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authorMax Kellermann <max@duempel.org>2009-01-25 13:05:16 +0100
committerMax Kellermann <max@duempel.org>2009-01-25 13:05:16 +0100
commitd887b6353f07932cbc66fa1ccdb33c1c03a4ab0b (patch)
tree415b22594bc3a8f26d046d21538f36c9f10a6054 /src/output
parent27baf6913ea2415300c8d6650773e5eb10a01943 (diff)
downloadmpd-d887b6353f07932cbc66fa1ccdb33c1c03a4ab0b.tar.gz
mpd-d887b6353f07932cbc66fa1ccdb33c1c03a4ab0b.tar.xz
mpd-d887b6353f07932cbc66fa1ccdb33c1c03a4ab0b.zip
alsa: no CamelCase
Renamed types, functions, variables.
Diffstat (limited to 'src/output')
-rw-r--r--src/output/alsa_plugin.c196
1 files changed, 103 insertions, 93 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index bd6ccd83b..e76bf82f0 100644
--- a/src/output/alsa_plugin.c
+++ b/src/output/alsa_plugin.c
@@ -40,38 +40,38 @@ enum {
typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
snd_pcm_uframes_t size);
-typedef struct _AlsaData {
+struct alsa_data {
char *device;
/** the mode flags passed to snd_pcm_open */
int mode;
- snd_pcm_t *pcmHandle;
+ snd_pcm_t *pcm;
alsa_writei_t *writei;
unsigned int buffer_time;
unsigned int period_time;
- int sampleSize;
- int useMmap;
+ int frame_size;
+ bool use_mmap;
struct mixer mixer;
-
-} AlsaData;
+};
static const char *
-alsa_device(const AlsaData *ad)
+alsa_device(const struct alsa_data *ad)
{
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->mode = 0;
- ret->pcmHandle = NULL;
+ ret->pcm = NULL;
ret->writei = snd_pcm_writei;
- ret->useMmap = 0;
+ ret->use_mmap = false;
ret->buffer_time = MPD_ALSA_BUFFER_TIME_US;
ret->period_time = MPD_ALSA_PERIOD_TIME_US;
@@ -81,7 +81,8 @@ static AlsaData *newAlsaData(void)
return ret;
}
-static void freeAlsaData(AlsaData * ad)
+static void
+alsa_data_free(struct alsa_data *ad)
{
g_free(ad->device);
mixer_finish(&ad->mixer);
@@ -89,11 +90,11 @@ static void freeAlsaData(AlsaData * ad)
}
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->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",
MPD_ALSA_BUFFER_TIME_US);
@@ -117,13 +118,13 @@ alsa_configure(AlsaData *ad, struct config_param *param)
}
static void *
-alsa_initDriver(G_GNUC_UNUSED struct audio_output *ao,
- G_GNUC_UNUSED const struct audio_format *audio_format,
- struct config_param *param)
+alsa_init(G_GNUC_UNUSED struct audio_output *ao,
+ G_GNUC_UNUSED const struct audio_format *audio_format,
+ struct config_param *param)
{
/* no need for pthread_once thread-safety when reading config */
static int free_global_registered;
- AlsaData *ad = newAlsaData();
+ struct alsa_data *ad = alsa_data_new();
if (!free_global_registered) {
atexit((void(*)(void))snd_config_update_free_global);
@@ -138,14 +139,16 @@ alsa_initDriver(G_GNUC_UNUSED struct audio_output *ao,
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;
@@ -161,7 +164,8 @@ static bool alsa_testDefault(void)
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) {
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;
}
-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_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
- unsigned int sample_rate = audioFormat->sample_rate;
- unsigned int channels = audioFormat->channels;
+ unsigned int sample_rate = audio_format->sample_rate;
+ unsigned int channels = audio_format->channels;
snd_pcm_uframes_t alsa_buffer_size;
snd_pcm_uframes_t alsa_period_size;
int err;
@@ -190,14 +195,14 @@ static bool alsa_openDevice(void *data, struct audio_format *audioFormat)
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",
- 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);
if (err < 0) {
- ad->pcmHandle = NULL;
+ ad->pcm = NULL;
goto error;
}
@@ -207,72 +212,72 @@ configure_hw:
snd_pcm_hw_params_alloca(&hwparams);
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)
goto error;
- if (ad->useMmap) {
- err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
+ if (ad->use_mmap) {
+ err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
SND_PCM_ACCESS_MMAP_INTERLEAVED);
if (err < 0) {
g_warning("Cannot set mmap'ed mode on ALSA device \"%s\": %s\n",
alsa_device(ad), snd_strerror(-err));
g_warning("Falling back to direct write mode\n");
- ad->useMmap = 0;
+ ad->use_mmap = false;
} else
ad->writei = snd_pcm_mmap_writei;
}
- if (!ad->useMmap) {
+ if (!ad->use_mmap) {
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);
if (err < 0)
goto error;
ad->writei = snd_pcm_writei;
}
- err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat);
- if (err == -EINVAL && audioFormat->bits != 16) {
+ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, bitformat);
+ if (err == -EINVAL && audio_format->bits != 16) {
/* 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);
if (err == 0) {
