diff options
Diffstat (limited to 'src/modules/alsa')
27 files changed, 849 insertions, 141 deletions
diff --git a/src/modules/alsa/alsa-mixer.c b/src/modules/alsa/alsa-mixer.c index 6f21e103..a5515e1b 100644 --- a/src/modules/alsa/alsa-mixer.c +++ b/src/modules/alsa/alsa-mixer.c @@ -479,6 +479,7 @@ static int element_get_volume(pa_alsa_element *e, snd_mixer_t *m, const pa_chann snd_mixer_elem_t *me; snd_mixer_selem_channel_id_t c; pa_channel_position_mask_t mask = 0; + pa_volume_t max_channel_volume = PA_VOLUME_MUTED; unsigned k; pa_assert(m); @@ -545,6 +546,9 @@ static int element_get_volume(pa_alsa_element *e, snd_mixer_t *m, const pa_chann f = from_alsa_volume(value, e->min_volume, e->max_volume); } + if (f > max_channel_volume) + max_channel_volume = f; + for (k = 0; k < cm->channels; k++) if (e->masks[c][e->n_channels-1] & PA_CHANNEL_POSITION_MASK(cm->map[k])) if (v->values[k] < f) @@ -555,7 +559,7 @@ static int element_get_volume(pa_alsa_element *e, snd_mixer_t *m, const pa_chann for (k = 0; k < cm->channels; k++) if (!(mask & PA_CHANNEL_POSITION_MASK(cm->map[k]))) - v->values[k] = PA_VOLUME_NORM; + v->values[k] = max_channel_volume; return 0; } @@ -677,6 +681,7 @@ static int element_set_volume(pa_alsa_element *e, snd_mixer_t *m, const pa_chann snd_mixer_elem_t *me; snd_mixer_selem_channel_id_t c; pa_channel_position_mask_t mask = 0; + pa_volume_t max_channel_volume = PA_VOLUME_MUTED; unsigned k; pa_assert(m); @@ -696,11 +701,21 @@ static int element_set_volume(pa_alsa_element *e, snd_mixer_t *m, const pa_chann for (c = 0; c <= SND_MIXER_SCHN_LAST; c++) { int r; pa_volume_t f = PA_VOLUME_MUTED; + pa_bool_t found = FALSE; for (k = 0; k < cm->channels; k++) - if (e->masks[c][e->n_channels-1] & PA_CHANNEL_POSITION_MASK(cm->map[k])) + if (e->masks[c][e->n_channels-1] & PA_CHANNEL_POSITION_MASK(cm->map[k])) { + found = TRUE; if (v->values[k] > f) f = v->values[k]; + } + + if (!found) { + /* Hmm, so this channel does not exist in the volume + * struct, so let's bind it to the overall max of the + * volume. */ + f = pa_cvolume_max(v); + } if (e->has_dB) { long value = to_alsa_dB(f); @@ -756,6 +771,9 @@ static int element_set_volume(pa_alsa_element *e, snd_mixer_t *m, const pa_chann f = from_alsa_volume(value, e->min_volume, e->max_volume); } + if (f > max_channel_volume) + max_channel_volume = f; + for (k = 0; k < cm->channels; k++) if (e->masks[c][e->n_channels-1] & PA_CHANNEL_POSITION_MASK(cm->map[k])) if (rv.values[k] < f) @@ -766,7 +784,7 @@ static int element_set_volume(pa_alsa_element *e, snd_mixer_t *m, const pa_chann for (k = 0; k < cm->channels; k++) if (!(mask & PA_CHANNEL_POSITION_MASK(cm->map[k]))) - rv.values[k] = PA_VOLUME_NORM; + rv.values[k] = max_channel_volume; *v = rv; return 0; @@ -2930,7 +2948,7 @@ static int profile_verify(pa_alsa_profile *p) { char **in; pa_bool_t duplicate = FALSE; - for (in = p->output_mapping_names; *in; in++) + for (in = name + 1; *in; in++) if (pa_streq(*name, *in)) { duplicate = TRUE; break; @@ -2945,6 +2963,9 @@ static int profile_verify(pa_alsa_profile *p) { } pa_idxset_put(p->output_mappings, m, NULL); + + if (p->supported) + m->supported++; } pa_xstrfreev(p->output_mapping_names); @@ -2963,7 +2984,7 @@ static int profile_verify(pa_alsa_profile *p) { char **in; pa_bool_t duplicate = FALSE; - for (in = p->input_mapping_names; *in; in++) + for (in = name + 1; *in; in++) if (pa_streq(*name, *in)) { duplicate = TRUE; break; @@ -2978,6 +2999,9 @@ static int profile_verify(pa_alsa_profile *p) { } pa_idxset_put(p->input_mappings, m, NULL); + + if (p->supported) + m->supported++; } pa_xstrfreev(p->input_mapping_names); diff --git a/src/modules/alsa/alsa-sink.c b/src/modules/alsa/alsa-sink.c index 2226bc6f..e7925902 100644 --- a/src/modules/alsa/alsa-sink.c +++ b/src/modules/alsa/alsa-sink.c @@ -32,16 +32,18 @@ #include <valgrind/memcheck.h> #endif -#include <pulse/xmalloc.h> -#include <pulse/util.h> -#include <pulse/timeval.h> #include <pulse/i18n.h> +#include <pulse/rtclock.h> +#include <pulse/timeval.h> +#include <pulse/util.h> +#include <pulse/xmalloc.h> #include <pulsecore/core.h> #include <pulsecore/module.h> #include <pulsecore/memchunk.h> #include <pulsecore/sink.h> #include <pulsecore/modargs.h> +#include <pulsecore/core-rtclock.h> #include <pulsecore/core-util.h> #include <pulsecore/sample-util.h> #include <pulsecore/log.h> @@ -50,7 +52,6 @@ #include <pulsecore/core-error.h> #include <pulsecore/thread-mq.h> #include <pulsecore/rtpoll.h> -#include <pulsecore/rtclock.h> #include <pulsecore/time-smoother.h> #include <modules/reserve-wrap.h> @@ -168,10 +169,10 @@ static int reserve_init(struct userdata *u, const char *dname) { if (pa_in_system_mode()) return 0; - /* We are resuming, try to lock the device */ if (!(rname = pa_alsa_get_reserve_name(dname))) return 0; + /* We are resuming, try to lock the device */ u->reserve = pa_reserve_wrapper_get(u->core, rname); pa_xfree(rname); @@ -221,7 +222,6 @@ static int reserve_monitor_init(struct userdata *u, const char *dname) { if (pa_in_system_mode()) return 0; - /* We are resuming, try to lock the device */ if (!