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alsa: no CamelCase

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Renamed types, functions, variables.
This commit is contained in:
Max Kellermann
2009-01-25 13:05:16 +01:00
parent 27baf6913e
commit d887b6353f
1 changed files with 103 additions and 93 deletions
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192
src/output/alsa_plugin.c
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@@ -40,38 +40,38 @@ enum {
typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer, typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
snd_pcm_uframes_t size); snd_pcm_uframes_t size);
typedef struct _AlsaData { struct alsa_data {
char *device; char *device;
/** the mode flags passed to snd_pcm_open */ /** the mode flags passed to snd_pcm_open */
int mode; int mode;
snd_pcm_t *pcmHandle; snd_pcm_t *pcm;
alsa_writei_t *writei; alsa_writei_t *writei;
unsigned int buffer_time; unsigned int buffer_time;
unsigned int period_time; unsigned int period_time;
int sampleSize; int frame_size;
int useMmap; bool use_mmap;
struct mixer mixer; struct mixer mixer;
};
} AlsaData;
static const char * static const char *
alsa_device(const AlsaData *ad) alsa_device(const struct alsa_data *ad)
{ {
return ad->device != NULL ? ad->device : default_device; return ad->device != NULL ? ad->device : default_device;
} }
static AlsaData *newAlsaData(void) static struct alsa_data *
alsa_data_new(void)
{ {
AlsaData *ret = g_new(AlsaData, 1); struct alsa_data *ret = g_new(struct alsa_data, 1);
ret->device = NULL; ret->device = NULL;
ret->mode = 0; ret->mode = 0;
ret->pcmHandle = NULL; ret->pcm = NULL;
ret->writei = snd_pcm_writei; ret->writei = snd_pcm_writei;
ret->useMmap = 0; ret->use_mmap = false;
ret->buffer_time = MPD_ALSA_BUFFER_TIME_US; ret->buffer_time = MPD_ALSA_BUFFER_TIME_US;
ret->period_time = MPD_ALSA_PERIOD_TIME_US; ret->period_time = MPD_ALSA_PERIOD_TIME_US;
@@ -81,7 +81,8 @@ static AlsaData *newAlsaData(void)
return ret; return ret;
} }
static void freeAlsaData(AlsaData * ad) static void
alsa_data_free(struct alsa_data *ad)
{ {
g_free(ad->device); g_free(ad->device);
mixer_finish(&ad->mixer); mixer_finish(&ad->mixer);
@@ -89,11 +90,11 @@ static void freeAlsaData(AlsaData * ad)
} }
static void static void
alsa_configure(AlsaData *ad, struct config_param *param) alsa_configure(struct alsa_data *ad, struct config_param *param)
{ {
ad->device = config_dup_block_string(param, "device", NULL); ad->device = config_dup_block_string(param, "device", NULL);
ad->useMmap = config_get_block_bool(param, "use_mmap", false); ad->use_mmap = config_get_block_bool(param, "use_mmap", false);
ad->buffer_time = config_get_block_unsigned(param, "buffer_time", ad->buffer_time = config_get_block_unsigned(param, "buffer_time",
MPD_ALSA_BUFFER_TIME_US); MPD_ALSA_BUFFER_TIME_US);
@@ -117,13 +118,13 @@ alsa_configure(AlsaData *ad, struct config_param *param)
} }
static void * static void *
alsa_initDriver(G_GNUC_UNUSED struct audio_output *ao, alsa_init(G_GNUC_UNUSED struct audio_output *ao,
G_GNUC_UNUSED const struct audio_format *audio_format, G_GNUC_UNUSED const struct audio_format *audio_format,
struct config_param *param) struct config_param *param)
{ {
/* no need for pthread_once thread-safety when reading config */ /* no need for pthread_once thread-safety when reading config */
static int free_global_registered; static int free_global_registered;
