path: root/0001-Pulseaudio-backend.patch

From 61f9ecdfb20787e916a871c9632f6d375de3242c Mon Sep 17 00:00:00 2001
From: Vladimir Protasov <eoranged@ya.ru>
Date: Wed, 23 Jul 2014 01:47:39 +0400
Subject: [PATCH 1/3] Pulseaudio backend.

http://moc.daper.net/node/831
Thanks for marienz.

Signed-off-by: Hans-Nikolai Viessmann <hv15@hw.ac.uk>
---
 audio.c      |  12 +
 configure.in |  15 ++
 options.c    |   5 +-
 pulse.c      | 705 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 pulse.h      |  14 ++
 5 files changed, 749 insertions(+), 2 deletions(-)
 create mode 100644 pulse.c
 create mode 100644 pulse.h

diff --git a/audio.c b/audio.c
index 57f6a97..839cc8a 100644
--- a/audio.c
+++ b/audio.c
@@ -32,6 +32,9 @@
 #include "log.h"
 #include "lists.h"
 
+#ifdef HAVE_PULSE
+# include "pulse.h"
+#endif
 #ifdef HAVE_OSS
 # include "oss.h"
 #endif
@@ -887,6 +890,15 @@ static void find_working_driver (lists_t_strs *drivers, struct hw_funcs *funcs)
 		}
 #endif
 
+#ifdef HAVE_PULSE
+		if (!strcasecmp(name, "pulseaudio")) {
+			pulse_funcs (funcs);
+			printf ("Trying PulseAudio...\n");
+			if (funcs->init(&hw_caps))
+				return;
+		}
+#endif
+
 #ifdef HAVE_OSS
 		if (!strcasecmp(name, "oss")) {
 			oss_funcs (funcs);
diff --git a/main.c b/main.c
--- a/main.c
+++ b/main.c
@@ -345,6 +345,9 @@
 #ifdef HAVE_SNDIO
 	printf (" SNDIO");
 #endif
+#ifdef HAVE_PULSE
+    printf(" PULSE");
+#endif
 #ifdef HAVE_ALSA
 	printf (" ALSA");
 #endif
diff --git a/configure.in b/configure.in
index 974403e..77432c8 100644
--- a/configure.in
+++ b/configure.in
@@ -157,6 +157,21 @@ then
 	                             [#include <db.h>]])
 fi
 
+AC_ARG_WITH(pulse, AS_HELP_STRING(--without-pulse,
+				  Compile without PulseAudio support.))
+
+if test "x$with_pulse" != "xno"
+then
+	PKG_CHECK_MODULES(PULSE, [libpulse],
+		 [SOUND_DRIVERS="$SOUND_DRIVERS PULSE"
+		  EXTRA_OBJS="$EXTRA_OBJS pulse.o"
+		  AC_DEFINE([HAVE_PULSE], 1, [Define if you have PulseAudio.])
+		  EXTRA_LIBS="$EXTRA_LIBS $PULSE_LIBS"
+		  CFLAGS="$CFLAGS $PULSE_CFLAGS"],
+		  [true])
+fi
+
+
 AC_ARG_WITH(oss, AS_HELP_STRING([--without-oss],
                                 [Compile without OSS support]))
 
diff --git a/options.c b/options.c
index 3ec4a28..496b1f8 100644
--- a/options.c
+++ b/options.c
@@ -567,10 +567,11 @@ void options_init ()
 
 #ifdef OPENBSD
 	add_list ("SoundDriver", "SNDIO:JACK:OSS",
-	          CHECK_DISCRETE(5), "SNDIO", "Jack", "ALSA", "OSS", "null");
+		  CHECK_DISCRETE(5), "SNDIO", "PulseAudio", "Jack", "ALSA", "OSS", "null");
+
 #else
 	add_list ("SoundDriver", "Jack:ALSA:OSS",
-	          CHECK_DISCRETE(5), "SNDIO", "Jack", "ALSA", "OSS", "null");
+	          CHECK_DISCRETE(5), "SNDIO", "PulseAudio", "Jack", "ALSA", "OSS", "null");
 #endif
 
 	add_str  ("JackClientName", "moc", CHECK_NONE);
diff --git a/pulse.c b/pulse.c
new file mode 100644
index 0000000..d5724dd
--- /dev/null
+++ b/pulse.c
@@ -0,0 +1,705 @@
+/*
+ * MOC - music on console
+ * Copyright (C) 2011 Marien Zwart <marienz@marienz.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ */
+
+/* PulseAudio backend.
