s/ad/pd/ in the PluseAudio plugin (I forgot to rename when copying from alsa)

git-svn-id: https://svn.musicpd.org/mpd/trunk@4404 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
J. Alexander Treuman 2006-07-19 16:50:03 +00:00
parent 2728853ec1
commit 3187eb300b

View File

@ -56,28 +56,28 @@ static PulseData * newPulseData()
return ret;
}
static void freePulseData(PulseData * ad)
static void freePulseData(PulseData * pd)
{
if (ad->server) free(ad->server);
if (ad->sink) free(ad->sink);
free(ad);
if (pd->server) free(pd->server);
if (pd->sink) free(pd->sink);
free(pd);
}
static int pulse_initDriver(AudioOutput * audioOutput, ConfigParam * param)
{
BlockParam * server = NULL;
BlockParam * sink = NULL;
PulseData * ad;
PulseData * pd;
if (param) {
server = getBlockParam(param, "server");
sink = getBlockParam(param, "sink");
}
ad = newPulseData();
ad->server = server ? strdup(server->value) : NULL;
ad->sink = sink ? strdup(sink->value) : NULL;
audioOutput->data = ad;
pd = newPulseData();
pd->server = server ? strdup(server->value) : NULL;
pd->sink = sink ? strdup(sink->value) : NULL;
audioOutput->data = pd;
return 0;
}
@ -112,21 +112,21 @@ static int pulse_testDefault()
static int pulse_openDevice(AudioOutput * audioOutput)
{
PulseData * ad;
PulseData * pd;
AudioFormat * audioFormat;
pa_sample_spec ss;
time_t t;
int error;
t = time(NULL);
ad = audioOutput->data;
pd = audioOutput->data;
audioFormat = &audioOutput->outAudioFormat;
if (ad->connAttempts != 0 &&
(t - ad->lastAttempt) < CONN_ATTEMPT_INTERVAL) return -1;
if (pd->connAttempts != 0 &&
(t - pd->lastAttempt) < CONN_ATTEMPT_INTERVAL) return -1;
ad->connAttempts++;
ad->lastAttempt = t;
pd->connAttempts++;
pd->lastAttempt = t;
if (audioFormat->bits != 16) {
ERROR("PulseAudio doesn't support %i bit audio\n",
@ -138,17 +138,17 @@ static int pulse_openDevice(AudioOutput * audioOutput)
ss.rate = audioFormat->sampleRate;
ss.channels = audioFormat->channels;
ad->s = pa_simple_new(ad->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
ad->sink, audioOutput->name, &ss, NULL, NULL,
pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
pd->sink, audioOutput->name, &ss, NULL, NULL,
&error);
if (!ad->s) {
if (!pd->s) {
ERROR("Cannot connect to server in PulseAudio output " \
"\"%s\" (attempt %i): %s\n", audioOutput->name,
ad->connAttempts, pa_strerror(error));
pd->connAttempts, pa_strerror(error));
return -1;
}
ad->connAttempts = 0;
pd->connAttempts = 0;
audioOutput->open = 1;
DEBUG("PulseAudio output \"%s\" connected and playing %i bit, %i " \
@ -160,23 +160,23 @@ static int pulse_openDevice(AudioOutput * audioOutput)
static void pulse_dropBufferedAudio(AudioOutput * audioOutput)
{
PulseData * ad;
PulseData * pd;
int error;
ad = audioOutput->data;
if (pa_simple_flush(ad->s, &error) < 0)
pd = audioOutput->data;
if (pa_simple_flush(pd->s, &error) < 0)
WARNING("Flush failed in PulseAudio output \"%s\": %s\n",
audioOutput->name, pa_strerror(error));
}
static void pulse_closeDevice(AudioOutput * audioOutput)
{
PulseData * ad;
PulseData * pd;
ad = audioOutput->data;
if (ad->s) {
pa_simple_drain(ad->s, NULL);
pa_simple_free(ad->s);
pd = audioOutput->data;
if (pd->s) {
pa_simple_drain(pd->s, NULL);
pa_simple_free(pd->s);
}
audioOutput->open = 0;
@ -185,12 +185,12 @@ static void pulse_closeDevice(AudioOutput * audioOutput)
static int pulse_playAudio(AudioOutput * audioOutput, char * playChunk,
int size)
{
PulseData * ad;
PulseData * pd;
int error;
ad = audioOutput->data;
pd = audioOutput->data;
if (pa_simple_write(ad->s, playChunk, size, &error) < 0) {
if (pa_simple_write(pd->s, playChunk, size, &error) < 0) {
ERROR("PulseAudio output \"%s\" disconnecting due to write " \
"error: %s\n", audioOutput->name, pa_strerror(error));
pulse_closeDevice(audioOutput);