(rname = pa_alsa_get_reserve_name(dname))) return 0; @@ -494,6 +494,9 @@ static int mmap_write(struct userdata *u, pa_usec_t *sleep_usec, pa_bool_t polle if (frames > pa_mempool_block_size_max(u->sink->core->mempool)/u->frame_size) frames = pa_mempool_block_size_max(u->sink->core->mempool)/u->frame_size; + if (frames == 0) + break; + /* Check these are multiples of 8 bit */ pa_assert((areas[0].first & 7) == 0); pa_assert((areas[0].step & 7)== 0); @@ -631,7 +634,8 @@ static int unix_write(struct userdata *u, pa_usec_t *sleep_usec, pa_bool_t polle frames = snd_pcm_writei(u->pcm_handle, (const uint8_t*) p + u->memchunk.index, (snd_pcm_uframes_t) frames); pa_memblock_release(u->memchunk.memblock); - pa_assert(frames != 0); + if (frames == 0) + break; if (PA_UNLIKELY(frames < 0)) { @@ -707,7 +711,7 @@ static void update_smoother(struct userdata *u) { /* Hmm, if the timestamp is 0, then it wasn't set and we take the current time */ if (now1 <= 0) - now1 = pa_rtclock_usec(); + now1 = pa_rtclock_now(); now2 = pa_bytes_to_usec((uint64_t) position, &u->sink->sample_spec); @@ -721,7 +725,7 @@ static pa_usec_t sink_get_latency(struct userdata *u) { pa_assert(u); - now1 = pa_rtclock_usec(); + now1 = pa_rtclock_now(); now2 = pa_smoother_get(u->smoother, now1); delay = (int64_t) pa_bytes_to_usec(u->write_count, &u->sink->sample_spec) - (int64_t) now2; @@ -752,7 +756,7 @@ static int suspend(struct userdata *u) { pa_assert(u); pa_assert(u->pcm_handle); - pa_smoother_pause(u->smoother, pa_rtclock_usec()); + pa_smoother_pause(u->smoother, pa_rtclock_now()); /* Let's suspend -- we don't call snd_pcm_drain() here since that might * take awfully long with our long buffer sizes today. */ @@ -838,7 +842,6 @@ static int unsuspend(struct userdata *u) { pa_log_info("Trying resume..."); - snd_config_update_free_global(); if ((err = snd_pcm_open(&u->pcm_handle, u->device_name, SND_PCM_STREAM_PLAYBACK, /*SND_PCM_NONBLOCK|*/ SND_PCM_NO_AUTO_RESAMPLE| @@ -1213,7 +1216,6 @@ static void thread_func(void *userdata) { pa_make_realtime(u->core->realtime_priority); pa_thread_mq_install(&u->thread_mq); - pa_rtpoll_install(u->rtpoll); for (;;) { int ret; @@ -1247,7 +1249,7 @@ static void thread_func(void *userdata) { pa_log_info("Starting playback."); snd_pcm_start(u->pcm_handle); - pa_smoother_resume(u->smoother, pa_rtclock_usec(), TRUE); + pa_smoother_resume(u->smoother, pa_rtclock_now(), TRUE); } update_smoother(u); @@ -1276,7 +1278,7 @@ static void thread_func(void *userdata) { /* Convert from the sound card time domain to the * system time domain */ - cusec = pa_smoother_translate(u->smoother, pa_rtclock_usec(), sleep_usec); + cusec = pa_smoother_translate(u->smoother, pa_rtclock_now(), sleep_usec); /* pa_log_debug("Waking up in %0.2fms (system clock).", (double) cusec / PA_USEC_PER_MSEC); */ @@ -1335,7 +1337,7 @@ finish: pa_log_debug("Thread shutting down"); } -static void set_sink_name(pa_sink_new_data *data, pa_modargs *ma, const char *device_id, const char *device_name) { +static void set_sink_name(pa_sink_new_data *data, pa_modargs *ma, const char *device_id, const char *device_name, pa_alsa_mapping *mapping) { const char *n; char *t; @@ -1356,7 +1358,11 @@ static void set_sink_name(pa_sink_new_data *data, pa_modargs *ma, const char *de data->namereg_fail = FALSE; } - t = pa_sprintf_malloc("alsa_output.%s", n); + if (mapping) + t = pa_sprintf_malloc("alsa_output.%s.%s", n, mapping->name); + else + t = pa_sprintf_malloc("alsa_output.%s", n); + pa_sink_new_data_set_name(data, t); pa_xfree(t); } @@ -1578,7 +1584,7 @@ pa_sink *pa_alsa_sink_new(pa_module *m, pa_modargs *ma, const char*driver, pa_ca TRUE, TRUE, 5, - pa_rtclock_usec(), + pa_rtclock_now(), TRUE); dev_id = pa_modargs_get_value( @@ -1679,7 +1685,7 @@ pa_sink *pa_alsa_sink_new(pa_module *m, pa_modargs *ma, const char*driver, pa_ca data.driver = driver; data.module = m; data.card = card; - set_sink_name(&data, ma, dev_id, u->device_name); + set_sink_name(&data, ma, dev_id, u->device_name, mapping); pa_sink_new_data_set_sample_spec(&data, &ss); pa_sink_new_data_set_channel_map(&data, &map); diff --git a/src/modules/alsa/alsa-source.c b/src/modules/alsa/alsa-source.c index f2e4e234..41bb768b 100644 --- a/src/modules/alsa/alsa-source.c +++ b/src/modules/alsa/alsa-source.c @@ -28,10 +28,11 @@ #include <asoundlib.h> -#include <pulse/xmalloc.h> -#include <pulse/util.h> -#include <pulse/timeval.h> #include <pulse/i18n.h> +#include <pulse/rtclock.h> +#include <pulse/timeval.h> +#include <pulse/util.h> +#include <pulse/xmalloc.h> #include <pulsecore/core-error.h> #include <pulsecore/core.h> @@ -39,6 +40,7 @@ #include <pulsecore/memchunk.h> #include <pulsecore/sink.h> #include <pulsecore/modargs.h> +#include <pulsecore/core-rtclock.h> #include <pulsecore/core-util.h> #include <pulsecore/sample-util.h> #include <pulsecore/log.h> @@ -48,7 +50,6 @@ #include <pulsecore/thread-mq.h> #include <pulsecore/rtpoll.h> #include <pulsecore/time-smoother.h> -#include <pulsecore/rtclock.h> #include <modules/reserve-wrap.h> @@ -472,6 +473,9 @@ static int mmap_read(struct userdata *u, pa_usec_t *sleep_usec, pa_bool_t polled if (frames > pa_mempool_block_size_max(u->source->core->mempool)/u->frame_size) frames = pa_mempool_block_size_max(u->source->core->mempool)/u->frame_size; + if (frames == 0) + break; + /* Check these are multiples of 8 bit */ pa_assert((areas[0].first & 7) == 0); pa_assert((areas[0].step & 7)== 0); @@ -598,7 +602,10 @@ static int unix_read(struct userdata *u, pa_usec_t *sleep_usec, pa_bool_t polled frames = snd_pcm_readi(u->pcm_handle, (uint8_t*) p, (snd_pcm_uframes_t) frames); pa_memblock_release(chunk.memblock); - pa_assert(frames != 0); + if (frames == 0) { + pa_memblock_unref(chunk.memblock); + break; + } if (PA_UNLIKELY(frames < 0)) { pa_memblock_unref(chunk.memblock); @@ -669,7 +676,7 @@ static void update_smoother(struct userdata *u) { /* Hmm, if the timestamp is 0, then it wasn't set and we take the current time */ if (now1 <= 0) - now1 = pa_rtclock_usec(); + now1 = pa_rtclock_now(); now2 = pa_bytes_to_usec(position, &u->source->sample_spec); @@ -682,7 +689,7 @@ static pa_usec_t source_get_latency(struct userdata *u) { pa_assert(u); - now1 = pa_rtclock_usec(); + now1 = pa_rtclock_now(); now2 = pa_smoother_get(u->smoother, now1); delay = (int64_t) now2 - (int64_t) pa_bytes_to_usec(u->read_count, &u->source->sample_spec); @@ -707,7 +714,7 @@ static int suspend(struct userdata *u) { pa_assert(u); pa_assert(u->pcm_handle); - pa_smoother_pause(u->smoother, pa_rtclock_usec()); + pa_smoother_pause(u->smoother, pa_rtclock_now()); /* Let's suspend */ snd_pcm_close(u->pcm_handle); @@ -787,8 +794,6 @@ static int unsuspend(struct userdata *u) { pa_log_info("Trying resume..."); - snd_config_update_free_global(); - if ((err = snd_pcm_open(&u->pcm_handle, u->device_name, SND_PCM_STREAM_CAPTURE, /*SND_PCM_NONBLOCK|*/ SND_PCM_NO_AUTO_RESAMPLE| @@ -835,7 +840,7 @@ static int unsuspend(struct userdata *u) { /* FIXME: We need to reload the volume somehow */ snd_pcm_start(u->pcm_handle); - pa_smoother_resume(u->smoother, pa_rtclock_usec(), TRUE); + pa_smoother_resume(u->smoother, pa_rtclock_now(), TRUE); pa_log_info("Resumed successfully..."); @@ -1096,7 +1101,6 @@ static void thread_func(void *userdata) { pa_make_realtime(u->core->realtime_priority); pa_thread_mq_install(&u->thread_mq); - pa_rtpoll_install(u->rtpoll); for (;;) { int ret; @@ -1133,7 +1137,7 @@ static void thread_func(void *userdata) { /* Convert from the sound card time domain to the * system time domain */ - cusec = pa_smoother_translate(u->smoother, pa_rtclock_usec(), sleep_usec); + cusec = pa_smoother_translate(u->smoother, pa_rtclock_now(), sleep_usec); /* pa_log_debug("Waking up in %0.2fms (system clock).", (double) cusec / PA_USEC_PER_MSEC); */ @@ -1187,7 +1191,7 @@ finish: pa_log_debug("Thread shutting down"); } -static void set_source_name(pa_source_new_data *data, pa_modargs *ma, const char *device_id, const char *device_name) { +static void set_source_name(pa_source_new_data *data, pa_modargs *ma, const char *device_id, const char *device_name, pa_alsa_mapping *mapping) { const char *n; char *t; @@ -1208,7 +1212,11 @@ static void set_source_name(pa_source_new_data *data, pa_modargs *ma, const char data->namereg_fail = FALSE; } - t = pa_sprintf_malloc("alsa_input.%s", n); + if (mapping) + t = pa_sprintf_malloc("alsa_input.%s.%s", n, mapping->name); + else + t = pa_sprintf_malloc("alsa_input.%s", n); + pa_source_new_data_set_name(data, t); pa_xfree(t); } @@ -1429,7 +1437,7 @@ pa_source *pa_alsa_source_new(pa_module *m, pa_modargs *ma, const char*driver, p TRUE, TRUE, 5, - pa_rtclock_usec(), + pa_rtclock_now(), FALSE); dev_id = pa_modargs_get_value( @@ -1528,7 +1536,7 @@ pa_source *pa_alsa_source_new(pa_module *m, pa_modargs *ma, const char*driver, p data.driver = driver; data.module = m; data.card = card; - set_source_name(&data, ma, dev_id, u->device_name); + set_source_name(&data, ma, dev_id, u->device_name, mapping); pa_source_new_data_set_sample_spec(&data, &ss); pa_source_new_data_set_channel_map(&data, &map); diff --git a/src/modules/alsa/alsa-util.c b/src/modules/alsa/alsa-util.c index 0204c28b..1f3e5dcd 100644 --- a/src/modules/alsa/alsa-util.c +++ b/src/modules/alsa/alsa-util.c @@ -735,38 +735,43 @@ static void alsa_error_handler(const char *file, int line, const char *function, static pa_atomic_t n_error_handler_installed = PA_ATOMIC_INIT(0); -void pa_alsa_redirect_errors_inc(void) { +void pa_alsa_refcnt_inc(void) { /* This is not really thread safe, but we do our best */ if (pa_atomic_inc(&n_error_handler_installed) == 0) snd_lib_error_set_handler(alsa_error_handler); } -void pa_alsa_redirect_errors_dec(void) { +void pa_alsa_refcnt_dec(void) { int r; pa_assert_se((r = pa_atomic_dec(&n_error_handler_installed)) >= 1); - if (r == 1) + if (r == 1) { snd_lib_error_set_handler(NULL); + snd_config_update_free_global(); + } } pa_bool_t pa_alsa_init_description(pa_proplist *p) { - const char *s; + const char *d, *k; pa_assert(p); if (pa_device_init_description(p)) return TRUE; - if ((s = pa_proplist_gets(p, "alsa.card_name"))) { - pa_proplist_sets(p, PA_PROP_DEVICE_DESCRIPTION, s); - return TRUE; - } + if (!(d = pa_proplist_gets(p, "alsa.card_name"))) + d = pa_proplist_gets(p, "alsa.name"); - if ((s = pa_proplist_gets(p, "alsa.name"))) { - pa_proplist_sets(p, PA_PROP_DEVICE_DESCRIPTION, s); - return TRUE; - } + if (!d) + return FALSE; + + k = pa_proplist_gets(p, PA_PROP_DEVICE_PROFILE_DESCRIPTION); + + if (d && k) + pa_proplist_setf(p, PA_PROP_DEVICE_DESCRIPTION, _("%s %s"), d, k); + else if (d) + pa_proplist_sets(p, PA_PROP_DEVICE_DESCRIPTION, d); return FALSE; } diff --git a/src/modules/alsa/alsa-util.h b/src/modules/alsa/alsa-util.h index c2f0e5b7..830a922e 100644 --- a/src/modules/alsa/alsa-util.h +++ b/src/modules/alsa/alsa-util.h @@ -114,8 +114,8 @@ snd_pcm_t *pa_alsa_open_by_template( void pa_alsa_dump(pa_log_level_t level, snd_pcm_t *pcm); void pa_alsa_dump_status(snd_pcm_t *pcm); -void pa_alsa_redirect_errors_inc(void); -void pa_alsa_redirect_errors_dec(void); +void pa_alsa_refcnt_inc(void); +void pa_alsa_refcnt_dec(void); void pa_alsa_init_proplist_pcm_info(pa_core *c, pa_proplist *p, snd_pcm_info_t *pcm_info); void pa_alsa_init_proplist_card(pa_core *c, pa_proplist *p, int card); diff --git a/src/modules/alsa/mixer/paths/analog-input-aux.conf b/src/modules/alsa/mixer/paths/analog-input-aux.conf index 8f480567..db78eb48 100644 --- a/src/modules/alsa/mixer/paths/analog-input-aux.conf +++ b/src/modules/alsa/mixer/paths/analog-input-aux.conf @@ -1,4 +1,22 @@ -# For devices, where we have an Aux element +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; For devices where an 'Aux' element exists +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 90 @@ -29,4 +47,16 @@ override-map.2 = all-left,all-right switch = off volume = off +[Element Mic/Line] +switch = off +volume = off + +[Element TV Tuner] +switch = off +volume = off + +[Element FM] +switch = off +volume = off + .include analog-input.