AlsaData *ad = newAlsaData(); struct alsa_data *ad = alsa_data_new();
if (!free_global_registered) { if (!free_global_registered) {
atexit((void(*)(void))snd_config_update_free_global); atexit((void(*)(void))snd_config_update_free_global);
@@ -138,14 +139,16 @@ alsa_initDriver(G_GNUC_UNUSED struct audio_output *ao,
return ad; return ad;
} }
static void alsa_finishDriver(void *data) static void
alsa_finish(void *data)
{ {
AlsaData *ad = data; struct alsa_data *ad = data;
freeAlsaData(ad); alsa_data_free(ad);
} }
static bool alsa_testDefault(void) static bool
alsa_test_default_device(void)
{ {
snd_pcm_t *handle; snd_pcm_t *handle;
@@ -161,7 +164,8 @@ static bool alsa_testDefault(void)
return true; return true;
} }
static snd_pcm_format_t get_bitformat(const struct audio_format *af) static snd_pcm_format_t
get_bitformat(const struct audio_format *af)
{ {
switch (af->bits) { switch (af->bits) {
case 8: return SND_PCM_FORMAT_S8; case 8: return SND_PCM_FORMAT_S8;
@@ -172,14 +176,15 @@ static snd_pcm_format_t get_bitformat(const struct audio_format *af)
return SND_PCM_FORMAT_UNKNOWN; return SND_PCM_FORMAT_UNKNOWN;
} }
static bool alsa_openDevice(void *data, struct audio_format *audioFormat) static bool
alsa_open(void *data, struct audio_format *audio_format)
{ {
AlsaData *ad = data; struct alsa_data *ad = data;
snd_pcm_format_t bitformat; snd_pcm_format_t bitformat;
snd_pcm_hw_params_t *hwparams; snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams; snd_pcm_sw_params_t *swparams;
unsigned int sample_rate = audioFormat->sample_rate; unsigned int sample_rate = audio_format->sample_rate;
unsigned int channels = audioFormat->channels; unsigned int channels = audio_format->channels;
snd_pcm_uframes_t alsa_buffer_size; snd_pcm_uframes_t alsa_buffer_size;
snd_pcm_uframes_t alsa_period_size; snd_pcm_uframes_t alsa_period_size;
int err; int err;
@@ -190,14 +195,14 @@ static bool alsa_openDevice(void *data, struct audio_format *audioFormat)
mixer_open(&ad->mixer); mixer_open(&ad->mixer);
if ((bitformat = get_bitformat(audioFormat)) == SND_PCM_FORMAT_UNKNOWN) if ((bitformat = get_bitformat(audio_format)) == SND_PCM_FORMAT_UNKNOWN)
g_warning("ALSA device \"%s\" doesn't support %u bit audio\n", g_warning("ALSA device \"%s\" doesn't support %u bit audio\n",
alsa_device(ad), audioFormat->bits); alsa_device(ad), audio_format->bits);
err = snd_pcm_open(&ad->pcmHandle, alsa_device(ad), err = snd_pcm_open(&ad->pcm, alsa_device(ad),
SND_PCM_STREAM_PLAYBACK, ad->mode); SND_PCM_STREAM_PLAYBACK, ad->mode);
if (err < 0) { if (err < 0) {
ad->pcmHandle = NULL; ad->pcm = NULL;
goto error; goto error;
} }
@@ -207,72 +212,72 @@ configure_hw:
snd_pcm_hw_params_alloca(&hwparams); snd_pcm_hw_params_alloca(&hwparams);
cmd = "snd_pcm_hw_params_any"; cmd = "snd_pcm_hw_params_any";
err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams); err = snd_pcm_hw_params_any(ad->pcm, hwparams);
if (err < 0) if (err < 0)
goto error; goto error;
if (ad->useMmap) { if (ad->use_mmap) {
err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams, err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
SND_PCM_ACCESS_MMAP_INTERLEAVED); SND_PCM_ACCESS_MMAP_INTERLEAVED);
if (err < 0) { if (err < 0) {
g_warning("Cannot set mmap'ed mode on ALSA device \"%s\": %s\n", g_warning("Cannot set mmap'ed mode on ALSA device \"%s\": %s\n",