+ *
+ * FEATURES:
+ *
+ * Does not autostart a PulseAudio server, but uses an already-started
+ * one, which should be better than alsa-through-pulse.
+ *
+ * Supports control of either our stream's or our entire sink's volume
+ * while we are actually playing. Volume control while paused is
+ * intentionally unsupported: the PulseAudio documentation strongly
+ * suggests not passing in an initial volume when creating a stream
+ * (allowing the server to track this instead), and we do not know
+ * which sink to control if we do not have a stream open.
+ *
+ * IMPLEMENTATION:
+ *
+ * Most client-side (resource allocation) errors are fatal. Failure to
+ * create a server context or stream is not fatal (and MOC should cope
+ * with these failures too), but server communication failures later
+ * on are currently not handled (MOC has no great way for us to tell
+ * it we no longer work, and I am not sure if attempting to reconnect
+ * is worth it or even a good idea).
+ *
+ * The pulse "simple" API is too simple: it combines connecting to the
+ * server and opening a stream into one operation, while I want to
+ * connect to the server when MOC starts (and fall back to a different
+ * backend if there is no server), and I cannot open a stream at that
+ * time since I do not know the audio format yet.
+ *
+ * PulseAudio strongly recommends we use a high-latency connection,
+ * which the MOC frontend code might not expect from its audio
+ * backend. We'll see.
+ *
+ * We map MOC's percentage volumes linearly to pulse's PA_VOLUME_MUTED
+ * (0) .. PA_VOLUME_NORM range. This is what the PulseAudio docs recommend
+ * ( http://pulseaudio.org/wiki/WritingVolumeControlUIs ). It does mean
+ * PulseAudio volumes above PA_VOLUME_NORM do not work well with MOC.
+ *
+ * Comments in audio.h claim "All functions are executed only by one
+ * thread" (referring to the function in the hw_funcs struct). This is
+ * a blatant lie. Most of them are invoked off the "output buffer"
+ * thread (out_buf.c) but at least the "playing" thread (audio.c)
+ * calls audio_close which calls our close function. We can mostly
+ * ignore this problem because we serialize on the pulseaudio threaded
+ * mainloop lock. But it does mean that functions that are normally
+ * only called between open and close (like reset) are sometimes
+ * called without us having a stream. Bulletproof, therefore:
+ * serialize setting/unsetting our global stream using the threaded
+ * mainloop lock, and check for that stream being non-null before
+ * using it.
+ *
+ * I am not convinced there are no further dragons lurking here: can
+ * the "playing" thread(s) close and reopen our output stream while
+ * the "output buffer" thread is sending output there? We can bail if
+ * our stream is simply closed, but we do not currently detect it
+ * being reopened and no longer using the same sample format, which
+ * might have interesting results...
+ *
+ * Also, read_mixer is called from the main server thread (handling
+ * commands). This crashed me once when it got at a stream that was in
+ * the "creating" state and therefore did not have a valid stream
+ * index yet. Fixed by only assigning to the stream global when the
+ * stream is valid.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#define DEBUG
+
+#include <pulse/pulseaudio.h>
+#include "common.h"
+#include "log.h"
+#include "audio.h"
+
+
+/* The pulse mainloop and context are initialized in pulse_init and
+ * destroyed in pulse_shutdown.