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-fm.conf b/src/modules/alsa/mixer/paths/analog-input-fm.conf index 0f78f39f..baf674aa 100644 --- a/src/modules/alsa/mixer/paths/analog-input-fm.conf +++ b/src/modules/alsa/mixer/paths/analog-input-fm.conf @@ -1,4 +1,22 @@ -# For devices where we have an FM element +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; For devices where an 'FM' element exists +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 70 diff --git a/src/modules/alsa/mixer/paths/analog-input-linein.conf b/src/modules/alsa/mixer/paths/analog-input-linein.conf index b6ba738c..4be5722d 100644 --- a/src/modules/alsa/mixer/paths/analog-input-linein.conf +++ b/src/modules/alsa/mixer/paths/analog-input-linein.conf @@ -1,4 +1,22 @@ -# For devices, where we have a Line element +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; For devices where a 'Line' element exists +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 90 diff --git a/src/modules/alsa/mixer/paths/analog-input-mic-line.conf b/src/modules/alsa/mixer/paths/analog-input-mic-line.conf index 7d4addf7..f7f30854 100644 --- a/src/modules/alsa/mixer/paths/analog-input-mic-line.conf +++ b/src/modules/alsa/mixer/paths/analog-input-mic-line.conf @@ -1,4 +1,22 @@ -# For devices where we have a Mic/Lineb element +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; For devices where a 'Mic/Line' element exists +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 90 @@ -42,3 +60,4 @@ switch = off volume = off .include analog-input.conf.common +.include analog-input-mic.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input-mic.conf b/src/modules/alsa/mixer/paths/analog-input-mic.conf index 004cd24a..2a36f2f3 100644 --- a/src/modules/alsa/mixer/paths/analog-input-mic.conf +++ b/src/modules/alsa/mixer/paths/analog-input-mic.conf @@ -1,4 +1,22 @@ -# For devices where we have a Mic element +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; For devices where a 'Mic' element exists +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 100 diff --git a/src/modules/alsa/mixer/paths/analog-input-mic.conf.common b/src/modules/alsa/mixer/paths/analog-input-mic.conf.common index d67ee4e3..b35e7af8 100644 --- a/src/modules/alsa/mixer/paths/analog-input-mic.conf.common +++ b/src/modules/alsa/mixer/paths/analog-input-mic.conf.common @@ -1,3 +1,23 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; Common element for all microphone inputs +; +; See analog-output.conf.common for an explanation on the directives + ;;; 'Mic Select' [Element Mic Select] @@ -15,6 +35,7 @@ priority = 19 [Element Mic Boost (+20dB)] switch = select +volume = merge [Option Mic Boost (+20dB):on] name = input-boost-on @@ -24,6 +45,7 @@ name = input-boost-off [Element Mic Boost] switch = select +volume = merge [Option Mic Boost:on] name = input-boost-on diff --git a/src/modules/alsa/mixer/paths/analog-input-tvtuner.conf b/src/modules/alsa/mixer/paths/analog-input-tvtuner.conf index ea0a0b72..8531ec70 100644 --- a/src/modules/alsa/mixer/paths/analog-input-tvtuner.conf +++ b/src/modules/alsa/mixer/paths/analog-input-tvtuner.conf @@ -1,4 +1,22 @@ -# For devices, where we have a TV Tuner element +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; For devices where a 'TV Tuner' element exists +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 70 diff --git a/src/modules/alsa/mixer/paths/analog-input-video.conf b/src/modules/alsa/mixer/paths/analog-input-video.conf index 27acc254..74c76f07 100644 --- a/src/modules/alsa/mixer/paths/analog-input-video.conf +++ b/src/modules/alsa/mixer/paths/analog-input-video.conf @@ -1,4 +1,22 @@ -# For devices, where we have a Video element +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; For devices where a 'Video' element exists +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 70 @@ -28,4 +46,16 @@ volume = merge override-map.1 = all override-map.2 = all-left,all-right +[Element Mic/Line] +switch = off +volume = off + +[Element TV Tuner] +switch = off +volume = off + +[Element FM] +switch = off +volume = off + .include analog-input.conf.common diff --git a/src/modules/alsa/mixer/paths/analog-input.conf b/src/modules/alsa/mixer/paths/analog-input.conf index b221bb44..5055f90a 100644 --- a/src/modules/alsa/mixer/paths/analog-input.conf +++ b/src/modules/alsa/mixer/paths/analog-input.conf @@ -1,4 +1,23 @@ -# A fallback for devices that lack seperate Mic/Line/Aux/Video elements +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; A fallback for devices that lack seperate Mic/Line/Aux/Video/TV +; Tuner/FM elements +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 100 diff --git a/src/modules/alsa/mixer/paths/analog-input.conf.common b/src/modules/alsa/mixer/paths/analog-input.conf.common index d34afd04..6728a6ae 100644 --- a/src/modules/alsa/mixer/paths/analog-input.conf.common +++ b/src/modules/alsa/mixer/paths/analog-input.conf.common @@ -1,30 +1,48 @@ -# Mixer path for PulseAudio's ALSA backend. If multiple options by the -# same id are discovered they will be suffixed with a number to -# distuingish them, in the same order they appear here. +# This file is part of PulseAudio. # -# Source selection should use the following names: +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. # -# input -- If we don't know the exact kind of input -# input-microphone -# input-microphone-internal -# input-microphone-external -# input-linein -# input-video -# input-radio -# input-docking-microphone -# input-docking-linein -# input-docking -# -# We explicitly don't want to wrap the following sources: -# -# CD -# Synth/MIDI -# Phone -# Mix -# Digital/SPDIF -# Master -# PC Speaker +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. # +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; Mixer path for PulseAudio's ALSA backend, common elements for all +; input paths. If multiple options by the same id are discovered they +; will be suffixed with a number to distuingish them, in the same +; order they appear here. +; +; Source selection should use the following names: +; +; input -- If we don't know the exact kind of input +; input-microphone +; input-microphone-internal +; input-microphone-external +; input-linein +; input-video +; input-radio +; input-docking-microphone +; input-docking-linein +; input-docking +; +; We explicitly don't want to wrap the following sources: +; +; CD +; Synth/MIDI +; Phone +; Mix +; Digital/SPDIF +; Master +; PC Speaker +; +; See analog-output.conf.common for an explanation on the directives ;;; 'Input Source Select' diff --git a/src/modules/alsa/mixer/paths/analog-output-headphones.conf b/src/modules/alsa/mixer/paths/analog-output-headphones.conf index 1a172d4c..c018e0eb 100644 --- a/src/modules/alsa/mixer/paths/analog-output-headphones.conf +++ b/src/modules/alsa/mixer/paths/analog-output-headphones.conf @@ -1,4 +1,22 @@ -# Path for mixers that have a Headphone slider +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; Path for mixers that have a 'Headphone' control +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 90 diff --git a/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf b/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf index 67031762..7a267890 100644 --- a/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf +++ b/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf @@ -1,5 +1,23 @@ -# Intended for usage in laptops that have a seperate LFE speaker -# connected to the Master mono connector +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; Intended for usage in laptops that have a seperate LFE speaker +; connected to the Master mono connector +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 40 diff --git a/src/modules/alsa/mixer/paths/analog-output-mono.conf b/src/modules/alsa/mixer/paths/analog-output-mono.conf index a23d9b79..f6cb9f8a 100644 --- a/src/modules/alsa/mixer/paths/analog-output-mono.conf +++ b/src/modules/alsa/mixer/paths/analog-output-mono.conf @@ -1,4 +1,22 @@ -# Intended for usage on boards that have a seperate Mono output plug. +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; Intended for usage on boards that have a seperate Mono output plug. +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 50 diff --git a/src/modules/alsa/mixer/paths/analog-output.conf b/src/modules/alsa/mixer/paths/analog-output.conf index 15e703c4..ea108aaf 100644 --- a/src/modules/alsa/mixer/paths/analog-output.conf +++ b/src/modules/alsa/mixer/paths/analog-output.conf @@ -1,4 +1,22 @@ -# Intended for the 'default' output +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; Intended for the 'default' output +; +; See analog-output.conf.common for an explanation on the directives [General] priority = 100 diff --git a/src/modules/alsa/mixer/paths/analog-output.conf.common b/src/modules/alsa/mixer/paths/analog-output.conf.common index c38eccde..cc1185f4 100644 --- a/src/modules/alsa/mixer/paths/analog-output.conf.common +++ b/src/modules/alsa/mixer/paths/analog-output.conf.common @@ -1,26 +1,97 @@ -# Common part of all paths - -# [General] -# priority = ... -# description = ... -# -# [Option ...:...] -# name = ... -# priority = ... +# This file is part of PulseAudio. # -# [Element ...] -# required = ignore | switch | volume | enumeration | any -# required-absent = ignore | switch | volume +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. # -# switch = ignore | mute | off | on | select -# volume = ignore | merge | off | zero -# enumeration = ignore | select +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. # -# direction = playback | capture -# direction-try-other = no | yes -# -# override-map.1 = ... -# override-map.2 = ... +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; Common part of all paths + +; So here's generally how mixer paths are used by PA: PA goes through +; a mixer path file from top to bottom and checks if a mixer element +; described therein exists. If so it is added to the list of mixer +; elements PA will control, keeping the order it read them in. If a +; mixer element described here has set the required= or +; required-absent= directives a path might not be accepted as valid +; and is ignored in its entirety (see below). However usually if a +; element listed here is missing this one element is ignored but not +; the entire path. +; +; When a device shall be muted/unmuted *all* elements listed in a path +; file with "switch = mute" will be toggled. +; +; When a device shall change its volume, PA will got through the list +; of all elements with "volume = merge" and set the volume on the +; first element. If that element does not support dB volumes, this is +; where the story ends. If it does support dB volumes, PA divides the +; requested volume by the volume that was set on this element, and +; then go on to the next element with "volume = merge" and then set +; that there, and so on. That way the first volume element in the +; path will be the one that does the 'biggest' part of the overall +; volume adjustment, with the remaining elements usually being set to +; some value next to 0dB. This logic makes sure we get the full range +; over all volume sliders and a very high granularity of volumes +; already in hardware. +; +; All switches and enumerations set to "select" are exposed via the +; "port" functionality of sinks/sources. Basically every possible +; switch setting and every possible