alsa_device(ad), snd_strerror(-err)); alsa_device(ad), snd_strerror(-err));
g_warning("Falling back to direct write mode\n"); g_warning("Falling back to direct write mode\n");
ad->useMmap = 0; ad->use_mmap = false;
} else } else
ad->writei = snd_pcm_mmap_writei; ad->writei = snd_pcm_mmap_writei;
} }
if (!ad->useMmap) { if (!ad->use_mmap) {
cmd = "snd_pcm_hw_params_set_access"; cmd = "snd_pcm_hw_params_set_access";
err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams, err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED); SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) if (err < 0)
goto error; goto error;
ad->writei = snd_pcm_writei; ad->writei = snd_pcm_writei;
} }
err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat); err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, bitformat);
if (err == -EINVAL && audioFormat->bits != 16) { if (err == -EINVAL && audio_format->bits != 16) {
/* fall back to 16 bit, let pcm_convert.c do the conversion */ /* fall back to 16 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
SND_PCM_FORMAT_S16); SND_PCM_FORMAT_S16);
if (err == 0) { if (err == 0) {
g_debug("ALSA device \"%s\": converting %u bit to 16 bit\n", g_debug("ALSA device \"%s\": converting %u bit to 16 bit\n",
alsa_device(ad), audioFormat->bits); alsa_device(ad), audio_format->bits);
audioFormat->bits = 16; audio_format->bits = 16;
} }
} }
if (err < 0) { if (err < 0) {
g_warning("ALSA device \"%s\" does not support %u bit audio: %s\n", g_warning("ALSA device \"%s\" does not support %u bit audio: %s\n",
alsa_device(ad), audioFormat->bits, snd_strerror(-err)); alsa_device(ad), audio_format->bits, snd_strerror(-err));
goto fail; goto fail;
} }
err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams, err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
&channels); &channels);
if (err < 0) { if (err < 0) {
g_warning("ALSA device \"%s\" does not support %i channels: %s\n", g_warning("ALSA device \"%s\" does not support %i channels: %s\n",
alsa_device(ad), (int)audioFormat->channels, alsa_device(ad), (int)audio_format->channels,
snd_strerror(-err)); snd_strerror(-err));
goto fail; goto fail;
} }
audioFormat->channels = (int8_t)channels; audio_format->channels = (int8_t)channels;
err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams, err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams,
&sample_rate, NULL); &sample_rate, NULL);
if (err < 0 || sample_rate == 0) { if (err < 0 || sample_rate == 0) {
g_warning("ALSA device \"%s\" does not support %u Hz audio\n", g_warning("ALSA device \"%s\" does not support %u Hz audio\n",
alsa_device(ad), audioFormat->sample_rate); alsa_device(ad), audio_format->sample_rate);
goto fail; goto fail;
} }
audioFormat->sample_rate = sample_rate; audio_format->sample_rate = sample_rate;
if (ad->buffer_time > 0) { if (ad->buffer_time > 0) {
buffer_time = ad->buffer_time; buffer_time = ad->buffer_time;
cmd = "snd_pcm_hw_params_set_buffer_time_near"; cmd = "snd_pcm_hw_params_set_buffer_time_near";
err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams, err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm, hwparams,
&buffer_time, NULL); &buffer_time, NULL);
if (err < 0) if (err < 0)
goto error; goto error;
@@ -281,14 +286,14 @@ configure_hw:
if (period_time_ro > 0) { if (period_time_ro > 0) {
period_time = period_time_ro; period_time = period_time_ro;
cmd = "snd_pcm_hw_params_set_period_time_near"; cmd = "snd_pcm_hw_params_set_period_time_near";
