audio: Added a postmix callback to logical devices.
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.main
parent
7207bdce5d
commit
3a992af446
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@ -1126,6 +1126,73 @@ extern DECLSPEC void SDLCALL SDL_DestroyAudioStream(SDL_AudioStream *stream);
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*/
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extern DECLSPEC SDL_AudioStream *SDLCALL SDL_OpenAudioDeviceStream(SDL_AudioDeviceID devid, const SDL_AudioSpec *spec, SDL_AudioStreamCallback callback, void *userdata);
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/**
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* A callback that fires when data is about to be fed to an audio device.
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*
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* This is useful for accessing the final mix, perhaps for writing a
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* visualizer or applying a final effect to the audio data before playback.
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*
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* \sa SDL_SetAudioDevicePostmixCallback
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*/
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typedef void (SDLCALL *SDL_AudioPostmixCallback)(void *userdata, const SDL_AudioSpec *spec, float *buffer, int buflen);
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/**
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* Set a callback that fires when data is about to be fed to an audio device.
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*
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* This is useful for accessing the final mix, perhaps for writing a
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* visualizer or applying a final effect to the audio data before playback.
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*
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* The buffer is the final mix of all bound audio streams on an opened
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* device; this callback will fire regularly for any device that is both
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* opened and unpaused. If there is no new data to mix, either because no
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* streams are bound to the device or all the streams are empty, this
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* callback will still fire with the entire buffer set to silence.
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*
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* This callback is allowed to make changes to the data; the contents of
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* the buffer after this call is what is ultimately passed along to the
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* hardware.
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*
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* The callback is always provided the data in float format (values from
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* -1.0f to 1.0f), but the number of channels or sample rate may be
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* different than the format the app requested when opening the device; SDL
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* might have had to manage a conversion behind the scenes, or the playback
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* might have jumped to new physical hardware when a system default changed,
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* etc. These details may change between calls. Accordingly, the size of the
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* buffer might change between calls as well.
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*
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* This callback can run at any time, and from any thread; if you need to
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* serialize access to your app's data, you should provide and use a mutex or
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* other synchronization device.
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*
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* All of this to say: there are specific needs this callback can fulfill,
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* but it is not the simplest interface. Apps should generally provide audio
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* in their preferred format through an SDL_AudioStream and let SDL handle
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* the difference.
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*
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* This function is extremely time-sensitive; the callback should do the
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* least amount of work possible and return as quickly as it can. The longer
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* the callback runs, the higher the risk of audio dropouts or other problems.
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*
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* This function will block until the audio device is in between iterations,
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* so any existing callback that might be running will finish before this
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* function sets the new callback and returns.
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*
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* Setting a NULL callback function disables any previously-set callback.
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*
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* \param devid The ID of an opened audio device.
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* \param callback A callback function to be called. Can be NULL.
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* \param userdata App-controlled pointer passed to callback. Can be NULL.
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* \returns zero on success, -1 on error; call SDL_GetError() for more
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* information.
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*
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* \threadsafety It is safe to call this function from any thread.
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*
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* \since This function is available since SDL 3.0.0.
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*/
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extern DECLSPEC int SDLCALL SDL_SetAudioPostmixCallback(SDL_AudioDeviceID devid, SDL_AudioPostmixCallback callback, void *userdata);
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/**
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* Load the audio data of a WAVE file into memory.
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*
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@ -715,6 +715,7 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device)
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// can we do a basic copy without silencing/mixing the buffer? This is an extremely likely scenario, so we special-case it.
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const SDL_bool simple_copy = device->logical_devices && // there's a logical device
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!device->logical_devices->next && // there's only _ONE_ logical device
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!device->logical_devices->postmix && // there isn't a postmix callback
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!SDL_AtomicGet(&device->logical_devices->paused) && // it isn't paused
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device->logical_devices->bound_streams && // there's a bound stream
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!device->logical_devices->bound_streams->next_binding; // there's only _ONE_ bound stream.
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@ -731,7 +732,7 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device)
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SDL_memset(device_buffer + br, device->silence_value, buffer_size - br); // silence whatever we didn't write to.