enumeration setting will be +; combined and made into a "port". So make sure you don't list too +; many switches/enums for exposing, because the number of ports might +; rise exponentially. +; +; Only one path can be selected at a time. All paths that are valid +; for an audio device will be exposed as "port" for the sink/source. + + +; [General] +; priority = ... # Priority for this path +; description = ... +; +; [Option ...:...] # For each option of an enumeration or switch element +; # that shall be exposed as a sink/source port. Needs to +; # be named after the Element, followed by a colon, followed +; # by the option name, resp. on/off if the element is a switch. +; name = ... # Logical name to use in the path identifier +; priority = ... # Priority if this is made into a device port +; +; [Element ...] # For each element that we shall control +; required = ignore | switch | volume | enumeration | any # If set, require this element to be of this kind and available, +; # otherwise don't consider this path valid for the card +; required-absent = ignore | switch | volume # If set, require this element to not be of this kind and not +; # available, otherwise don't consider this path valid for the card +; +; switch = ignore | mute | off | on | select # What to do with this switch: ignore it, make it follow mute status, +; # always set it to off, always to on, or make it selectable as port. +; # If set to 'select' you need to define an Option section for on +; # and off +; volume = ignore | merge | off | zero # What to do with this volume: ignore it, merge it into the device +; # volume slider, always set it to the lowest value possible, or always +; # set it to 0 dB (for whatever that means) +; enumeration = ignore | select # What to do with this enumeration, ignore it or make it selectable +; # via device ports. If set to 'select' you need to define an Option section +; # for each of the items you want to expose +; direction = playback | capture # Is this relevant only for playback or capture? If not set this will implicitly be +; # set the direction of the PCM device is opened as. Generally this doesn't need to be set +; # unless you have a broken driver that has playback controls marked for capture or vice +; # versa +; direction-try-other = no | yes # If the element does not supported what is requested, try the other direction, too? +; +; override-map.1 = ... # Override the channel mask of the mixer control if the control only exposes a single channel +; override-map.2 = ... # Override the channel masks of the mixer control if the control only exposes two channels +; # Override maps should list for each element channel which high-level channels it controls via a +; # channel mask. A channel mask may either be the name of a single channel, or the words "all-left", +; # "all-right", "all-center", "all-front", "all-rear", and "all" to encode a specific subset of +; # channels in a mask [Element PCM] switch = mute diff --git a/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules b/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules new file mode 100644 index 00000000..ea1a2fed --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules @@ -0,0 +1,26 @@ +# do not edit this file, it will be overwritten on update + +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +SUBSYSTEM!="sound", GOTO="pulseaudio_end" +ACTION!="change", GOTO="pulseaudio_end" +KERNEL!="card*", GOTO="pulseaudio_end" + +SUBSYSTEMS=="usb", ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="1978", ENV{PULSE_PROFILE_SET}="native-instruments-audio8dj.conf" +SUBSYSTEMS=="usb", ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="0839", ENV{PULSE_PROFILE_SET}="native-instruments-audio4dj.conf" + +LABEL="pulseaudio_end" diff --git a/src/modules/alsa/mixer/profile-sets/default.conf b/src/modules/alsa/mixer/profile-sets/default.conf index bbe53410..ac41a8d3 100644 --- a/src/modules/alsa/mixer/profile-sets/default.conf +++ b/src/modules/alsa/mixer/profile-sets/default.conf @@ -1,22 +1,60 @@ -# Profile definitions for PulseAudio's ALSA backend +# This file is part of PulseAudio. # -# [Mapping id] -# device-strings = ... -# channel-map = ... -# description = ... -# paths-input = ... -# paths-output = ... -# element-input = ... -# element-output = ... -# priority = ... -# direction = any | input | output +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. # -# [Profile id] -# input-mappings = ... -# output-mappings = ... -# description = ... -# priority = ... -# skip-probe = no | yes +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; Default profile definitions for the ALSA backend of PulseAudio. This +; is used as fallback for all cards that have no special mapping +; assigned. (and should be good enough for the vast majority of +; cards). Use the udev property PULSE_PROFILE_SET to assign a +; different profile set than this one to a device. So what is this +; about? Simply, what we do here is map ALSA devices to how they are +; exposed in PA. We say which ALSA device string to use to open a +; device, which channel mapping to use then, and which mixer path to +; use. This is encoded in a 'mapping'. Multiple of these mappings can +; be bound together in a 'profile' which is then directly exposed in +; the UI as a card profile. Each mapping assigned to a profile will +; result in one sink/source to be created if the profile is selected +; for the card. + +; [General] +; auto-profiles = no | yes # Instead of defining all profiles manually, autogenerate +; # them by combining every input mapping with every output mapping. +; +; [Mapping id] +; device-strings = ... # ALSA device string. %f will be replaced by the card