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alsa: Convert /* */ comments to // comments.

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This commit is contained in:
Ryan C. Gordon
2023-09-30 12:02:51 -04:00
parent 64fee85c69
commit 776d9d0ee3
1 changed files with 86 additions and 96 deletions
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182
src/audio/alsa/SDL_alsa_audio.c
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@@ -28,15 +28,15 @@
#define SDL_ALSA_NON_BLOCKING 0
#endif
/* without the thread, you will detect devices on startup, but will not get further hotplug events. But that might be okay. */
// without the thread, you will detect devices on startup, but will not get further hotplug events. But that might be okay.
#ifndef SDL_ALSA_HOTPLUG_THREAD
#define SDL_ALSA_HOTPLUG_THREAD 1
#endif
/* Allow access to a raw mixing buffer */
// Allow access to a raw mixing buffer
#include <sys/types.h>
#include <signal.h> /* For kill() */
#include <signal.h> // For kill()
#include <string.h>
#include "../SDL_audio_c.h"
@@ -99,14 +99,14 @@ static int load_alsa_sym(const char *fn, void **addr)
{
*addr = SDL_LoadFunction(alsa_handle, fn);
if (*addr == NULL) {
/* Don't call SDL_SetError(): SDL_LoadFunction already did. */
// Don't call SDL_SetError(): SDL_LoadFunction already did.
return 0;
}
return 1;
}
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
// cast funcs to char* first, to please GCC's strict aliasing rules.
#define SDL_ALSA_SYM(x) \
if (!load_alsa_sym(#x, (void **)(char *)&ALSA_##x)) \
return -1
@@ -180,7 +180,7 @@ static int LoadALSALibrary(void)
alsa_handle = SDL_LoadObject(alsa_library);
if (alsa_handle == NULL) {
retval = -1;
/* Don't call SDL_SetError(): SDL_LoadObject already did. */
// Don't call SDL_SetError(): SDL_LoadObject already did.
} else {
retval = load_alsa_syms();
if (retval < 0) {
@@ -203,14 +203,14 @@ static int LoadALSALibrary(void)
return 0;
}
#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */
#endif // SDL_AUDIO_DRIVER_ALSA_DYNAMIC
static const char *get_audio_device(void *handle, const int channels)
{
SDL_assert(handle != NULL); // SDL2 used NULL to mean "default" but that's not true in SDL3.
if (SDL_strcmp((const char *) handle, "default") == 0) {
const char *device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */
const char *device = SDL_getenv("AUDIODEV"); // Is there a standard variable name?
if (device != NULL) {
return device;
} else if (channels == 6) {
@@ -224,12 +224,11 @@ static const char *get_audio_device(void *handle, const int channels)
return (const char *)handle;
}
/* !!! FIXME: is there a channel swizzler in alsalib instead? */
/*
* https://bugzilla.libsdl.org/show_bug.cgi?id=110
* "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
* and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
*/
// !!! FIXME: is there a channel swizzler in alsalib instead?
// https://bugzilla.libsdl.org/show_bug.cgi?id=110
// "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
// and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
#define SWIZ6(T) \
static void swizzle_alsa_channels_6_##T(void *buffer, const Uint32 bufferlen) \
{ \
@@ -246,13 +245,13 @@ static const char *get_audio_device(void *handle, const int channels)
} \
}
/* !!! FIXME: is there a channel swizzler in alsalib instead? */
/* !!! FIXME: this screams for a SIMD shuffle operation. */
/*
* https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/mapping-stream-formats-to-speaker-configurations
* For Linux ALSA, this appears to be FL-FR-RL-RR-C-LFE-SL-SR
* and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-SL-SR-RL-RR"
*/
// !!! FIXME: is there a channel swizzler in alsalib instead?
// !!! FIXME: this screams for a SIMD shuffle operation.
