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@@ -225,20 +225,20 @@ static const char *get_audio_device(void *handle, const int channels)
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}
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/* This function waits until it is possible to write a full sound buffer */
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-static void ALSA_WaitDevice(SDL_AudioDevice *_this)
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+static void ALSA_WaitDevice(SDL_AudioDevice *device)
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{
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#if SDL_ALSA_NON_BLOCKING
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- const snd_pcm_sframes_t needed = (snd_pcm_sframes_t)_this->spec.samples;
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- while (SDL_AtomicGet(&_this->enabled)) {
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- const snd_pcm_sframes_t rc = ALSA_snd_pcm_avail(_this->hidden->pcm_handle);
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+ const snd_pcm_sframes_t needed = (snd_pcm_sframes_t)device->spec.samples;
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+ while (SDL_AtomicGet(&device->enabled)) {
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+ const snd_pcm_sframes_t rc = ALSA_snd_pcm_avail(device->hidden->pcm_handle);
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if ((rc < 0) && (rc != -EAGAIN)) {
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/* Hmm, not much we can do - abort */
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fprintf(stderr, "ALSA snd_pcm_avail failed (unrecoverable): %s\n",
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ALSA_snd_strerror(rc));
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- SDL_OpenedAudioDeviceDisconnected(_this);
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+ SDL_OpenedAudioDeviceDisconnected(device);
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return;
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} else if (rc < needed) {
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- const Uint32 delay = ((needed - (SDL_max(rc, 0))) * 1000) / _this->spec.freq;
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+ const Uint32 delay = ((needed - (SDL_max(rc, 0))) * 1000) / device->spec.freq;
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SDL_Delay(SDL_max(delay, 10));
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} else {
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break; /* ready to go! */
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@@ -311,15 +311,15 @@ CHANNEL_SWIZZLE(SWIZ8)
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#undef SWIZ8
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/*
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- * Called right before feeding _this->hidden->mixbuf to the hardware. Swizzle
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+ * Called right before feeding device->hidden->mixbuf to the hardware. Swizzle
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* channels from Windows/Mac order to the format alsalib will want.
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*/
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-static void swizzle_alsa_channels(SDL_AudioDevice *_this, void *buffer, Uint32 bufferlen)
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+static void swizzle_alsa_channels(SDL_AudioDevice *device, void *buffer, Uint32 bufferlen)
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{
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- switch (_this->spec.channels) {
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+ switch (device->spec.channels) {
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#define CHANSWIZ(chans) \
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case chans: \
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- switch ((_this->spec.format & (0xFF))) { \
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+ switch ((device->spec.format & (0xFF))) { \
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case 8: \
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swizzle_alsa_channels_##chans##_Uint8(buffer, bufferlen); \
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break; \
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@@ -348,22 +348,22 @@ static void swizzle_alsa_channels(SDL_AudioDevice *_this, void *buffer, Uint32 b
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#ifdef SND_CHMAP_API_VERSION
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/* Some devices have the right channel map, no swizzling necessary */
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-static void no_swizzle(SDL_AudioDevice *_this, void *buffer, Uint32 bufferlen)
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+static void no_swizzle(SDL_AudioDevice *device, void *buffer, Uint32 bufferlen)
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{
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}
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#endif /* SND_CHMAP_API_VERSION */
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-static void ALSA_PlayDevice(SDL_AudioDevice *_this)
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+static void ALSA_PlayDevice(SDL_AudioDevice *device)
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{
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- const Uint8 *sample_buf = (const Uint8 *)_this->hidden->mixbuf;
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- const int frame_size = ((SDL_AUDIO_BITSIZE(_this->spec.format)) / 8) *
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- _this->spec.channels;
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- snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t)_this->spec.samples);
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+ const Uint8 *sample_buf = (const Uint8 *)device->hidden->mixbuf;
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+ const int frame_size = ((SDL_AUDIO_BITSIZE(device->spec.format)) / 8) *
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+ device->spec.channels;
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+ snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t)device->spec.samples);
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- _this->hidden->swizzle_func(_this, _this->hidden->mixbuf, frames_left);
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+ device->hidden->swizzle_func(device, device->hidden->mixbuf, frames_left);
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- while (frames_left > 0 && SDL_AtomicGet(&_this->enabled)) {
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- int status = ALSA_snd_pcm_writei(_this->hidden->pcm_handle,
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+ while (frames_left > 0 && SDL_AtomicGet(&device->enabled)) {
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+ int status = ALSA_snd_pcm_writei(device->hidden->pcm_handle,
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sample_buf, frames_left);
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if (status < 0) {
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@@ -373,20 +373,20 @@ static void ALSA_PlayDevice(SDL_AudioDevice *_this)
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SDL_Delay(1);
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continue;
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}
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- status = ALSA_snd_pcm_recover(_this->hidden->pcm_handle, status, 0);
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+ status = ALSA_snd_pcm_recover(device->hidden->pcm_handle, status, 0);
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if (status < 0) {
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/* Hmm, not much we can do - abort */
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SDL_LogError(SDL_LOG_CATEGORY_AUDIO,
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"ALSA write failed (unrecoverable): %s\n",
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ALSA_snd_strerror(status));
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- SDL_OpenedAudioDeviceDisconnected(_this);
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+ SDL_OpenedAudioDeviceDisconnected(device);
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return;
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}
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continue;
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} else if (status == 0) {
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/* No frames were written (no available space in pcm device).