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}
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} else { // need to actually mix (or silence the buffer)
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float *mix_buffer = (float *) ((device->spec.format == SDL_AUDIO_F32) ? device_buffer : device->mix_buffer);
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float *final_mix_buffer = (float *) ((device->spec.format == SDL_AUDIO_F32) ? device_buffer : device->mix_buffer);
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const int needed_samples = buffer_size / SDL_AUDIO_BYTESIZE(device->spec.format);
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const int work_buffer_size = needed_samples * sizeof (float);
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SDL_AudioSpec outspec;
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@ -742,13 +743,20 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device)
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outspec.channels = device->spec.channels;
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outspec.freq = device->spec.freq;
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SDL_memset(mix_buffer, '\0', buffer_size); // start with silence.
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SDL_memset(final_mix_buffer, '\0', work_buffer_size); // start with silence.
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for (SDL_LogicalAudioDevice *logdev = device->logical_devices; logdev != NULL; logdev = logdev->next) {
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if (SDL_AtomicGet(&logdev->paused)) {
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continue; // paused? Skip this logical device.
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}
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const SDL_AudioPostmixCallback postmix = logdev->postmix;
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float *mix_buffer = final_mix_buffer;
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if (postmix) {
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mix_buffer = device->postmix_buffer;
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SDL_memset(mix_buffer, '\0', work_buffer_size); // start with silence.
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}
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for (SDL_AudioStream *stream = logdev->bound_streams; stream != NULL; stream = stream->next_binding) {
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SDL_SetAudioStreamFormat(stream, NULL, &outspec);
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@ -764,12 +772,18 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device)
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MixFloat32Audio(mix_buffer, (float *) device->work_buffer, br);
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}
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}
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if (postmix) {
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SDL_assert(mix_buffer == device->postmix_buffer);
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postmix(logdev->postmix_userdata, &outspec, mix_buffer, work_buffer_size);
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MixFloat32Audio(final_mix_buffer, mix_buffer, work_buffer_size);
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}
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}
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if (((Uint8 *) mix_buffer) != device_buffer) {
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if (((Uint8 *) final_mix_buffer) != device_buffer) {
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// !!! FIXME: we can't promise the device buf is aligned/padded for SIMD.
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//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);
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ConvertAudio(needed_samples / device->spec.channels, mix_buffer, SDL_AUDIO_F32, device->spec.channels, device->work_buffer, device->spec.format, device->spec.channels, NULL);
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//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);
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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);
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SDL_memcpy(device_buffer, device->work_buffer, buffer_size);
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}
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}
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@ -837,21 +851,37 @@ SDL_bool SDL_CaptureAudioThreadIterate(SDL_AudioDevice *device)
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current_audio.impl.FlushCapture(device); // nothing wants data, dump anything pending.
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} else {
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// this SHOULD NOT BLOCK, as we are holding a lock right now. Block in WaitCaptureDevice!
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const int rc = current_audio.impl.CaptureFromDevice(device, device->work_buffer, device->buffer_size);
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if (rc < 0) { // uhoh, device failed for some reason!
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int br = current_audio.impl.CaptureFromDevice(device, device->work_buffer, device->buffer_size);
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if (br < 0) { // uhoh, device failed for some reason!
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retval = SDL_FALSE;
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} else if (rc > 0) { // queue the new data to each bound stream.
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} else if (br > 0) { // queue the new data to each bound stream.
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for (SDL_LogicalAudioDevice *logdev = device->logical_devices; logdev != NULL; logdev = logdev->next) {
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if (SDL_AtomicGet(&logdev->paused)) {
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continue; // paused? Skip this logical device.
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}
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void *output_buffer = device->work_buffer;
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SDL_AudioSpec outspec;
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SDL_copyp(&outspec, &device->spec);
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// I don't know why someone would want a postmix on a capture device, but we offer it for API consistency.
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if (logdev->postmix) {
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// move to float format.