// https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/mapping-stream-formats-to-speaker-configurations
// For Linux ALSA, this appears to be FL-FR-RL-RR-C-LFE-SL-SR
// and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-SL-SR-RL-RR"
#define SWIZ8(T) \
static void swizzle_alsa_channels_8_##T(void *buffer, const Uint32 bufferlen) \
{ \
@@ -287,10 +286,8 @@ CHANNEL_SWIZZLE(SWIZ8)
#undef SWIZ6
#undef SWIZ8
/*
* Called right before feeding device->hidden->mixbuf to the hardware. Swizzle
* channels from Windows/Mac order to the format alsalib will want.
*/
// Called right before feeding device->hidden->mixbuf to the hardware. Swizzle
// channels from Windows/Mac order to the format alsalib will want.
static void swizzle_alsa_channels(SDL_AudioDevice *device, void *buffer, Uint32 bufferlen)
{
switch (device->spec.channels) {
@@ -324,13 +321,13 @@ static void swizzle_alsa_channels(SDL_AudioDevice *device, void *buffer, Uint32
}
#ifdef SND_CHMAP_API_VERSION
/* Some devices have the right channel map, no swizzling necessary */
// Some devices have the right channel map, no swizzling necessary
static void no_swizzle(SDL_AudioDevice *device, void *buffer, Uint32 bufferlen)
{
}
#endif /* SND_CHMAP_API_VERSION */
#endif // SND_CHMAP_API_VERSION
/* This function waits until it is possible to write a full sound buffer */
// This function waits until it is possible to write a full sound buffer
static void ALSA_WaitDevice(SDL_AudioDevice *device)
{
const int fulldelay = (int) ((((Uint64) device->sample_frames) * 1000) / device->spec.freq);
@@ -341,7 +338,7 @@ static void ALSA_WaitDevice(SDL_AudioDevice *device)
if (rc < 0 && (rc != -EAGAIN)) {
const int status = ALSA_snd_pcm_recover(device->hidden->pcm_handle, rc, 0);
if (status < 0) {
/* Hmm, not much we can do - abort */
// Hmm, not much we can do - abort
SDL_LogError(SDL_LOG_CATEGORY_AUDIO, "ALSA snd_pcm_avail failed (unrecoverable): %s\n",
ALSA_snd_strerror(rc));
SDL_AudioDeviceDisconnected(device);
@@ -371,15 +368,14 @@ static int ALSA_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buf
sample_buf, frames_left);
//SDL_Log("ALSA PLAYDEVICE: WROTE %d of %d bytes", (status >= 0) ? ((int) (status * frame_size)) : status, (int) (frames_left * frame_size));
if (status < 0) {
if (status == -EAGAIN) {
/* Apparently snd_pcm_recover() doesn't handle this case -
does it assume snd_pcm_wait() above? */
if (status == -EAGAIN) { // !!! FIXME: should this happen if we used snd_pcm_wait and queried for available space?!
// Apparently snd_pcm_recover() doesn't handle this case - does it assume snd_pcm_wait() above?
SDL_Delay(1);
continue;
}
status = ALSA_snd_pcm_recover(device->hidden->pcm_handle, status, 0);
if (status < 0) {
/* Hmm, not much we can do - abort */
// Hmm, not much we can do - abort
SDL_LogError(SDL_LOG_CATEGORY_AUDIO,
"ALSA write failed (unrecoverable): %s",
ALSA_snd_strerror(status));
@@ -387,8 +383,8 @@ static int ALSA_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buf
}
continue;
} else if (status == 0) {
/* No frames were written (no available space in pcm device).
Allow other threads to catch up. */
// !!! FIXME: should this happen if we used snd_pcm_wait and queried for available space?!
// No frames were written (no available space in pcm device). Allow other threads to catch up.
