buffer, buffer_size);
}
}
@@ -837,21 +851,37 @@ SDL_bool SDL_CaptureAudioThreadIterate(SDL_AudioDevice *device)
current_audio.impl.FlushCapture(device); // nothing wants data, dump anything pending.
} else {
// this SHOULD NOT BLOCK, as we are holding a lock right now. Block in WaitCaptureDevice!
const int rc = current_audio.impl.CaptureFromDevice(device, device->work_buffer, device->buffer_size);
if (rc < 0) { // uhoh, device failed for some reason!
int br = current_audio.impl.CaptureFromDevice(device, device->work_buffer, device->buffer_size);
if (br < 0) { // uhoh, device failed for some reason!
retval = SDL_FALSE;
} else if (rc > 0) { // queue the new data to each bound stream.
} else if (br > 0) { // queue the new data to each bound stream.
for (SDL_LogicalAudioDevice *logdev = device->logical_devices; logdev != NULL; logdev = logdev->next) {
if (SDL_AtomicGet(&logdev->paused)) {
continue; // paused? Skip this logical device.
}
void *output_buffer = device->work_buffer;
SDL_AudioSpec outspec;
SDL_copyp(&outspec, &device->spec);
// I don't know why someone would want a postmix on a capture device, but we offer it for API consistency.
if (logdev->postmix) {
// move to float format.
output_buffer = device->postmix_buffer;
outspec.format = SDL_AUDIO_F32;
const int frames = br / SDL_AUDIO_FRAMESIZE(device->spec);
br = frames * SDL_AUDIO_FRAMESIZE(outspec);
ConvertAudio(frames, device->work_buffer, device->spec.format, outspec.channels, device->postmix_buffer, SDL_AUDIO_F32, outspec.channels, NULL);
logdev->postmix(logdev->postmix_userdata, &outspec, device->postmix_buffer, br);
}
for (SDL_AudioStream *stream = logdev->bound_streams; stream != NULL; stream = stream->next_binding) {
/* this will hold a lock on `stream` while putting. We don't explicitly lock the streams
for iterating here because the binding linked list can only change while the device lock is held.
(we _do_ lock the stream during binding/unbinding to make sure that two threads can't try to bind
the same stream to different devices at the same time, though.) */
if (SDL_PutAudioStreamData(stream, device->work_buffer, rc) < 0) {
SDL_SetAudioStreamFormat(stream, &outspec, NULL);
if (SDL_PutAudioStreamData(stream, output_buffer, br) < 0) {
// oh crud, we probably ran out of memory. This is possibly an overreaction to kill the audio device, but it's likely the whole thing is going down in a moment anyhow.
retval = SDL_FALSE;
break;
@@ -1138,6 +1168,9 @@ static void ClosePhysicalAudioDevice(SDL_AudioDevice *device)
SDL_aligned_free(device->mix_buffer);
device->mix_buffer = NULL;
SDL_aligned_free(device->postmix_buffer);
device->postmix_buffer = NULL;
SDL_memcpy(&device->spec, &device->default_spec, sizeof (SDL_AudioSpec));
device->sample_frames = 0;
device->silence_value = SDL_GetSilenceValueForFormat(device->spec.format);
@@ -1395,6 +1428,28 @@ SDL_bool SDL_IsAudioDevicePaused(SDL_AudioDeviceID devid)
return retval;
}
int SDL_SetAudioPostmixCallback(SDL_AudioDeviceID devid, SDL_AudioPostmixCallback callback, void *userdata)
{
SDL_LogicalAudioDevice *logdev = ObtainLogicalAudioDevice(devid);
int retval = 0;
if (logdev) {
SDL_AudioDevice *device = logdev->physical_device;
if (!device->postmix_buffer) {
device->postmix_buffer = (float *)SDL_aligned_alloc(SDL_SIMDGetAlignment(), device->work_buffer_size);
if (device->mix_buffer == NULL) {
retval = SDL_OutOfMemory();
}
}
if (retval == 0) {
logdev->postmix = callback;
logdev->postmix_userdata = userdata;
}
SDL_UnlockMutex(device->lock);
}
return retval;
}
int SDL_BindAudioStreams(SDL_AudioDeviceID devid, SDL_AudioStream **streams, int num_streams)
{
@@ -1782,6 +1837,14 @@ int SDL_AudioDeviceFormatChangedAlreadyLocked(SDL_AudioDevice *device, const SDL
kill_device = SDL_TRUE;
}
if (device->postmix_buffer) {
SDL_aligned_free(device->postmix_buffer);
device->postmix_buffer = (float *)SDL_aligned_alloc(SDL_SIMDGetAlignment(), device->work_buffer_size);
if (!device->postmix_buffer) {
kill_device = SDL_TRUE;
}
}
SDL_aligned_free(device->mix_buffer);
device->mix_buffer = NULL;
if (device->spec.format != SDL_AUDIO_F32) {