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output_buffer = device->postmix_buffer;
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outspec.format = SDL_AUDIO_F32;
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const int frames = br / SDL_AUDIO_FRAMESIZE(device->spec);
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br = frames * SDL_AUDIO_FRAMESIZE(outspec);
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ConvertAudio(frames, device->work_buffer, device->spec.format, outspec.channels, device->postmix_buffer, SDL_AUDIO_F32, outspec.channels, NULL);
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logdev->postmix(logdev->postmix_userdata, &outspec, device->postmix_buffer, br);
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}
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for (SDL_AudioStream *stream = logdev->bound_streams; stream != NULL; stream = stream->next_binding) {
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/* this will hold a lock on `stream` while putting. We don't explicitly lock the streams
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for iterating here because the binding linked list can only change while the device lock is held.
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(we _do_ lock the stream during binding/unbinding to make sure that two threads can't try to bind
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the same stream to different devices at the same time, though.) */
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if (SDL_PutAudioStreamData(stream, device->work_buffer, rc) < 0) {
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SDL_SetAudioStreamFormat(stream, &outspec, NULL);
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if (SDL_PutAudioStreamData(stream, output_buffer, br) < 0) {
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// 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.
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retval = SDL_FALSE;
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break;
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@ -1138,6 +1168,9 @@ static void ClosePhysicalAudioDevice(SDL_AudioDevice *device)
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SDL_aligned_free(device->mix_buffer);
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device->mix_buffer = NULL;
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SDL_aligned_free(device->postmix_buffer);
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device->postmix_buffer = NULL;
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SDL_memcpy(&device->spec, &device->default_spec, sizeof (SDL_AudioSpec));
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device->sample_frames = 0;
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device->silence_value = SDL_GetSilenceValueForFormat(device->spec.format);
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@ -1395,6 +1428,28 @@ SDL_bool SDL_IsAudioDevicePaused(SDL_AudioDeviceID devid)
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return retval;
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}
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int SDL_SetAudioPostmixCallback(SDL_AudioDeviceID devid, SDL_AudioPostmixCallback callback, void *userdata)
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{
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SDL_LogicalAudioDevice *logdev = ObtainLogicalAudioDevice(devid);
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int retval = 0;
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if (logdev) {
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SDL_AudioDevice *device = logdev->physical_device;
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if (!device->postmix_buffer) {
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device->postmix_buffer = (float *)SDL_aligned_alloc(SDL_SIMDGetAlignment(), device->work_buffer_size);
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if (device->mix_buffer == NULL) {
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retval = SDL_OutOfMemory();
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}
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}
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if (retval == 0) {
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logdev->postmix = callback;
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logdev->postmix_userdata = userdata;
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}
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SDL_UnlockMutex(device->lock);
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}
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return retval;
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}
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int SDL_BindAudioStreams(SDL_AudioDeviceID devid, SDL_AudioStream **streams, int num_streams)
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{
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kill_device = SDL_TRUE;
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}
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if (device->postmix_buffer) {
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SDL_aligned_free(device->postmix_buffer);
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device->postmix_buffer = (float *)SDL_aligned_alloc(SDL_SIMDGetAlignment(), device->work_buffer_size);
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if (!device->postmix_buffer) {
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kill_device = SDL_TRUE;
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}
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}
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SDL_aligned_free(device->mix_buffer);
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device->mix_buffer = NULL;
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if (device->spec.format != SDL_AUDIO_F32) {
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@ -214,6 +214,12 @@ struct SDL_LogicalAudioDevice
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// SDL_TRUE if device was opened with SDL_OpenAudioDeviceStream (so it forbids binding changes, etc).
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SDL_bool simplified;
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// If non-NULL, callback into the app that lets them access the final postmix buffer.
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SDL_AudioPostmixCallback postmix;
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// App-supplied pointer for postmix callback.
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void *postmix_userdata;
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// double-linked list of opened devices on the same physical device.
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SDL_LogicalAudioDevice *next;
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SDL_LogicalAudioDevice *prev;
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// Scratch buffers used for mixing.
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Uint8 *work_buffer;
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Uint8 *mix_buffer;
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float *postmix_buffer;
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// Size of work_buffer (and mix_buffer) in bytes.