SDL_Delay((frames_left / 2 * 1000) / device->spec.freq);
}
@@ -427,10 +423,10 @@ static int ALSA_CaptureFromDevice(SDL_AudioDevice *device, void *buffer, int buf
if (status == -EAGAIN) {
break; // Can this even happen? Go back to WaitCaptureDevice, where the device lock isn't held.
} else if (status < 0) {
/*printf("ALSA: capture error %d\n", status);*/
//printf("ALSA: capture error %d\n", status);
status = ALSA_snd_pcm_recover(device->hidden->pcm_handle, status, 0);
if (status < 0) {
/* Hmm, not much we can do - abort */
// Hmm, not much we can do - abort
SDL_LogError(SDL_LOG_CATEGORY_AUDIO,
"ALSA read failed (unrecoverable): %s\n",
ALSA_snd_strerror(status));
@@ -439,7 +435,7 @@ static int ALSA_CaptureFromDevice(SDL_AudioDevice *device, void *buffer, int buf
break; // Go back to WaitCaptureDevice, where the device lock isn't held.
}
/*printf("ALSA: captured %d bytes\n", status * frame_size);*/
//printf("ALSA: captured %d bytes\n", status * frame_size);
sample_buf += status * frame_size;
frames_left -= status;
}
@@ -458,9 +454,7 @@ static void ALSA_CloseDevice(SDL_AudioDevice *device)
{
if (device->hidden) {
if (device->hidden->pcm_handle) {
/* Wait for the submitted audio to drain
ALSA_snd_pcm_drop() can hang, so don't use that.
*/
// Wait for the submitted audio to drain. ALSA_snd_pcm_drop() can hang, so don't use that.
SDL_Delay(((device->sample_frames * 1000) / device->spec.freq) * 2);
ALSA_snd_pcm_close(device->hidden->pcm_handle);
}
@@ -476,11 +470,11 @@ static int ALSA_set_buffer_size(SDL_AudioDevice *device, snd_pcm_hw_params_t *pa
snd_pcm_uframes_t persize;
unsigned int periods;
/* Copy the hardware parameters for this setup */
// Copy the hardware parameters for this setup
snd_pcm_hw_params_alloca(&hwparams);
ALSA_snd_pcm_hw_params_copy(hwparams, params);
/* Attempt to match the period size to the requested buffer size */
// Attempt to match the period size to the requested buffer size
persize = device->sample_frames;
status = ALSA_snd_pcm_hw_params_set_period_size_near(
device->hidden->pcm_handle, hwparams, &persize, NULL);
@@ -488,7 +482,7 @@ static int ALSA_set_buffer_size(SDL_AudioDevice *device, snd_pcm_hw_params_t *pa
return -1;
}
/* Need to at least double buffer */
// Need to at least double buffer
periods = 2;
status = ALSA_snd_pcm_hw_params_set_periods_min(
device->hidden->pcm_handle, hwparams, &periods, NULL);
@@ -502,7 +496,7 @@ static int ALSA_set_buffer_size(SDL_AudioDevice *device, snd_pcm_hw_params_t *pa
return -1;
}
/* "set" the hardware with the desired parameters */
// "set" the hardware with the desired parameters
status = ALSA_snd_pcm_hw_params(device->hidden->pcm_handle, hwparams);
if (status < 0) {
return -1;
@@ -510,7 +504,7 @@ static int ALSA_set_buffer_size(SDL_AudioDevice *device, snd_pcm_hw_params_t *pa
device->sample_frames = persize;
/* This is useful for debugging */
// This is useful for debugging
if (SDL_getenv("SDL_AUDIO_ALSA_DEBUG")) {
snd_pcm_uframes_t bufsize;
@@ -529,14 +523,14 @@ static int ALSA_OpenDevice(SDL_AudioDevice *device)
const SDL_bool iscapture = device->iscapture;
int status = 0;
/* Initialize all variables that we clean on shutdown */
// Initialize all variables that we clean on shutdown
device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
if (device->hidden == NULL) {
return SDL_OutOfMemory();
}
/* Open the audio device */
/* Name of device should depend on # channels in spec */
// Open the audio device
