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