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Allow other threads to catch up. */
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- Uint32 delay = (frames_left / 2 * 1000) / _this->spec.freq;
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+ Uint32 delay = (frames_left / 2 * 1000) / device->spec.freq;
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SDL_Delay(delay);
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}
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@@ -395,34 +395,34 @@ static void ALSA_PlayDevice(SDL_AudioDevice *_this)
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}
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}
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-static Uint8 *ALSA_GetDeviceBuf(SDL_AudioDevice *_this)
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+static Uint8 *ALSA_GetDeviceBuf(SDL_AudioDevice *device)
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{
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- return _this->hidden->mixbuf;
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+ return device->hidden->mixbuf;
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}
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-static int ALSA_CaptureFromDevice(SDL_AudioDevice *_this, void *buffer, int buflen)
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+static int ALSA_CaptureFromDevice(SDL_AudioDevice *device, void *buffer, int buflen)
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{
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Uint8 *sample_buf = (Uint8 *)buffer;
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- const int frame_size = ((SDL_AUDIO_BITSIZE(_this->spec.format)) / 8) *
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- _this->spec.channels;
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+ const int frame_size = ((SDL_AUDIO_BITSIZE(device->spec.format)) / 8) *
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+ device->spec.channels;
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const int total_frames = buflen / frame_size;
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snd_pcm_uframes_t frames_left = total_frames;
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snd_pcm_uframes_t wait_time = frame_size / 2;
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SDL_assert((buflen % frame_size) == 0);
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- while (frames_left > 0 && SDL_AtomicGet(&_this->enabled)) {
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+ while (frames_left > 0 && SDL_AtomicGet(&device->enabled)) {
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int status;
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- status = ALSA_snd_pcm_readi(_this->hidden->pcm_handle,
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+ status = ALSA_snd_pcm_readi(device->hidden->pcm_handle,
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sample_buf, frames_left);
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if (status == -EAGAIN) {
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- ALSA_snd_pcm_wait(_this->hidden->pcm_handle, wait_time);
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+ ALSA_snd_pcm_wait(device->hidden->pcm_handle, wait_time);
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status = 0;
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} else if (status < 0) {
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/*printf("ALSA: capture error %d\n", status);*/
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- status = ALSA_snd_pcm_recover(_this->hidden->pcm_handle, status, 0);
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+ status = ALSA_snd_pcm_recover(device->hidden->pcm_handle, status, 0);
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if (status < 0) {
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/* Hmm, not much we can do - abort */
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SDL_LogError(SDL_LOG_CATEGORY_AUDIO,
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@@ -438,32 +438,32 @@ static int ALSA_CaptureFromDevice(SDL_AudioDevice *_this, void *buffer, int bufl
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frames_left -= status;
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}
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- _this->hidden->swizzle_func(_this, buffer, total_frames - frames_left);
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+ device->hidden->swizzle_func(device, buffer, total_frames - frames_left);
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return (total_frames - frames_left) * frame_size;
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}
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-static void ALSA_FlushCapture(SDL_AudioDevice *_this)
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+static void ALSA_FlushCapture(SDL_AudioDevice *device)
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{
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- ALSA_snd_pcm_reset(_this->hidden->pcm_handle);
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+ ALSA_snd_pcm_reset(device->hidden->pcm_handle);
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}
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-static void ALSA_CloseDevice(SDL_AudioDevice *_this)
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+static void ALSA_CloseDevice(SDL_AudioDevice *device)
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{
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- if (_this->hidden->pcm_handle) {
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+ if (device->hidden->pcm_handle) {
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/* Wait for the submitted audio to drain
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ALSA_snd_pcm_drop() can hang, so don't use that.
