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