+ */
+static pa_threaded_mainloop *mainloop = NULL;
+static pa_context *context = NULL;
+
+/* The stream is initialized in pulse_open and destroyed in pulse_close. */
+static pa_stream *stream = NULL;
+
+static int showing_sink_volume = 0;
+
+/* Callbacks that do nothing but wake up the mainloop. */
+
+static void context_state_callback (pa_context *context ATTR_UNUSED,
+				    void *userdata)
+{
+	pa_threaded_mainloop *m = userdata;
+
+	pa_threaded_mainloop_signal (m, 0);
+}
+
+static void stream_state_callback (pa_stream *stream ATTR_UNUSED,
+				   void *userdata)
+{
+	pa_threaded_mainloop *m = userdata;
+
+	pa_threaded_mainloop_signal (m, 0);
+}
+
+static void stream_write_callback (pa_stream *stream ATTR_UNUSED,
+				   size_t nbytes ATTR_UNUSED, void *userdata)
+{
+	pa_threaded_mainloop *m = userdata;
+
+	pa_threaded_mainloop_signal (m, 0);
+}
+
+/* Initialize pulse mainloop and context. Failure to connect to the
+ * pulse daemon is nonfatal, everything else is fatal (as it
+ * presumably means we ran out of resources).
+ */
+static int pulse_init (struct output_driver_caps *caps)
+{
+	pa_context *c;
+	pa_proplist *proplist;
+
+	assert (!mainloop);
+	assert (!context);
+
+	mainloop = pa_threaded_mainloop_new ();
+	if (!mainloop)
+		fatal ("Cannot create PulseAudio mainloop");
+
+	if (pa_threaded_mainloop_start (mainloop) < 0)
+		fatal ("Cannot start PulseAudio mainloop");
+
+	/* TODO: possibly add more props.
+	 *
+	 * There are a few we could set in proplist.h but nothing I
+	 * expect to be very useful.
+	 *
+	 * http://pulseaudio.org/wiki/ApplicationProperties recommends
+	 * setting at least application.name, icon.name and media.role.
+	 *
+	 * No need to set application.name here, the name passed to
+	 * pa_context_new_with_proplist overrides it.
+	 */
+	proplist = pa_proplist_new ();
+	if (!proplist)
+		fatal ("Cannot allocate PulseAudio proplist");
+
+	pa_proplist_sets (proplist,
+			  PA_PROP_APPLICATION_VERSION, PACKAGE_VERSION);
+	pa_proplist_sets (proplist, PA_PROP_MEDIA_ROLE, "music");
+	pa_proplist_sets (proplist, PA_PROP_APPLICATION_ID, "net.daper.moc");
+
+	pa_threaded_mainloop_lock (mainloop);
+
+	c = pa_context_new_with_proplist (
+		pa_threaded_mainloop_get_api (mainloop),
+		PACKAGE_NAME, proplist);
+	pa_proplist_free (proplist);
+
+	if (!c)
+		fatal ("Cannot allocate PulseAudio context");
+
+	pa_context_set_state_callback (c, context_state_callback, mainloop);
+
+	/* Ignore return value, rely on state being set properly */
+	pa_context_connect (c, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL);
+
+	while (1) {
+		pa_context_state_t state = pa_context_get_state (c);
+
+		if (state == PA_CONTEXT_READY)
+			break;
+
+		if (!PA_CONTEXT_IS_GOOD (state)) {
+			error ("PulseAudio connection failed: %s",
+			       pa_strerror (pa_context_errno (c)));
+
+			goto unlock_and_fail;
+		}
+
+		debug ("waiting for context to become ready...");
+		pa_threaded_mainloop_wait (mainloop);
+	}
+
+	/* Only set the global now that the context is actually ready */
+	context = c;
+
+	pa_threaded_mainloop_unlock (mainloop);
+
+	/* We just make up the hardware capabilities, since pulse is
+	 * supposed to be abstracting these out. Assume pulse will
+	 * deal with anything we want to throw at it, and that we will
+	 * only want mono or stereo audio.