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*/
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- Uint32 delay = ((_this->spec.samples * 1000) / _this->spec.freq) * 2;
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+ Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq) * 2;
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SDL_Delay(delay);
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- ALSA_snd_pcm_close(_this->hidden->pcm_handle);
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+ ALSA_snd_pcm_close(device->hidden->pcm_handle);
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}
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- SDL_free(_this->hidden->mixbuf);
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- SDL_free(_this->hidden);
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+ SDL_free(device->hidden->mixbuf);
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+ SDL_free(device->hidden);
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}
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-static int ALSA_set_buffer_size(SDL_AudioDevice *_this, snd_pcm_hw_params_t *params)
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+static int ALSA_set_buffer_size(SDL_AudioDevice *device, snd_pcm_hw_params_t *params)
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{
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int status;
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snd_pcm_hw_params_t *hwparams;
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@@ -475,9 +475,9 @@ static int ALSA_set_buffer_size(SDL_AudioDevice *_this, snd_pcm_hw_params_t *par
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ALSA_snd_pcm_hw_params_copy(hwparams, params);
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/* Attempt to match the period size to the requested buffer size */
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- persize = _this->spec.samples;
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+ persize = device->spec.samples;
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status = ALSA_snd_pcm_hw_params_set_period_size_near(
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- _this->hidden->pcm_handle, hwparams, &persize, NULL);
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+ device->hidden->pcm_handle, hwparams, &persize, NULL);
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if (status < 0) {
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return -1;
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}
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@@ -485,24 +485,24 @@ static int ALSA_set_buffer_size(SDL_AudioDevice *_this, snd_pcm_hw_params_t *par
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/* Need to at least double buffer */
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periods = 2;
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status = ALSA_snd_pcm_hw_params_set_periods_min(
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- _this->hidden->pcm_handle, hwparams, &periods, NULL);
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+ device->hidden->pcm_handle, hwparams, &periods, NULL);
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if (status < 0) {
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return -1;
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}
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status = ALSA_snd_pcm_hw_params_set_periods_first(
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- _this->hidden->pcm_handle, hwparams, &periods, NULL);
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+ device->hidden->pcm_handle, hwparams, &periods, NULL);
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if (status < 0) {
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return -1;
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}
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/* "set" the hardware with the desired parameters */
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- status = ALSA_snd_pcm_hw_params(_this->hidden->pcm_handle, hwparams);
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+ status = ALSA_snd_pcm_hw_params(device->hidden->pcm_handle, hwparams);
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if (status < 0) {
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return -1;
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}
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- _this->spec.samples = persize;
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+ device->spec.samples = persize;
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/* This is useful for debugging */
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if (SDL_getenv("SDL_AUDIO_ALSA_DEBUG")) {
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@@ -518,10 +518,10 @@ static int ALSA_set_buffer_size(SDL_AudioDevice *_this, snd_pcm_hw_params_t *par
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return 0;
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}
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-static int ALSA_OpenDevice(SDL_AudioDevice *_this, const char *devname)
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+static int ALSA_OpenDevice(SDL_AudioDevice *device, const char *devname)
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{
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int status = 0;
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- SDL_bool iscapture = _this->iscapture;
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+ SDL_bool iscapture = device->iscapture;
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snd_pcm_t *pcm_handle = NULL;
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snd_pcm_hw_params_t *hwparams = NULL;
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snd_pcm_sw_params_t *swparams = NULL;
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@@ -536,16 +536,16 @@ static int ALSA_OpenDevice(SDL_AudioDevice *_this, const char *devname)
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#endif
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/* Initialize all variables that we clean on shutdown */
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- _this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*_this->hidden));
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- if (_this->hidden == NULL) {
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+ device->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*device->hidden));
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+ if (device->hidden == NULL) {
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return SDL_OutOfMemory();
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}
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- SDL_zerop(_this->hidden);
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+ SDL_zerop(device->hidden);
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/* Open the audio device */
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/* Name of device should depend on # channels in spec */
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status = ALSA_snd_pcm_open(&pcm_handle,
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- get_audio_device(_this->handle, _this->spec.channels),
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+ get_audio_device(device->handle, device->spec.channels),
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iscapture ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
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SND_PCM_NONBLOCK);
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@@ -553,7 +553,7 @@ static int ALSA_OpenDevice(SDL_AudioDevice *_this, const char *devname)
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return SDL_SetError("ALSA: Couldn't open audio device: %s", ALSA_snd_strerror(status));
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}
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- _this->hidden->pcm_handle = pcm_handle;
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+ device->hidden->pcm_handle = pcm_handle;
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/* Figure out what the hardware is capable of */
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snd_pcm_hw_params_alloca(&hwparams);
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@@ -570,7 +570,7 @@ static int ALSA_OpenDevice(SDL_AudioDevice *_this, const char *devname)
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}
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/* Try for a closest match on audio format */
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- closefmts = SDL_ClosestAudioFormats(_this->spec.format);
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+ closefmts = SDL_ClosestAudioFormats(device->spec.format);
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while ((test_format = *(closefmts++)) != 0) {
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switch (test_format) {
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case SDL_AUDIO_U8:
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@@ -607,19 +607,19 @@ static int ALSA_OpenDevice(SDL_AudioDevice *_this, const char *devname)
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if (!test_format) {
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return SDL_SetError("%s: Unsupported audio format", "alsa");
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}
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- _this->spec.format = test_format;
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+ device->spec.format = test_format;
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/* Validate number of channels and determine if swizzling is necessary
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* Assume original swizzling, until proven otherwise.