// Name of device should depend on # channels in spec
snd_pcm_t *pcm_handle = NULL;
status = ALSA_snd_pcm_open(&pcm_handle,
get_audio_device(device->handle, device->spec.channels),
@@ -549,7 +543,7 @@ static int ALSA_OpenDevice(SDL_AudioDevice *device)
device->hidden->pcm_handle = pcm_handle;
/* Figure out what the hardware is capable of */
// Figure out what the hardware is capable of
snd_pcm_hw_params_t *hwparams = NULL;
snd_pcm_hw_params_alloca(&hwparams);
status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams);
@@ -557,14 +551,14 @@ static int ALSA_OpenDevice(SDL_AudioDevice *device)
return SDL_SetError("ALSA: Couldn't get hardware config: %s", ALSA_snd_strerror(status));
}
/* SDL only uses interleaved sample output */
// SDL only uses interleaved sample output
status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (status < 0) {
return SDL_SetError("ALSA: Couldn't set interleaved access: %s", ALSA_snd_strerror(status));
}
/* Try for a closest match on audio format */
// Try for a closest match on audio format
snd_pcm_format_t format = 0;
const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
SDL_AudioFormat test_format;
@@ -606,9 +600,8 @@ static int ALSA_OpenDevice(SDL_AudioDevice *device)
}
device->spec.format = test_format;
/* Validate number of channels and determine if swizzling is necessary
* Assume original swizzling, until proven otherwise.
*/
// Validate number of channels and determine if swizzling is necessary.
// Assume original swizzling, until proven otherwise.
device->hidden->swizzle_func = swizzle_alsa_channels;
#ifdef SND_CHMAP_API_VERSION
snd_pcm_chmap_t *chmap = ALSA_snd_pcm_get_chmap(pcm_handle);
@@ -620,11 +613,11 @@ static int ALSA_OpenDevice(SDL_AudioDevice *device)
device->hidden->swizzle_func = no_swizzle;
}
}
free(chmap); /* This should NOT be SDL_free() */
free(chmap); // This should NOT be SDL_free()
}
#endif /* SND_CHMAP_API_VERSION */
#endif // SND_CHMAP_API_VERSION
/* Set the number of channels */
// Set the number of channels
status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams,
device->spec.channels);
unsigned int channels = device->spec.channels;
@@ -636,7 +629,7 @@ static int ALSA_OpenDevice(SDL_AudioDevice *device)
device->spec.channels = channels;
}
/* Set the audio rate */
// Set the audio rate
unsigned int rate = device->spec.freq;
status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
&rate, NULL);
@@ -645,13 +638,13 @@ static int ALSA_OpenDevice(SDL_AudioDevice *device)
}
device->spec.freq = rate;
/* Set the buffer size, in samples */
// Set the buffer size, in samples
status = ALSA_set_buffer_size(device, hwparams);
if (status < 0) {
return SDL_SetError("Couldn't set hardware audio parameters: %s", ALSA_snd_strerror(status));
}
/* Set the software parameters */
// Set the software parameters
snd_pcm_sw_params_t *swparams = NULL;
snd_pcm_sw_params_alloca(&swparams);
status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);
@@ -692,8 +685,7 @@ static int ALSA_OpenDevice(SDL_AudioDevice *device)
ALSA_snd_pcm_start(pcm_handle);
/* We're ready to rock and roll. :-) */
return 0;
return 0; // We're ready to rock and roll. :-)
}
typedef struct ALSA_Device
@@ -714,10 +706,10 @@ static void add_device(const SDL_bool iscapture, const char *name, void *hint, A
return;
}
/* Not all alsa devices are enumerable via snd_device_name_get_hint
(i.e. bluetooth devices). Therefore if hint is passed in to this
function as NULL, assume name contains desc.