+	 */
+	caps->min_channels = 1;
+	caps->max_channels = 2;
+	caps->formats = (SFMT_S8 | SFMT_S16 | SFMT_S32 |
+			 SFMT_FLOAT | SFMT_BE | SFMT_LE);
+
+	return 1;
+
+unlock_and_fail:
+
+	pa_context_unref (c);
+
+	pa_threaded_mainloop_unlock (mainloop);
+
+	pa_threaded_mainloop_stop (mainloop);
+	pa_threaded_mainloop_free (mainloop);
+	mainloop = NULL;
+
+	return 0;
+}
+
+static void pulse_shutdown (void)
+{
+	pa_threaded_mainloop_lock (mainloop);
+
+	pa_context_disconnect (context);
+	pa_context_unref (context);
+	context = NULL;
+
+	pa_threaded_mainloop_unlock (mainloop);
+
+	pa_threaded_mainloop_stop (mainloop);
+	pa_threaded_mainloop_free (mainloop);
+	mainloop = NULL;
+}
+
+static int pulse_open (struct sound_params *sound_params)
+{
+	pa_sample_spec ss;
+	pa_buffer_attr ba;
+	pa_stream *s;
+
+	assert (!stream);
+	/* Initialize everything to -1, which in practice gets us
+	 * about 2 seconds of latency (which is fine). This is not the
+	 * same as passing NULL for this struct, which gets us an
+	 * unnecessarily short alsa-like latency.
+	 */
+	ba.fragsize = (uint32_t) -1;
+	ba.tlength = (uint32_t) -1;
+	ba.prebuf = (uint32_t) -1;
+	ba.minreq = (uint32_t) -1;
+	ba.maxlength = (uint32_t) -1;
+
+	ss.channels = sound_params->channels;
+	ss.rate = sound_params->rate;
+	switch (sound_params->fmt) {
+	case SFMT_U8:
+		ss.format = PA_SAMPLE_U8;
+		break;
+	case SFMT_S16 | SFMT_LE:
+		ss.format = PA_SAMPLE_S16LE;
+		break;
+	case SFMT_S16 | SFMT_BE:
+		ss.format = PA_SAMPLE_S16BE;
+		break;
+	case SFMT_FLOAT | SFMT_LE:
+		ss.format = PA_SAMPLE_FLOAT32LE;
+		break;
+	case SFMT_FLOAT | SFMT_BE:
+		ss.format = PA_SAMPLE_FLOAT32BE;
+		break;
+	case SFMT_S32 | SFMT_LE:
+		ss.format = PA_SAMPLE_S32LE;
+		break;
+	case SFMT_S32 | SFMT_BE:
+		ss.format = PA_SAMPLE_S32BE;
+		break;
+
+	default:
+		fatal ("pulse: got unrequested format");
+	}
+
+	debug ("opening stream");
+
+	pa_threaded_mainloop_lock (mainloop);
+
+	/* TODO: figure out if there are useful stream properties to set.
+	 *
+	 * I do not really see any in proplist.h that we can set from
+	 * here (there are media title/artist/etc props but we do not
+	 * have that data available here).
+	 */
+	s = pa_stream_new (context, "music", &ss, NULL);
+	if (!s)
+		fatal ("pulse: stream allocation failed");
+
+	pa_stream_set_state_callback (s, stream_state_callback, mainloop);
+	pa_stream_set_write_callback (s, stream_write_callback, mainloop);
+
+	/* Ignore return value, rely on failed stream state instead. */
+	pa_stream_connect_playback (
+		s, NULL, &ba,
+		PA_STREAM_INTERPOLATE_TIMING |
+		PA_STREAM_AUTO_TIMING_UPDATE |
+		PA_STREAM_ADJUST_LATENCY,
+		NULL, NULL);
+
+	while (1) {
+		pa_stream_state_t state = pa_stream_get_state (s);
+
+		if (state == PA_STREAM_READY)
+			break;
+
+		if (!PA_STREAM_IS_GOOD (state)) {
+			error ("PulseAudio stream connection failed");
+
+			goto fail;
+		}
+
+		debug ("waiting for stream to become ready...");
+		pa_threaded_mainloop_wait (mainloop);
+	}
+
+	/* Only set the global stream now that it is actually ready */
+	stream = s;
+
+	pa_threaded_mainloop_unlock (mainloop);
+
+	return 1;
+
+fail:
+	pa_stream_unref (s);
+
+	pa_threaded_mainloop_unlock (mainloop);
+	return 0;
+}
+
+static void pulse_close (void)
+{
+	debug ("closing stream");
+
+	pa_threaded_mainloop_lock (mainloop);
+
+	pa_stream_disconnect (stream);
+	pa_stream_unref (stream);
+	stream = NULL;
+
+	pa_threaded_mainloop_unlock (mainloop);
+}
+
+static int pulse_play (const char *buff, const size_t size)
+{
+	size_t offset = 0;
+
+	debug ("Got %d bytes to play", (int)size);
+
+	pa_threaded_mainloop_lock (mainloop);
+
+	/* The buffer is usually writable when we get here, and there
+	 * are usually few (if any) writes after the first one. So
+	 * there is no point in doing further writes directly from the
+	 * callback: we can just do all writes from this thread.