identifier. +; channel-map = ... # Channel mapping to use for this device +; description = ... +; paths-input = ... # A list of mixer paths to use. Every path in this list will be probed. +; # If multiple are found to be working they will be available as device ports +; paths-output = ... +; element-input = ... # Instead of configuring a full mixer path simply configure a single +; # mixer element for volume/mute handling +; element-output = ... +; priority = ... +; direction = any | input | output # Only useful for? +; +; [Profile id] +; input-mappings = ... # Lists mappings for sources on this profile, those mapping must be +; # defined in this file too +; output-mappings = ... # Lists mappings for sinks on this profile, those mappings must be +; # defined in this file too +; description = ... +; priority = ... # Numeric value to deduce priority for this profile +; skip-probe = no | yes # Skip probing for availability? If this is yes then this profile +; # will be assumed as working without probing. Makes initialization +; # a bit faster but only works if the card is really known well. [General] auto-profiles = yes @@ -99,6 +137,7 @@ channel-map = left,right priority = 4 direction = output +; An example for defining multiple-sink profiles #[Profile output:analog-stereo+output:iec958-stereo+input:analog-stereo] #description = Foobar #output-mappings = analog-stereo iec958-stereo diff --git a/src/modules/alsa/mixer/profile-sets/native-instruments-audio4dj.conf b/src/modules/alsa/mixer/profile-sets/native-instruments-audio4dj.conf new file mode 100644 index 00000000..2b835308 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/native-instruments-audio4dj.conf @@ -0,0 +1,91 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; Native Instruments Audio 4 DJ +; +; This card has two stereo pairs of input and two stereo pairs of +; output, named channels A and B. Channel B has an additional +; Headphone connector. +; +; We knowingly only define a subset of the theoretically possible +; mapping combinations as profiles here. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-a] +description = Analog Stereo Channel A +device-strings = hw:%f,0,0 +channel-map = left,right + +[Mapping analog-stereo-b-output] +description = Analog Stereo Channel B (Headphones) +device-strings = hw:%f,0,1 +channel-map = left,right +direction = output + +[Mapping analog-stereo-b-input] +description = Analog Stereo Channel B +device-strings = hw:%f,0,1 +channel-map = left,right +direction = input + +[Profile output:analog-stereo-all+input:analog-stereo-all] +description = Analog Stereo Duplex Channels A, B (Headphones) +output-mappings = analog-stereo-a analog-stereo-b-output +input-mappings = analog-stereo-a analog-stereo-b-input +priority = 100 +skip-probe = yes + +[Profile output:analog-stereo-a+input:analog-stereo-a] +description = Analog Stereo Duplex Channel A +output-mappings = analog-stereo-a +input-mappings = analog-stereo-a +priority = 40 +skip-probe = yes + +[Profile output:analog-stereo-b+input:analog-stereo-b] +description = Analog Stereo Duplex Channel B (Headphones) +output-mappings = analog-stereo-b-output +input-mappings = analog-stereo-b-input +priority = 50 +skip-probe = yes + +[Profile output:analog-stereo-a] +description = Analog Stereo Output Channel A +output-mappings = analog-stereo-a +priority = 5 +skip-probe = yes + +[Profile output:analog-stereo-b] +description = Analog Stereo Output Channel B (Headphones) +output-mappings = analog-stereo-b-output +priority = 6 +skip-probe = yes + +[Profile input:analog-stereo-a] +description = Analog Stereo Input Channel A +input-mappings = analog-stereo-a +priority = 2 +skip-probe = yes + +[Profile input:analog-stereo-b] +description = Analog Stereo Input Channel B +input-mappings = analog-stereo-b-input +priority = 1 +skip-probe = yes diff --git a/src/modules/alsa/mixer/profile-sets/native-instruments-audio8dj.conf b/src/modules/alsa/mixer/profile-sets/native-instruments-audio8dj.conf new file mode 100644 index 00000000..3fe3cc56 --- /dev/null +++ b/src/modules/alsa/mixer/profile-sets/native-instruments-audio8dj.conf @@ -0,0 +1,162 @@ +# This file is part of PulseAudio. +# +# PulseAudio is free software; you can redistribute it and/or modify +# it under the terms of the GNU Lesser General Public License as +# published by the Free Software Foundation; either version 2.1 of the +# License, or (at your option) any later version. +# +# PulseAudio is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU Lesser General Public License +# along with PulseAudio; if not, write to the Free Software Foundation, +# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + +; Native Instruments Audio 8 DJ +; +; This card has four stereo pairs of input and four stereo pairs of +; output, named channels A to D. Channel C has an additional Mic/Line +; connector, channel D an additional Headphone connector. +; +; We knowingly only define a subset of the theoretically possible +; mapping combinations as profiles here. +; +; See default.conf for an explanation on the directives used here. + +[General] +auto-profiles = no + +[Mapping analog-stereo-a] +description = Analog Stereo Channel A +device-strings = hw:%f,0,0 +channel-map = left,right + +[Mapping analog-stereo-b] +description = Analog Stereo Channel B +device-strings = hw:%f,0,1 +channel-map = left,right + +# Since we want to set a different description for channel C's/D's input +# and output we define two seperate mappings for them +[Mapping analog-stereo-c-output] +description = Analog Stereo Channel C +device-strings = hw:%f,0,2 +channel-map = left,right +direction = output + +[Mapping analog-stereo-c-input] +description = Analog Stereo Channel