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int work_buffer_size;
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@ -904,6 +904,7 @@ SDL3_0.0.0 {
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SDL_SetWindowFocusable;
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SDL_GetAudioStreamFrequencyRatio;
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SDL_SetAudioStreamFrequencyRatio;
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SDL_SetAudioPostmixCallback;
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# extra symbols go here (don't modify this line)
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local: *;
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};
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@ -929,3 +929,4 @@
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#define SDL_SetWindowFocusable SDL_SetWindowFocusable_REAL
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#define SDL_GetAudioStreamFrequencyRatio SDL_GetAudioStreamFrequencyRatio_REAL
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#define SDL_SetAudioStreamFrequencyRatio SDL_SetAudioStreamFrequencyRatio_REAL
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#define SDL_SetAudioPostmixCallback SDL_SetAudioPostmixCallback_REAL
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@ -975,3 +975,4 @@ SDL_DYNAPI_PROC(int,SDL_GDKGetDefaultUser,(XUserHandle *a),(a),return)
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SDL_DYNAPI_PROC(int,SDL_SetWindowFocusable,(SDL_Window *a, SDL_bool b),(a,b),return)
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SDL_DYNAPI_PROC(float,SDL_GetAudioStreamFrequencyRatio,(SDL_AudioStream *a),(a),return)
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SDL_DYNAPI_PROC(int,SDL_SetAudioStreamFrequencyRatio,(SDL_AudioStream *a, float b),(a,b),return)
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SDL_DYNAPI_PROC(int,SDL_SetAudioPostmixCallback,(SDL_AudioDeviceID a, SDL_AudioPostmixCallback b, void *c),(a,b,c),return)
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167
test/testaudio.c
167
test/testaudio.c
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#include "testutils.h"
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#define POOF_LIFETIME 250
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#define VISUALIZER_WIDTH 100
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#define VISUALIZER_HEIGHT 50
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typedef struct Texture
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{
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SDL_bool iscapture;
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SDL_AudioSpec spec;
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Thing *physdev;
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SDL_bool visualizer_enabled;
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SDL_bool visualizer_updated;
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SDL_Texture *visualizer;
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SDL_Mutex *postmix_lock;
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float *postmix_buffer;
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int postmix_buflen;
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int postmix_allocated;
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SDL_AudioSpec postmix_spec;
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SDL_AtomicInt postmix_updated;
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} logdev;
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struct {
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SDL_AudioSpec spec;
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static Thing *things = NULL;
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static char *current_titlebar = NULL;
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static Thing *mouseover_thing = NULL;
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static Thing *droppable_highlighted_thing = NULL;
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static Thing *dragging_thing = NULL;
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static int dragging_button = -1;
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return;
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}
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if (mouseover_thing == thing) {
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mouseover_thing = NULL;
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}
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if (droppable_highlighted_thing == thing) {
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droppable_highlighted_thing = NULL;
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}
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if (dragging_thing == thing) {