+	 */
+
+	/* Break out of the loop if some other thread manages to close
+	 * our stream underneath us.
+	 */
+	while (stream) {
+		size_t towrite = MIN(pa_stream_writable_size (stream),
+				     size - offset);
+		debug ("writing %d bytes", (int)towrite);
+
+		/* We have no working way of dealing with errors
+		 * (see below). */
+		if (pa_stream_write(stream, buff + offset, towrite,
+				    NULL, 0, PA_SEEK_RELATIVE))
+			error ("pa_stream_write failed");
+
+		offset += towrite;
+
+		if (offset >= size)
+			break;
+
+		pa_threaded_mainloop_wait (mainloop);
+	}
+
+	pa_threaded_mainloop_unlock (mainloop);
+
+	debug ("Done playing!");
+
+	/* We should always return size, calling code does not deal
+	 * well with anything else. Only read the rest if you want to
+	 * know why.
+	 *
+	 * The output buffer reader thread (out_buf.c:read_thread)
+	 * repeatedly loads some 64k/0.1s of audio into a buffer on
+	 * the stack, then calls audio_send_pcm repeatedly until this
+	 * entire buffer has been processed (similar to the loop in
+	 * this function). audio_send_pcm applies the softmixer and
+	 * equalizer, then feeds the result to this function, passing
+	 * through our return value.
+	 *
+	 * So if we return less than size the equalizer/softmixer is
+	 * re-applied to the remaining data, which is silly. Also,
+	 * audio_send_pcm checks for our return value being zero and
+	 * calls fatal() if it is, so try to always process *some*
+	 * data. Also, out_buf.c uses the return value of this
+	 * function from the last run through its inner loop to update
+	 * its time attribute, which means it will be interestingly
+	 * off if that loop ran more than once.
+	 *
+	 * Oh, and alsa.c seems to think it can return -1 to indicate
+	 * failure, which will cause out_buf.c to rewind its buffer
+	 * (to before its start, usually).
+	 */
+	return size;
+}
+
+static void volume_cb (const pa_cvolume *v, void *userdata)
+{
+	int *result = userdata;
+
+	if (v)
+		*result = 100 * pa_cvolume_avg (v) / PA_VOLUME_NORM;
+
+	pa_threaded_mainloop_signal (mainloop, 0);
+}
+
+static void sink_volume_cb (pa_context *c ATTR_UNUSED,
+			    const pa_sink_info *i, int eol ATTR_UNUSED,
+			    void *userdata)
+{
+	volume_cb (i ? &i->volume : NULL, userdata);
+}
+
+static void sink_input_volume_cb (pa_context *c ATTR_UNUSED,
+				  const pa_sink_input_info *i,
+				  int eol ATTR_UNUSED,
+				  void *userdata ATTR_UNUSED)
+{
+	volume_cb (i ? &i->volume : NULL, userdata);
+}
+
+static int pulse_read_mixer (void)
+{
+	pa_operation *op;
+	int result = 0;
+
+	debug ("read mixer");
+
+	pa_threaded_mainloop_lock (mainloop);
+
+	if (stream) {
+		if (showing_sink_volume)
+			op = pa_context_get_sink_info_by_index (
+				context, pa_stream_get_device_index (stream),
+				sink_volume_cb, &result);
+		else
+			op = pa_context_get_sink_input_info (
+				context, pa_stream_get_index (stream),
+				sink_input_volume_cb, &result);
+
+		while (pa_operation_get_state (op) == PA_OPERATION_RUNNING)
+			pa_threaded_mainloop_wait (mainloop);
+
+		pa_operation_unref (op);
+	}
