Rework module-combine to work with glitch-free core; add new max_request field to...

[pulseaudio] / src / modules / alsa-util.h

diff --git a/src/modules/alsa-util.h b/src/modules/alsa-util.h
index 675856c6bdaf2a330fb5e9d40401ae80d17ea2bb..4de8bcd24eb867f8a902938c669e39dfb6eca775 100644 (file)
--- a/src/modules/alsa-util.h
+++ b/src/modules/alsa-util.h
@@ -1,11 +1,12 @@
 #ifndef fooalsautilhfoo
 #define fooalsautilhfoo
 
-/* $Id$ */
-
 /***
   This file is part of PulseAudio.
 
+  Copyright 2004-2006 Lennart Poettering
+  Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB
+
   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 of the License,
@@ -26,20 +27,71 @@
 
 #include <pulse/sample.h>
 #include <pulse/mainloop-api.h>
-
 #include <pulse/channelmap.h>
+#include <pulse/proplist.h>
+
+#include <pulsecore/rtpoll.h>
 
-struct pa_alsa_fdlist;
+typedef struct pa_alsa_fdlist pa_alsa_fdlist;
 
 struct pa_alsa_fdlist *pa_alsa_fdlist_new(void);
 void pa_alsa_fdlist_free(struct pa_alsa_fdlist *fdl);
+int pa_alsa_fdlist_set_mixer(struct pa_alsa_fdlist *fdl, snd_mixer_t *mixer_handle, pa_mainloop_api* m);
 
-int pa_alsa_fdlist_init_pcm(struct pa_alsa_fdlist *fdl, snd_pcm_t *pcm_handle, pa_mainloop_api* m, void (*cb)(void *userdata), void *userdata);
-int pa_alsa_fdlist_init_mixer(struct pa_alsa_fdlist *fdl, snd_mixer_t *mixer_handle, pa_mainloop_api* m);
+int pa_alsa_set_hw_params(
+        snd_pcm_t *pcm_handle,
+        pa_sample_spec *ss,
+        uint32_t *periods,
+        snd_pcm_uframes_t *period_size,
+        snd_pcm_uframes_t tsched_size,
+        pa_bool_t *use_mmap,
+        pa_bool_t *use_tsched,
+        pa_bool_t require_exact_channel_number);
 
-int pa_alsa_set_hw_params(snd_pcm_t *pcm_handle, pa_sample_spec *ss, uint32_t *periods, snd_pcm_uframes_t *period_size);
+int pa_alsa_set_sw_params(snd_pcm_t *pcm, snd_pcm_uframes_t avail_min);
 
 int pa_alsa_prepare_mixer(snd_mixer_t *mixer, const char *dev);
 snd_mixer_elem_t *pa_alsa_find_elem(snd_mixer_t *mixer, const char *name, const char *fallback);
 
+snd_pcm_t *pa_alsa_open_by_device_id(
+        const char *dev_id,
+        char **dev,
+        pa_sample_spec *ss,
+        pa_channel_map* map,
+        int mode,
+        uint32_t *nfrags,
+        snd_pcm_uframes_t *period_size,
+        snd_pcm_uframes_t tsched_size,
+        pa_bool_t *use_mmap,
+        pa_bool_t *use_tsched);
+
+snd_pcm_t *pa_alsa_open_by_device_string(
+        const char *device,
+        char **dev,
+        pa_sample_spec *ss,
+        pa_channel_map* map,
+        int mode,
+        uint32_t *nfrags,
+        snd_pcm_uframes_t *period_size,
+        snd_pcm_uframes_t tsched_size,
+        pa_bool_t *use_mmap,
+        pa_bool_t *use_tsched);
+
+int pa_alsa_calc_mixer_map(snd_mixer_elem_t *elem, const pa_channel_map *channel_map, snd_mixer_selem_channel_id_t mixer_map[], pa_bool_t playback);
+
+void pa_alsa_0dB_playback(snd_mixer_elem_t *elem);
+void pa_alsa_0dB_capture(snd_mixer_elem_t *elem);
+
+void pa_alsa_dump(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_init_proplist(pa_proplist *p, snd_pcm_info_t *pcm_info);
+
+int pa_alsa_recover_from_poll(snd_pcm_t *pcm, int revents);
+
+pa_rtpoll_item* pa_alsa_build_pollfd(snd_pcm_t *pcm, pa_rtpoll *rtpoll);
+
 #endif