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*/
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- _this->hidden->swizzle_func = swizzle_alsa_channels;
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+ device->hidden->swizzle_func = swizzle_alsa_channels;
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#ifdef SND_CHMAP_API_VERSION
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chmap = ALSA_snd_pcm_get_chmap(pcm_handle);
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if (chmap) {
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if (ALSA_snd_pcm_chmap_print(chmap, sizeof(chmap_str), chmap_str) > 0) {
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if (SDL_strcmp("FL FR FC LFE RL RR", chmap_str) == 0 ||
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SDL_strcmp("FL FR FC LFE SL SR", chmap_str) == 0) {
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- _this->hidden->swizzle_func = no_swizzle;
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+ device->hidden->swizzle_func = no_swizzle;
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}
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}
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free(chmap); /* This should NOT be SDL_free() */
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@@ -628,27 +628,27 @@ static int ALSA_OpenDevice(SDL_AudioDevice *_this, const char *devname)
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/* Set the number of channels */
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status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams,
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- _this->spec.channels);
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- channels = _this->spec.channels;
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+ device->spec.channels);
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+ channels = device->spec.channels;
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if (status < 0) {
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status = ALSA_snd_pcm_hw_params_get_channels(hwparams, &channels);
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if (status < 0) {
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return SDL_SetError("ALSA: Couldn't set audio channels");
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}
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- _this->spec.channels = channels;
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+ device->spec.channels = channels;
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}
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/* Set the audio rate */
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- rate = _this->spec.freq;
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+ rate = device->spec.freq;
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status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
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&rate, NULL);
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if (status < 0) {
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return SDL_SetError("ALSA: Couldn't set audio frequency: %s", ALSA_snd_strerror(status));
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}
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- _this->spec.freq = rate;
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+ device->spec.freq = rate;
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/* Set the buffer size, in samples */
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- status = ALSA_set_buffer_size(_this, hwparams);
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+ status = ALSA_set_buffer_size(device, hwparams);
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if (status < 0) {
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return SDL_SetError("Couldn't set hardware audio parameters: %s", ALSA_snd_strerror(status));
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}
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@@ -659,7 +659,7 @@ static int ALSA_OpenDevice(SDL_AudioDevice *_this, const char *devname)
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if (status < 0) {
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return SDL_SetError("ALSA: Couldn't get software config: %s", ALSA_snd_strerror(status));
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}
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- status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, _this->spec.samples);
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+ status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, device->spec.samples);
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if (status < 0) {
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return SDL_SetError("Couldn't set minimum available samples: %s", ALSA_snd_strerror(status));
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}
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@@ -674,16 +674,16 @@ static int ALSA_OpenDevice(SDL_AudioDevice *_this, const char *devname)
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}
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/* Calculate the final parameters for this audio specification */
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- SDL_CalculateAudioSpec(&_this->spec);
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+ SDL_CalculateAudioSpec(&device->spec);
|
|
|
|
|
|
/* Allocate mixing buffer */
|
|
|
if (!iscapture) {
|
|
|
- _this->hidden->mixlen = _this->spec.size;
|
|
|
- _this->hidden->mixbuf = (Uint8 *)SDL_malloc(_this->hidden->mixlen);
|
|
|
- if (_this->hidden->mixbuf == NULL) {
|
|
|
+ device->hidden->mixlen = device->spec.size;
|
|
|
+ device->hidden->mixbuf = (Uint8 *)SDL_malloc(device->hidden->mixlen);
|
|
|
+ if (device->hidden->mixbuf == NULL) {
|
|
|
return SDL_OutOfMemory();
|
|
|
}
|
|
|
- SDL_memset(_this->hidden->mixbuf, _this->spec.silence, _this->hidden->mixlen);
|
|
|
+ SDL_memset(device->hidden->mixbuf, device->spec.silence, device->hidden->mixlen);
|
|
|
}
|
|
|
|
|
|
#if !SDL_ALSA_NON_BLOCKING
|