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dragging_thing = NULL;
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}
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switch (thing->what) {
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case THING_POOF: break;
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case THING_NULL: break;
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case THING_TRASHCAN: break;
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case THING_LOGDEV:
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case THING_LOGDEV_CAPTURE:
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SDL_CloseAudioDevice(thing->data.logdev.devid);
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SDL_DestroyTexture(thing->data.logdev.visualizer);
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SDL_DestroyMutex(thing->data.logdev.postmix_lock);
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SDL_free(thing->data.logdev.postmix_buffer);
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break;
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case THING_PHYSDEV:
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case THING_PHYSDEV_CAPTURE:
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SDL_free(thing->data.physdev.name);
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break;
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case THING_WAV:
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SDL_free(thing->data.wav.buf);
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break;
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case THING_STREAM:
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SDL_DestroyAudioStream(thing->data.stream.stream);
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break;
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@ -748,6 +779,135 @@ static void LogicalDeviceThing_ondrop(Thing *thing, int button, float x, float y
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}
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}
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static void SDLCALL PostmixCallback(void *userdata, const SDL_AudioSpec *spec, float *buffer, int buflen)
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{
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Thing *thing = (Thing *) userdata;
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SDL_LockMutex(thing->data.logdev.postmix_lock);
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if (thing->data.logdev.postmix_allocated < buflen) {
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void *ptr = SDL_realloc(thing->data.logdev.postmix_buffer, buflen);
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if (!ptr) {
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SDL_UnlockMutex(thing->data.logdev.postmix_lock);
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return; /* oh well. */
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}
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thing->data.logdev.postmix_buffer = (float *) ptr;
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thing->data.logdev.postmix_allocated = buflen;
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}
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SDL_copyp(&thing->data.logdev.postmix_spec, spec);
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SDL_memcpy(thing->data.logdev.postmix_buffer, buffer, buflen);
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thing->data.logdev.postmix_buflen = buflen;
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SDL_AtomicSet(&thing->data.logdev.postmix_updated, 1);
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SDL_UnlockMutex(thing->data.logdev.postmix_lock);
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}
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static void UpdateVisualizer(SDL_Renderer *renderer, SDL_Texture *visualizer, const int channels, const float *buffer, const int buflen)
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{
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static const SDL_Color channel_colors[8] = {
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{ 255, 255, 255, 255 },
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{ 255, 0, 0, 255 },
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{ 0, 255, 0, 255 },
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{ 0, 0, 255, 255 },
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{ 255, 255, 0, 255 },
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{ 0, 255, 255, 255 },
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{ 255, 0, 255, 255 },
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{ 127, 127, 127, 255 }
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};