C (Line/Mic) +device-strings = hw:%f,0,2 +channel-map = left,right +direction = input + +[Mapping analog-stereo-d-output] +description = Analog Stereo Channel D (Headphones) +device-strings = hw:%f,0,3 +channel-map = left,right +direction = output + +[Mapping analog-stereo-d-input] +description = Analog Stereo Channel D +device-strings = hw:%f,0,3 +channel-map = left,right +direction = input + +[Profile output:analog-stereo-all+input:analog-stereo-all] +description = Analog Stereo Duplex Channels A, B, C (Line/Mic), D (Headphones) +output-mappings = analog-stereo-a analog-stereo-b analog-stereo-c-output analog-stereo-d-output +input-mappings = analog-stereo-a analog-stereo-b analog-stereo-c-input analog-stereo-d-input +priority = 100 +skip-probe = yes + +[Profile output:analog-stereo-d+input:analog-stereo-c] +description = Analog Stereo Channel D (Headphones) Output, Channel C (Line/Mic) Input +output-mappings = analog-stereo-d-output +input-mappings = analog-stereo-c-input +priority = 90 +skip-probe = yes + +[Profile output:analog-stereo-c-d+input:analog-stereo-c-d] +description = Analog Stereo Duplex Channels C (Line/Mic), D (Line/Mic) +output-mappings = analog-stereo-c-output analog-stereo-d-output +input-mappings = analog-stereo-c-input analog-stereo-d-input +priority = 80 +skip-probe = yes + +[Profile output:analog-stereo-a+input:analog-stereo-a] +description = Analog Stereo Duplex Channel A +output-mappings = analog-stereo-a +input-mappings = analog-stereo-a +priority = 50 +skip-probe = yes + +[Profile output:analog-stereo-b+input:analog-stereo-b] +description = Analog Stereo Duplex Channel B +output-mappings = analog-stereo-b +input-mappings = analog-stereo-b +priority = 40 +skip-probe = yes + +[Profile output:analog-stereo-c+input:analog-stereo-c] +description = Analog Stereo Duplex Channel C (Line/Mic) +output-mappings = analog-stereo-c-output +input-mappings = analog-stereo-c-input +priority = 60 +skip-probe = yes + +[Profile output:analog-stereo-d+input:analog-stereo-d] +description = Analog Stereo Duplex Channel D (Headphones) +output-mappings = analog-stereo-d-output +input-mappings = analog-stereo-d-input +priority = 70 +skip-probe = yes + +[Profile output:analog-stereo-a] +description = Analog Stereo Output Channel A +output-mappings = analog-stereo-a +priority = 6 +skip-probe = yes + +[Profile output:analog-stereo-b] +description = Analog Stereo Output Channel B +output-mappings = analog-stereo-b +priority = 5 +skip-probe = yes + +[Profile output:analog-stereo-c] +description = Analog Stereo Output Channel C +output-mappings = analog-stereo-c-output +priority = 7 +skip-probe = yes + +[Profile output:analog-stereo-d] +description = Analog Stereo Output Channel D (Headphones) +output-mappings = analog-stereo-d-output +priority = 8 +skip-probe = yes + +[Profile input:analog-stereo-a] +description = Analog Stereo Input Channel A +input-mappings = analog-stereo-a +priority = 2 +skip-probe = yes + +[Profile input:analog-stereo-b] +description = Analog Stereo Input Channel B +input-mappings = analog-stereo-b +priority = 1 +skip-probe = yes + +[Profile input:analog-stereo-c] +description = Analog Stereo Input Channel C (Line/Mic) +input-mappings = analog-stereo-c-input +priority = 4 +skip-probe = yes + +[Profile input:analog-stereo-d] +description = Analog Stereo Input Channel D +input-mappings = analog-stereo-d-input +priority = 3 +skip-probe = yes diff --git a/src/modules/alsa/module-alsa-card.c b/src/modules/alsa/module-alsa-card.c index e8a7f206..55f6a6e2 100644 --- a/src/modules/alsa/module-alsa-card.c +++ b/src/modules/alsa/module-alsa-card.c @@ -287,8 +287,7 @@ int pa__init(pa_module *m) { const char *description; char *fn = NULL; - pa_alsa_redirect_errors_inc(); - snd_config_update_free_global(); + pa_alsa_refcnt_inc(); pa_assert(m); @@ -443,6 +442,5 @@ void pa__done(pa_module*m) { pa_xfree(u); finish: - snd_config_update_free_global(); - pa_alsa_redirect_errors_dec(); + pa_alsa_refcnt_dec(); } diff --git a/src/modules/alsa/module-alsa-sink.c b/src/modules/alsa/module-alsa-sink.c index 058ea205..3aa89b2a 100644 --- a/src/modules/alsa/module-alsa-sink.c +++ b/src/modules/alsa/module-alsa-sink.c @@ -82,8 +82,7 @@ int pa__init(pa_module*m) { pa_assert(m); - pa_alsa_redirect_errors_inc(); - snd_config_update_free_global(); + pa_alsa_refcnt_inc(); if (!(ma = pa_modargs_new(m->argument, valid_modargs))) { pa_log("Failed to parse module arguments"); @@ -124,6 +123,5 @@ void pa__done(pa_module*m) { if ((sink = m->userdata)) pa_alsa_sink_free(sink); - snd_config_update_free_global(); - pa_alsa_redirect_errors_dec(); + pa_alsa_refcnt_dec(); } diff --git a/src/modules/alsa/module-alsa-source.c b/src/modules/alsa/module-alsa-source.c index 3bd1b451..23da4185 100644 --- a/src/modules/alsa/module-alsa-source.c +++ b/src/modules/alsa/module-alsa-source.c @@ -37,6 +37,7 @@ #include <pulse/timeval.h> #include <pulsecore/core-error.h> +#include <pulsecore/core-rtclock.h> #include <pulsecore/core.h> #include <pulsecore/module.h> #include <pulsecore/memchunk.h> @@ -51,7 +52,6 @@ #include <pulsecore/thread-mq.h> #include <pulsecore/rtpoll.h> #include <pulsecore/time-smoother.h> -#include <pulsecore/rtclock.h> #include "alsa-util.h" #include "alsa-source.h" @@ -106,8 +106,7 @@ int pa__init(pa_module*m) { pa_assert(m); - pa_alsa_redirect_errors_inc(); - snd_config_update_free_global(); + pa_alsa_refcnt_inc(); if (!(ma = pa_modargs_new(m->argument, valid_modargs))) { pa_log("Failed to parse module arguments"); @@ -148,6 +147,5 @@ void pa__done(pa_module*m) { if ((source = m->userdata)) pa_alsa_source_free(source); - snd_config_update_free_global(); - pa_alsa_redirect_errors_dec(); + pa_alsa_refcnt_dec(); } |