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moc: refactor options, add patches

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Add support for conditional compile options, add pulseaudio and ffmpeg4
support patches.
This commit is contained in:
Eugene 2019-05-03 13:38:25 +03:00
parent 5ad1e9f0bf
commit eaeb587429
3 changed files with 917 additions and 9 deletions
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93
pkgs/applications/audio/moc/default.nix
View File

@ -1,9 +1,35 @@
{ stdenv, fetchurl, ncurses, pkgconfig, alsaLib, flac, libmad, speex, ffmpeg
, libvorbis, libmpc, libsndfile, libjack2, db, libmodplug, timidity, libid3tag
, libtool
{ stdenv, fetchurl, pkgconfig
, ncurses, db , popt, libtool
# Sound sub-systems
, alsaSupport ? true, alsaLib
, pulseSupport ? true, libpulseaudio, autoreconfHook
, jackSupport ? true, libjack2
, ossSupport ? true
# Audio formats
, aacSupport ? true, faad2, libid3tag
, flacSupport ? true, flac
, midiSupport ? true, timidity
, modplugSupport ? true, libmodplug
, mp3Support ? true, libmad
, musepackSupport ? true, libmpc, libmpcdec, taglib
, vorbisSupport ? true, libvorbis
, speexSupport ? true, speex
, ffmpegSupport ? true, ffmpeg
, sndfileSupport ? true, libsndfile
, wavpackSupport ? true, wavpack
# Misc
, withffmpeg4 ? false, ffmpeg_4
, curlSupport ? true, curl
, samplerateSupport ? true, libsamplerate
, withDebug ? false
}:
stdenv.mkDerivation rec {
let
opt = stdenv.lib.optional;
mkFlag = c: f: if c then "--with-${f}" else "--without-${f}";
in stdenv.mkDerivation rec {
name = "moc-${version}";
version = "2.5.2";
@ -12,18 +38,67 @@ stdenv.mkDerivation rec {
sha256 = "026v977kwb0wbmlmf6mnik328plxg8wykfx9ryvqhirac0aq39pk";
};
nativeBuildInputs = [ pkgconfig ];
patches = []
++ opt withffmpeg4 ./moc-ffmpeg4.patch
++ opt pulseSupport ./pulseaudio.patch;
buildInputs = [
ncurses alsaLib flac libmad speex ffmpeg libvorbis libmpc libsndfile libjack2
db libmodplug timidity libid3tag libtool
nativeBuildInputs = [ pkgconfig ]
++ opt pulseSupport autoreconfHook;
buildInputs = [ ncurses db popt libtool ]
# Sound sub-systems
++ opt alsaSupport alsaLib
++ opt pulseSupport libpulseaudio
++ opt jackSupport libjack2
# Audio formats
++ opt (aacSupport || mp3Support) libid3tag
++ opt aacSupport faad2
++ opt flacSupport flac
++ opt midiSupport timidity
++ opt modplugSupport libmodplug
++ opt mp3Support libmad
++ opt musepackSupport [ libmpc libmpcdec taglib ]
++ opt vorbisSupport libvorbis
++ opt speexSupport speex
++ opt (ffmpegSupport && !withffmpeg4) ffmpeg
++ opt (ffmpegSupport && withffmpeg4) ffmpeg_4
++ opt sndfileSupport libsndfile
++ opt wavpackSupport wavpack
# Misc
++ opt curlSupport curl
++ opt samplerateSupport libsamplerate;
configureFlags = [
# Sound sub-systems
(mkFlag alsaSupport "alsa")
(mkFlag pulseSupport "pulse")
(mkFlag jackSupport "jack")
(mkFlag ossSupport "oss")
# Audio formats
(mkFlag aacSupport "aac")
(mkFlag flacSupport "flac")
(mkFlag midiSupport "timidity")
(mkFlag modplugSupport "modplug")
(mkFlag mp3Support "mp3")
(mkFlag musepackSupport "musepack")
(mkFlag vorbisSupport "vorbis")
(mkFlag speexSupport "speex")
(mkFlag ffmpegSupport "ffmpeg")
(mkFlag sndfileSupport "sndfile")
(mkFlag wavpackSupport "wavpack")
# Misc
(mkFlag curlSupport "curl")
(mkFlag samplerateSupport "samplerate")
("--enable-debug=" + (if withDebug then "yes" else "no"))
"--disable-cache"
"--without-rcc"
];
meta = with stdenv.lib; {
description = "An ncurses console audio player designed to be powerful and easy to use";
homepage = http://moc.daper.net/;
license = licenses.gpl2;
maintainers = with maintainers; [ pSub jagajaga ];
maintainers = with maintainers; [ aethelz pSub jagajaga ];
platforms = platforms.linux;
};
}

33
pkgs/applications/audio/moc/moc-ffmpeg4.patch Normal file
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@ -0,0 +1,33 @@
Index: decoder_plugins/ffmpeg/ffmpeg.c
===================================================================
--- /decoder_plugins/ffmpeg/ffmpeg.c (revisión: 2963)
+++ /decoder_plugins/ffmpeg/ffmpeg.c (copia de trabajo)
@@ -697,7 +697,7 @@
* FFmpeg/LibAV in use. For some versions this will be caught in
* *_find_stream_info() above and misreported as an unfound codec
* parameters error. */
- if (data->codec->capabilities & CODEC_CAP_EXPERIMENTAL) {
+ if (data->codec->capabilities & AV_CODEC_CAP_EXPERIMENTAL) {
decoder_error (&data->error, ERROR_FATAL, 0,
"The codec is experimental and may damage MOC: %s",
data->codec->name);
@@ -705,8 +705,8 @@
}
set_downmixing (data);
- if (data->codec->capabilities & CODEC_CAP_TRUNCATED)
- data->enc->flags |= CODEC_FLAG_TRUNCATED;
+ if (data->codec->capabilities & AV_CODEC_CAP_TRUNCATED)
+ data->enc->flags |= AV_CODEC_FLAG_TRUNCATED;
if (avcodec_open2 (data->enc, data->codec, NULL) < 0)
{
@@ -725,7 +725,7 @@
data->sample_width = sfmt_Bps (data->fmt);
- if (data->codec->capabilities & CODEC_CAP_DELAY)
+ if (data->codec->capabilities & AV_CODEC_CAP_DELAY)
data->delay = true;
data->seek_broken = is_seek_broken (data);
data->timing_broken = is_timing_broken (data->ic);

800
pkgs/applications/audio/moc/pulseaudio.patch Normal file
View File

@ -0,0 +1,800 @@
diff --git a/audio.c b/audio.c
--- 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
@@ -893,6 +896,15 @@
}
#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/configure.in b/configure.in
--- a/configure.in
+++ b/configure.in
@@ -162,6 +162,21 @@
AC_MSG_ERROR([BerkeleyDB (libdb) not found.]))
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
--- a/options.c
+++ b/options.c
@@ -572,10 +572,11 @@
#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
--- /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
--- /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