err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams, err = snd_pcm_hw_params_set_period_time_near(ad->pcm, hwparams,
&period_time, NULL); &period_time, NULL);
if (err < 0) if (err < 0)
goto error; goto error;
} }
cmd = "snd_pcm_hw_params"; cmd = "snd_pcm_hw_params";
err = snd_pcm_hw_params(ad->pcmHandle, hwparams); err = snd_pcm_hw_params(ad->pcm, hwparams);
if (err == -EPIPE && --retry > 0 && period_time_ro > 0) { if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
period_time_ro = period_time_ro >> 1; period_time_ro = period_time_ro >> 1;
goto configure_hw; goto configure_hw;
@@ -312,32 +317,32 @@ configure_hw:
snd_pcm_sw_params_alloca(&swparams); snd_pcm_sw_params_alloca(&swparams);
cmd = "snd_pcm_sw_params_current"; cmd = "snd_pcm_sw_params_current";
err = snd_pcm_sw_params_current(ad->pcmHandle, swparams); err = snd_pcm_sw_params_current(ad->pcm, swparams);
if (err < 0) if (err < 0)
goto error; goto error;
cmd = "snd_pcm_sw_params_set_start_threshold"; cmd = "snd_pcm_sw_params_set_start_threshold";
err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams, err = snd_pcm_sw_params_set_start_threshold(ad->pcm, swparams,
alsa_buffer_size - alsa_buffer_size -
alsa_period_size); alsa_period_size);
if (err < 0) if (err < 0)
goto error; goto error;
cmd = "snd_pcm_sw_params_set_avail_min"; cmd = "snd_pcm_sw_params_set_avail_min";
err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams, err = snd_pcm_sw_params_set_avail_min(ad->pcm, swparams,
alsa_period_size); alsa_period_size);
if (err < 0) if (err < 0)
goto error; goto error;
cmd = "snd_pcm_sw_params"; cmd = "snd_pcm_sw_params";
err = snd_pcm_sw_params(ad->pcmHandle, swparams); err = snd_pcm_sw_params(ad->pcm, swparams);
if (err < 0) if (err < 0)
goto error; goto error;
ad->sampleSize = audio_format_frame_size(audioFormat); ad->frame_size = audio_format_frame_size(audio_format);
g_debug("ALSA device \"%s\" will be playing %i bit, %u channel audio at %u Hz\n", g_debug("ALSA device \"%s\" will be playing %i bit, %u channel audio at %u Hz\n",
alsa_device(ad), audioFormat->bits, channels, sample_rate); alsa_device(ad), audio_format->bits, channels, sample_rate);
return true; return true;
@@ -350,13 +355,14 @@ error:
alsa_device(ad), snd_strerror(-err)); alsa_device(ad), snd_strerror(-err));
} }
fail: fail:
if (ad->pcmHandle) if (ad->pcm)
snd_pcm_close(ad->pcmHandle); snd_pcm_close(ad->pcm);
ad->pcmHandle = NULL; ad->pcm = NULL;
return false; return false;
} }
static int alsa_errorRecovery(AlsaData * ad, int err) static int
alsa_recover(struct alsa_data *ad, int err)
{ {
if (err == -EPIPE) { if (err == -EPIPE) {
g_debug("Underrun on ALSA device \"%s\"\n", alsa_device(ad)); g_debug("Underrun on ALSA device \"%s\"\n", alsa_device(ad));
@@ -364,23 +370,23 @@ static int alsa_errorRecovery(AlsaData * ad, int err)
g_debug("ALSA device \"%s\" was suspended\n", alsa_device(ad)); g_debug("ALSA device \"%s\" was suspended\n", alsa_device(ad));
} }
switch (snd_pcm_state(ad->pcmHandle)) { switch (snd_pcm_state(ad->pcm)) {
case SND_PCM_STATE_PAUSED: case SND_PCM_STATE_PAUSED:
err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0); err = snd_pcm_pause(ad->pcm, /* disable */ 0);
break; break;
case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_SUSPENDED:
err = snd_pcm_resume(ad->pcmHandle); err = snd_pcm_resume(ad->pcm);
if (err == -EAGAIN) if (err == -EAGAIN)