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SDL_SetRenderTarget(renderer, visualizer);
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SDL_SetRenderDrawColor(renderer, 0, 0, 0, 0);
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SDL_RenderClear(renderer);
|
||||
|
||||
if (buffer && buflen) {
|
||||
const int frames = (buflen / sizeof (float)) / channels;
|
||||
const int skip = frames / (VISUALIZER_WIDTH * 2);
|
||||
int i, j;
|
||||
|
||||
for (i = channels - 1; i >= 0; i--) {
|
||||
const SDL_Color *color = &channel_colors[i % SDL_arraysize(channel_colors)];
|
||||
SDL_FPoint points[VISUALIZER_WIDTH + 2];
|
||||
float prevx = 0.0f;
|
||||
int pointidx = 1;
|
||||
|
||||
points[0].x = 0.0f;
|
||||
points[0].y = VISUALIZER_HEIGHT * 0.5f;
|
||||
|
||||
for (j = 0; j < (SDL_arraysize(points)-1); j++) {
|
||||
const float val = buffer[((j * skip) * channels) + i];
|
||||
const float x = prevx + 2;
|
||||
const float y = (VISUALIZER_HEIGHT * 0.5f) - (VISUALIZER_HEIGHT * (val * 0.5f));
|
||||
SDL_assert(pointidx < SDL_arraysize(points));
|
||||
points[pointidx].x = x;
|
||||
points[pointidx].y = y;
|
||||
pointidx++;
|
||||
prevx = x;
|
||||
}
|
||||
|
||||
SDL_SetRenderDrawColor(renderer, color->r, color->g, color->b, 255);
|
||||
SDL_RenderLines(renderer, points, pointidx);
|
||||
}
|
||||
}
|
||||
|
||||
SDL_SetRenderTarget(renderer, NULL);
|
||||
}
|
||||
|
||||
static void LogicalDeviceThing_ontick(Thing *thing, Uint64 now)
|
||||
{
|
||||
const SDL_bool ismousedover = (thing == mouseover_thing) ? SDL_TRUE : SDL_FALSE;
|
||||
|
||||
if (!thing->data.logdev.visualizer || !thing->data.logdev.postmix_lock) { /* need these to work, skip if they failed. */
|
||||
return;
|
||||
}
|
||||
|
||||
if (thing->data.logdev.visualizer_enabled != ismousedover) {
|
||||
thing->data.logdev.visualizer_enabled = ismousedover;
|
||||
if (!ismousedover) {
|
||||
SDL_SetAudioPostmixCallback(thing->data.logdev.devid, NULL, NULL);
|
||||
} else {
|
||||
if (thing->data.logdev.postmix_buffer) {
|
||||
SDL_memset(thing->data.logdev.postmix_buffer, '\0', thing->data.logdev.postmix_buflen);
|
||||
}
|
||||
SDL_AtomicSet(&thing->data.logdev.postmix_updated, 1); /* so this will at least clear the texture later. */
|
||||
SDL_SetAudioPostmixCallback(thing->data.logdev.devid, PostmixCallback, thing);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
static void LogicalDeviceThing_ondraw(Thing *thing, SDL_Renderer *renderer)
|
||||
{
|
||||
if (thing->data.logdev.visualizer_enabled) {
|
||||
SDL_FRect dst;
|
||||
dst.w = thing->rect.w;
|
||||
dst.h = thing->rect.h;
|
||||
dst.x = thing->rect.x + ((thing->rect.w - dst.w) / 2);
|
||||
dst.y = thing->rect.y + ((thing->rect.h - dst.h) / 2);
|
||||
|
||||
if (SDL_AtomicGet(&thing->data.logdev.postmix_updated)) {
|
||||
float *buffer;
|
||||
int channels;
|
||||
int buflen;
|
||||
|
||||
SDL_LockMutex(thing->data.logdev.postmix_lock);
|
||||
channels = thing->data.logdev.postmix_spec.channels;
|
||||
buflen = thing->data.logdev.postmix_buflen;
|
||||
buffer = (float *) SDL_malloc(thing->data.logdev.postmix_buflen);
|
||||
if (buffer) {
|
||||
SDL_memcpy(buffer, thing->data.logdev.postmix_buffer, thing->data.logdev.postmix_buflen);
|
||||
SDL_AtomicSet(&thing->data.logdev.postmix_updated, 0);
|
||||
}
|
||||
SDL_UnlockMutex(thing->data.logdev.postmix_lock);
|
||||
|
||||
UpdateVisualizer(renderer, thing->data.logdev.visualizer, channels, buffer, buflen);
|
||||
SDL_free(buffer);
|
||||
}
|
||||
|
||||
SDL_SetRenderDrawColor(renderer, 255, 255, 255, 30);
|
||||
SDL_RenderTexture(renderer, thing->data.logdev.visualizer, NULL, &dst);
|
||||
}
|
||||
}
|
||||
|
||||
static Thing *CreateLogicalDeviceThing(Thing *parent, const SDL_AudioDeviceID which, const float x, const float y)
|
||||
{
|
||||
static const ThingType can_be_dropped_onto[] = { THING_TRASHCAN, THING_NULL };
|
||||
|
@ -760,9 +920,16 @@ static Thing *CreateLogicalDeviceThing(Thing *parent, const SDL_AudioDeviceID wh
|
|||
thing->data.logdev.devid = which;
|
||||
thing->data.logdev.iscapture = iscapture;
|
||||
thing->data.logdev.physdev = physthing;
|
||||
thing->data.logdev.visualizer = SDL_CreateTexture(state->renderers[0], SDL_PIXELFORMAT_RGBA8888, SDL_TEXTUREACCESS_TARGET, VISUALIZER_WIDTH, VISUALIZER_HEIGHT);
|
||||
thing->data.logdev.postmix_lock = SDL_CreateMutex();
|
||||
if (thing->data.logdev.visualizer) {
|
||||
SDL_SetTextureBlendMode(thing->data.logdev.visualizer, SDL_BLENDMODE_BLEND);
|
||||
}
|
||||
thing->line_connected_to = physthing;
|
||||
thing->ontick = LogicalDeviceThing_ontick;
|
||||
thing->ondrag = DeviceThing_ondrag;
|
||||
thing->ondrop = LogicalDeviceThing_ondrop;
|
||||
thing->ondraw = LogicalDeviceThing_ondraw;
|
||||
thing->can_be_dropped_onto = can_be_dropped_onto;
|
||||
|
||||
SetLogicalDeviceTitlebar(thing);
|
||||
|
|
Loading…
Reference in New Issue