+
+	pa_threaded_mainloop_unlock (mainloop);
+
+	return result;
+}
+
+static void pulse_set_mixer (int vol)
+{
+	pa_cvolume v;
+	pa_operation *op;
+
+	/* Setting volume for one channel does the right thing. */
+	pa_cvolume_set(&v, 1, vol * PA_VOLUME_NORM / 100);
+
+	pa_threaded_mainloop_lock (mainloop);
+
+	if (stream) {
+		if (showing_sink_volume)
+			op = pa_context_set_sink_volume_by_index (
+				context, pa_stream_get_device_index (stream),
+				&v, NULL, NULL);
+		else
+			op = pa_context_set_sink_input_volume (
+				context, pa_stream_get_index (stream),
+				&v, NULL, NULL);
+
+		pa_operation_unref (op);
+	}
+
+	pa_threaded_mainloop_unlock (mainloop);
+}
+
+static int pulse_get_buff_fill (void)
+{
+	/* This function is problematic. MOC uses it to for the "time
+	 * remaining" in the UI, but calls it more than once per
+	 * second (after each chunk of audio played, not for each
+	 * playback time update). We have to be fairly accurate here
+	 * for that time remaining to not jump weirdly. But PulseAudio
+	 * cannot give us a 100% accurate value here, as it involves a
+	 * server roundtrip. And if we call this a lot it suggests
+	 * switching to a mode where the value is interpolated, making
+	 * it presumably more inaccurate (see the flags we pass to
+	 * pa_stream_connect_playback).
+	 *
+	 * MOC also contains what I believe to be a race: it calls
+	 * audio_get_buff_fill "soon" (after playing the first chunk)
+	 * after starting playback of the next song, at which point we
+	 * still have part of the previous song buffered. This means
+	 * our position into the new song is negative, which fails an
+	 * assert (in out_buf.c:out_buf_time_get). There is no sane
+	 * way for us to detect this condition. I believe no other
+	 * backend triggers this because the assert sits after an
+	 * implicit float -> int seconds conversion, which means we
+	 * have to be off by at least an entire second to get a
+	 * negative value, and none of the other backends have buffers
+	 * that large (alsa buffers are supposedly a few 100 ms).
+	 */
+	pa_usec_t buffered_usecs = 0;
+	int buffered_bytes = 0;
+
+	pa_threaded_mainloop_lock (mainloop);
+
+	/* Using pa_stream_get_timing_info and returning the distance
+	 * between write_index and read_index would be more obvious,
+	 * but because of how the result is actually used I believe
+	 * using the latency value is slightly more correct, and it
+	 * makes the following crash-avoidance hack more obvious.
+	 */
+
+	/* This function will frequently fail the first time we call
+	 * it (pulse does not have the requested data yet). We ignore
+	 * that and just return 0.
+	 *
+	 * Deal with stream being NULL too, just in case this is
+	 * called in a racy fashion similar to how reset() is.
+	 */
+	if (stream &&
+	    pa_stream_get_latency (stream, &buffered_usecs, NULL) >= 0) {
+		/* Crash-avoidance HACK: floor our latency to at most
+		 * 1 second. It is usually more, but reporting that at
+		 * the start of playback crashes MOC, and we cannot
+		 * sanely detect when reporting it is safe.