Make sure not to free the storage associated with desc in this case */
// Not all alsa devices are enumerable via snd_device_name_get_hint
// (i.e. bluetooth devices). Therefore if hint is passed in to this
// function as NULL, assume name contains desc.
// Make sure not to free the storage associated with desc in this case
if (hint) {
desc = ALSA_snd_device_name_get_hint(hint, "DESC");
if (desc == NULL) {
@@ -730,32 +722,31 @@ static void add_device(const SDL_bool iscapture, const char *name, void *hint, A
SDL_assert(name != NULL);
/* some strings have newlines, like "HDA NVidia, HDMI 0\nHDMI Audio Output".
just chop the extra lines off, this seems to get a reasonable device
name without extra details. */
// some strings have newlines, like "HDA NVidia, HDMI 0\nHDMI Audio Output".
// just chop the extra lines off, this seems to get a reasonable device
// name without extra details.
ptr = SDL_strchr(desc, '\n');
if (ptr != NULL) {
*ptr = '\0';
}
/*printf("ALSA: adding %s device '%s' (%s)\n", iscapture ? "capture" : "output", name, desc);*/
//printf("ALSA: adding %s device '%s' (%s)\n", iscapture ? "capture" : "output", name, desc);
handle = SDL_strdup(name);
if (handle == NULL) {
if (hint) {
free(desc); /* This should NOT be SDL_free() */
free(desc); // This should NOT be SDL_free()
}
SDL_free(dev);
return;
}
/* Note that spec is NULL, because we are required to open the device before
* acquiring the mix format, making this information inaccessible at
* enumeration time
*/
// Note that spec is NULL, because we are required to open the device before
// acquiring the mix format, making this information inaccessible at
// enumeration time
SDL_AddAudioDevice(iscapture, desc, NULL, handle);
if (hint) {
free(desc); /* This should NOT be SDL_free() */
free(desc); // This should NOT be SDL_free()
}
dev->name = handle;
dev->iscapture = iscapture;
@@ -783,10 +774,10 @@ static void ALSA_HotplugIteration(SDL_bool *has_default_output, SDL_bool *has_de
unseen = hotplug_devices;
seen = NULL;
/* Apparently there are several different ways that ALSA lists
actual hardware. It could be prefixed with "hw:" or "default:"
or "sysdefault:" and maybe others. Go through the list and see
if we can find a preferred prefix for the system. */
// Apparently there are several different ways that ALSA lists
// actual hardware. It could be prefixed with "hw:" or "default:"
// or "sysdefault:" and maybe others. Go through the list and see
// if we can find a preferred prefix for the system.
for (int i = 0; hints[i]; i++) {
char *name = ALSA_snd_device_name_get_hint(hints[i], "NAME");
if (name == NULL) {
@@ -812,11 +803,11 @@ static void ALSA_HotplugIteration(SDL_bool *has_default_output, SDL_bool *has_de
}
}
free(name); /* This should NOT be SDL_free() */
free(name); // This should NOT be SDL_free()
}
}
/* look through the list of device names to find matches */
// look through the list of device names to find matches
if (match || (has_default >= 0)) { // did we find a device name prefix we like at all...?
for (int i = 0; hints[i]; i++) {
char *name = ALSA_snd_device_name_get_hint(hints[i], "NAME");
@@ -881,18 +872,18 @@ static void ALSA_HotplugIteration(SDL_bool *has_default_output, SDL_bool *has_de
}
}
free(name); /* This should NOT be SDL_free() */
free(name); // This should NOT be SDL_free()
}
}
ALSA_snd_device_name_free_hint(hints);
hotplug_devices = seen; /* now we have a known-good list of attached devices. */
hotplug_devices = seen; // now we have a known-good list of attached devices.
/* report anything still in unseen as removed. */
// report anything still in unseen as removed.