g_debug("ALSA device \"%s\": converting %u bit to 16 bit\n",
- alsa_device(ad), audioFormat->bits);
- audioFormat->bits = 16;
+ alsa_device(ad), audio_format->bits);
+ audio_format->bits = 16;
}
}
if (err < 0) {
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;
}
- err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams,
+ err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
&channels);
if (err < 0) {
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));
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);
if (err < 0 || sample_rate == 0) {
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;
}
- audioFormat->sample_rate = sample_rate;
+ audio_format->sample_rate = sample_rate;
if (ad->buffer_time > 0) {
buffer_time = ad->buffer_time;
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);
if (err < 0)
goto error;
@@ -281,14 +286,14 @@ configure_hw:
if (period_time_ro > 0) {
period_time = period_time_ro;
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);
if (err < 0)
goto error;
}
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) {
period_time_ro = period_time_ro >> 1;
goto configure_hw;
@@ -312,32 +317,32 @@ configure_hw:
snd_pcm_sw_params_alloca(&swparams);
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)
goto error;
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_period_size);
if (err < 0)
goto error;
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);
if (err < 0)
goto error;
cmd = "snd_pcm_sw_params";
- err = snd_pcm_sw_params(ad->pcmHandle, swparams);
+ err = snd_pcm_sw_params(ad->pcm, swparams);
if (err < 0)
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",
- alsa_device(ad), audioFormat->bits, channels, sample_rate);
+ alsa_device(ad), audio_format->bits, channels, sample_rate);
return true;
@@ -350,13 +355,14 @@ error:
alsa_device(ad), snd_strerror(-err));
}
fail:
- if (ad->pcmHandle)
- snd_pcm_close(ad->pcmHandle);
- ad->pcmHandle = NULL;
+ if (ad->pcm)
+ snd_pcm_close(ad->pcm);
+ ad->pcm = NULL;
return false;
}
-static int alsa_errorRecovery(AlsaData * ad, int err)
+static int
+alsa_recover(struct alsa_data *ad, int err)
{
if (err == -EPIPE) {
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));
}
- switch (snd_pcm_state(ad->pcmHandle)) {
+ switch (snd_pcm_state(ad->pcm)) {
case SND_PCM_STATE_PAUSED:
- err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0);
+ err = snd_pcm_pause(ad->pcm, /* disable */ 0);
break;
case SND_PCM_STATE_SUSPENDED:
- err = snd_pcm_resume(ad->pcmHandle);
+ err = snd_pcm_resume(ad->pcm);
if (err == -EAGAIN)
return 0;
/* fall-through to snd_pcm_prepare: */
case SND_PCM_STATE_SETUP:
case SND_PCM_STATE_XRUN:
- err = snd_pcm_prepare(ad->pcmHandle);
+ err = snd_pcm_prepare(ad->pcm);
break;
case SND_PCM_STATE_DISCONNECTED:
/* so alsa_closeDevice won't try to drain: */
- snd_pcm_close(ad->pcmHandle);
- ad->pcmHandle = NULL;
+ snd_pcm_close(ad->pcm);
+ ad->pcm = NULL;
break;
/* this is no error, so just keep running */
case SND_PCM_STATE_RUNNING:
@@ -394,52 +400,56 @@ static int alsa_errorRecovery(AlsaData * ad, int 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 (snd_pcm_state(ad->pcmHandle) == SND_PCM_STATE_RUNNING) {
- snd_pcm_drain(ad->pcmHandle);
- }
- snd_pcm_close(ad->pcmHandle);
- ad->pcmHandle = NULL;
+ if (ad->pcm != NULL) {
+ if (snd_pcm_state(ad->pcm) == SND_PCM_STATE_RUNNING)
+ snd_pcm_drain(ad->pcm);
+
+ snd_pcm_close(ad->pcm);
+ ad->pcm = NULL;
}
+
mixer_close(&ad->mixer);
}
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;
- size /= ad->sampleSize;
+ size /= ad->frame_size;
while (size > 0) {
- ret = ad->writei(ad->pcmHandle, playChunk, size);
+ ret = ad->writei(ad->pcm, chunk, size);
if (ret == -EAGAIN || ret == -EINTR)
continue;
if (ret < 0) {
- if (alsa_errorRecovery(ad, ret) < 0) {
+ if (alsa_recover(ad, ret) < 0) {
g_warning("closing ALSA device \"%s\" due to write "
"error: %s\n",
alsa_device(ad), snd_strerror(-errno));
return false;
}
+
continue;
}
- playChunk += ret * ad->sampleSize;
+ chunk += ret * ad->frame_size;
size -= ret;
}
@@ -449,18 +459,18 @@ alsa_playAudio(void *data, const char *playChunk, size_t size)
static bool
alsa_control(void *data, int cmd, void *arg)
{
- AlsaData *ad = data;
+ struct alsa_data *ad = data;
return mixer_control(&ad->mixer, cmd, arg);
}
const struct audio_output_plugin alsaPlugin = {
.name = "alsa",
- .test_default_device = alsa_testDefault,
- .init = alsa_initDriver,
- .finish = alsa_finishDriver,
- .open = alsa_openDevice,
- .play = alsa_playAudio,
- .cancel = alsa_dropBufferedAudio,
- .close = alsa_closeDevice,
+ .test_default_device = alsa_test_default_device,
+ .init = alsa_init,
+ .finish = alsa_finish,
+ .open = alsa_open,
+ .play = alsa_play,
+ .cancel = alsa_cancel,
+ .close = alsa_close,
.control = alsa_control
};