+		 */
+		if (buffered_usecs > 1000000)
+			buffered_usecs = 1000000;
+
+		buffered_bytes = pa_usec_to_bytes (
+			buffered_usecs,
+			pa_stream_get_sample_spec (stream));
+	}
+
+	pa_threaded_mainloop_unlock (mainloop);
+
+	debug ("buffer fill: %d usec / %d bytes",
+	       (int) buffered_usecs, (int) buffered_bytes);
+
+	return buffered_bytes;
+}
+
+static void flush_callback (pa_stream *s ATTR_UNUSED, int success,
+			    void *userdata)
+{
+	int *result = userdata;
+
+	*result = success;
+
+	pa_threaded_mainloop_signal (mainloop, 0);
+}
+
+static int pulse_reset (void)
+{
+	pa_operation *op;
+	int result = 0;
+
+	debug ("reset requested");
+
+	pa_threaded_mainloop_lock (mainloop);
+
+	/* We *should* have a stream here, but MOC is racy, so bulletproof */
+	if (stream) {
+		op = pa_stream_flush (stream, flush_callback, &result);
+
+		while (pa_operation_get_state (op) == PA_OPERATION_RUNNING)
+			pa_threaded_mainloop_wait (mainloop);
+
+		pa_operation_unref (op);
+	} else
+		logit ("pulse_reset() called without a stream");
+
+	pa_threaded_mainloop_unlock (mainloop);
+
+	return result;
+}
+
+static int pulse_get_rate (void)
+{
+	/* This is called once right after open. Do not bother making
+	 * this fast. */
+
+	int result;
+
+	pa_threaded_mainloop_lock (mainloop);
+
+	if (stream)
+		result = pa_stream_get_sample_spec (stream)->rate;
+	else {
+		error ("get_rate called without a stream");
+		result = 0;
+	}
+
+	pa_threaded_mainloop_unlock (mainloop);
+
+	return result;
+}
+
+static void pulse_toggle_mixer_channel (void)
+{
+	showing_sink_volume = !showing_sink_volume;
+}
+
+static void sink_name_cb (pa_context *c ATTR_UNUSED,
+			  const pa_sink_info *i, int eol ATTR_UNUSED,
+			  void *userdata)
+{
+	char **result = userdata;
+
+	if (i && !*result)
+		*result = xstrdup (i->name);
+
+	pa_threaded_mainloop_signal (mainloop, 0);
+}
+
+static void sink_input_name_cb (pa_context *c ATTR_UNUSED,
+				const pa_sink_input_info *i,
+				int eol ATTR_UNUSED,
+				void *userdata)
+{
+	char **result = userdata;
+
+	if (i && !*result)
+		*result = xstrdup (i->name);
+
+	pa_threaded_mainloop_signal (mainloop, 0);
+}
+
+static char *pulse_get_mixer_channel_name (void)
+{
+	char *result = NULL;
+	pa_operation *op;
+
+	pa_threaded_mainloop_lock (mainloop);
+
+	if (stream) {
+		if (showing_sink_volume)
+			op = pa_context_get_sink_info_by_index (
+				context, pa_stream_get_device_index (stream),
+				sink_name_cb, &result);
+		else
+			op = pa_context_get_sink_input_info (
+				context, pa_stream_get_index (stream),
+				sink_input_name_cb, &result);
+
+		while (pa_operation_get_state (op) == PA_OPERATION_RUNNING)
+			pa_threaded_mainloop_wait (mainloop);
+
+		pa_operation_unref (op);
+	}
+
+	pa_threaded_mainloop_unlock (mainloop);
+
+	if (!result)
+		result = xstrdup ("disconnected");
+
+	return result;
+}
+
+void pulse_funcs (struct hw_funcs *funcs)
+{
+	funcs->init = pulse_init;
+	funcs->shutdown = pulse_shutdown;
+	funcs->open = pulse_open;
+	funcs->close = pulse_close;
+	funcs->play = pulse_play;
+	funcs->read_mixer = pulse_read_mixer;
+	funcs->set_mixer = pulse_set_mixer;
+	funcs->get_buff_fill = pulse_get_buff_fill;
+	funcs->reset = pulse_reset;
+	funcs->get_rate = pulse_get_rate;
+	funcs->toggle_mixer_channel = pulse_toggle_mixer_channel;
+	funcs->get_mixer_channel_name = pulse_get_mixer_channel_name;
+}
diff --git a/pulse.h b/pulse.h
new file mode 100644
index 0000000..7f99067
--- /dev/null
+++ b/pulse.h
@@ -0,0 +1,14 @@
+#ifndef PULSE_H
+#define PULSE_H
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+void pulse_funcs (struct hw_funcs *funcs);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
-- 
2.12.1