ALSA_Device *next = NULL;
for (ALSA_Device *dev = unseen; dev; dev = next) {
/*printf("ALSA: removing usb %s device '%s'\n", dev->iscapture ? "capture" : "output", dev->name);*/
//printf("ALSA: removing usb %s device '%s'\n", dev->iscapture ? "capture" : "output", dev->name);
next = dev->next;
SDL_AudioDeviceDisconnected(SDL_FindPhysicalAudioDeviceByHandle(dev->name));
SDL_free(dev->name);
@@ -910,13 +901,13 @@ static int SDLCALL ALSA_HotplugThread(void *arg)
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_LOW);
while (!SDL_AtomicGet(&ALSA_hotplug_shutdown)) {
/* Block awhile before checking again, unless we're told to stop. */
// Block awhile before checking again, unless we're told to stop.
const Uint64 ticks = SDL_GetTicks() + 5000;
while (!SDL_AtomicGet(&ALSA_hotplug_shutdown) && SDL_GetTicks() < ticks) {
SDL_Delay(100);
}
ALSA_HotplugIteration(NULL, NULL); /* run the check. */
ALSA_HotplugIteration(NULL, NULL); // run the check.
}
return 0;
@@ -928,7 +919,7 @@ static void ALSA_DetectDevices(SDL_AudioDevice **default_output, SDL_AudioDevice
// ALSA doesn't have a concept of a changeable default device, afaik, so we expose a generic default
// device here. It's the best we can do at this level.
SDL_bool has_default_output = SDL_FALSE, has_default_capture = SDL_FALSE;
ALSA_HotplugIteration(&has_default_output, &has_default_capture); // run once now before a thread continues to check. */
ALSA_HotplugIteration(&has_default_output, &has_default_capture); // run once now before a thread continues to check.
if (has_default_output) {
*default_output = SDL_AddAudioDevice(/*iscapture=*/SDL_FALSE, "ALSA default output device", NULL, SDL_strdup("default"));
}
@@ -939,7 +930,7 @@ static void ALSA_DetectDevices(SDL_AudioDevice **default_output, SDL_AudioDevice
#if SDL_ALSA_HOTPLUG_THREAD
SDL_AtomicSet(&ALSA_hotplug_shutdown, 0);
ALSA_hotplug_thread = SDL_CreateThread(ALSA_HotplugThread, "SDLHotplugALSA", NULL);
/* if the thread doesn't spin, oh well, you just don't get further hotplug events. */
// if the thread doesn't spin, oh well, you just don't get further hotplug events.
#endif
}
@@ -956,9 +947,9 @@ static void ALSA_Deinitialize(void)
}
#endif
/* Shutting down! Clean up any data we've gathered. */
// Shutting down! Clean up any data we've gathered.
for (dev = hotplug_devices; dev; dev = next) {
/*printf("ALSA: at shutdown, removing %s device '%s'\n", dev->iscapture ? "capture" : "output", dev->name);*/
//printf("ALSA: at shutdown, removing %s device '%s'\n", dev->iscapture ? "capture" : "output", dev->name);
next = dev->next;
SDL_free(dev->name);
SDL_free(dev);
@@ -974,7 +965,6 @@ static SDL_bool ALSA_Init(SDL_AudioDriverImpl *impl)
return SDL_FALSE;
}
/* Set the function pointers */
impl->DetectDevices = ALSA_DetectDevices;
impl->OpenDevice = ALSA_OpenDevice;
impl->WaitDevice = ALSA_WaitDevice;
@@ -988,11 +978,11 @@ static SDL_bool ALSA_Init(SDL_AudioDriverImpl *impl)
impl->HasCaptureSupport = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE;
}
AudioBootStrap ALSA_bootstrap = {
"alsa", "ALSA PCM audio", ALSA_Init, SDL_FALSE
};
#endif /* SDL_AUDIO_DRIVER_ALSA */
#endif // SDL_AUDIO_DRIVER_ALSA