return 0; return 0;
/* fall-through to snd_pcm_prepare: */ /* fall-through to snd_pcm_prepare: */
case SND_PCM_STATE_SETUP: case SND_PCM_STATE_SETUP:
case SND_PCM_STATE_XRUN: case SND_PCM_STATE_XRUN:
err = snd_pcm_prepare(ad->pcmHandle); err = snd_pcm_prepare(ad->pcm);
break; break;
case SND_PCM_STATE_DISCONNECTED: case SND_PCM_STATE_DISCONNECTED:
/* so alsa_closeDevice won't try to drain: */ /* so alsa_closeDevice won't try to drain: */
snd_pcm_close(ad->pcmHandle); snd_pcm_close(ad->pcm);
ad->pcmHandle = NULL; ad->pcm = NULL;
break; break;
/* this is no error, so just keep running */ /* this is no error, so just keep running */
case SND_PCM_STATE_RUNNING: case SND_PCM_STATE_RUNNING:
@@ -394,52 +400,56 @@ static int alsa_errorRecovery(AlsaData * ad, int err)
return err; return err;
} }
static void alsa_dropBufferedAudio(void *data) static void
alsa_cancel(void *data)
{ {
AlsaData *ad = data; struct alsa_data *ad = data;
alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle)); alsa_recover(ad, snd_pcm_drop(ad->pcm));
} }
static void alsa_closeDevice(void *data) static void
alsa_close(void *data)
{ {
AlsaData *ad = data; struct alsa_data *ad = data;
if (ad->pcmHandle) { if (ad->pcm != NULL) {
if (snd_pcm_state(ad->pcmHandle) == SND_PCM_STATE_RUNNING) { if (snd_pcm_state(ad->pcm) == SND_PCM_STATE_RUNNING)
snd_pcm_drain(ad->pcmHandle); snd_pcm_drain(ad->pcm);
}
snd_pcm_close(ad->pcmHandle); snd_pcm_close(ad->pcm);
ad->pcmHandle = NULL; ad->pcm = NULL;
} }
mixer_close(&ad->mixer); mixer_close(&ad->mixer);
} }
static bool static bool
alsa_playAudio(void *data, const char *playChunk, size_t size) alsa_play(void *data, const char *chunk, size_t size)
{ {
AlsaData *ad = data; struct alsa_data *ad = data;
int ret; int ret;
size /= ad->sampleSize; size /= ad->frame_size;
while (size > 0) { while (size > 0) {
ret = ad->writei(ad->pcmHandle, playChunk, size); ret = ad->writei(ad->pcm, chunk, size);
if (ret == -EAGAIN || ret == -EINTR) if (ret == -EAGAIN || ret == -EINTR)
continue; continue;
if (ret < 0) { if (ret < 0) {
if (alsa_errorRecovery(ad, ret) < 0) { if (alsa_recover(ad, ret) < 0) {
g_warning("closing ALSA device \"%s\" due to write " g_warning("closing ALSA device \"%s\" due to write "
"error: %s\n", "error: %s\n",
alsa_device(ad), snd_strerror(-errno)); alsa_device(ad), snd_strerror(-errno));
return false; return false;
} }
continue; continue;
} }
playChunk += ret * ad->sampleSize; chunk += ret * ad->frame_size;
size -= ret; size -= ret;
} }
@@ -449,18 +459,18 @@ alsa_playAudio(void *data, const char *playChunk, size_t size)
static bool static bool
alsa_control(void *data, int cmd, void *arg) alsa_control(void *data, int cmd, void *arg)
{ {
AlsaData *ad = data; struct alsa_data *ad = data;
return mixer_control(&ad->mixer, cmd, arg); return mixer_control(&ad->mixer, cmd, arg);
} }
const struct audio_output_plugin alsaPlugin = { const struct audio_output_plugin alsaPlugin = {
.name = "alsa", .name = "alsa",
.test_default_device = alsa_testDefault, .test_default_device = alsa_test_default_device,
.init = alsa_initDriver, .init = alsa_init,
.finish = alsa_finishDriver, .finish = alsa_finish,
.open = alsa_openDevice, .open = alsa_open,
.play = alsa_playAudio, .play = alsa_play,
.cancel = alsa_dropBufferedAudio, .cancel = alsa_cancel,
.close = alsa_closeDevice, .close = alsa_close,
.control = alsa_control .control = alsa_control
}; };