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eaeb587429
Add support for conditional compile options, add pulseaudio and ffmpeg4 support patches.
801 lines
22 KiB
Diff
801 lines
22 KiB
Diff
diff --git a/audio.c b/audio.c
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--- a/audio.c
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+++ b/audio.c
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@@ -32,6 +32,9 @@
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#include "log.h"
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#include "lists.h"
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+#ifdef HAVE_PULSE
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+# include "pulse.h"
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+#endif
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#ifdef HAVE_OSS
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# include "oss.h"
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#endif
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@@ -893,6 +896,15 @@
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}
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#endif
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+#ifdef HAVE_PULSE
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+ if (!strcasecmp(name, "pulseaudio")) {
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+ pulse_funcs (funcs);
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+ printf ("Trying PulseAudio...\n");
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+ if (funcs->init(&hw_caps))
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+ return;
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+ }
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+#endif
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+
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#ifdef HAVE_OSS
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if (!strcasecmp(name, "oss")) {
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oss_funcs (funcs);
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diff --git a/configure.in b/configure.in
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--- a/configure.in
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+++ b/configure.in
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@@ -162,6 +162,21 @@
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AC_MSG_ERROR([BerkeleyDB (libdb) not found.]))
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fi
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+AC_ARG_WITH(pulse, AS_HELP_STRING(--without-pulse,
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+ Compile without PulseAudio support.))
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+
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+if test "x$with_pulse" != "xno"
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+then
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+ PKG_CHECK_MODULES(PULSE, [libpulse],
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+ [SOUND_DRIVERS="$SOUND_DRIVERS PULSE"
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+ EXTRA_OBJS="$EXTRA_OBJS pulse.o"
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+ AC_DEFINE([HAVE_PULSE], 1, [Define if you have PulseAudio.])
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+ EXTRA_LIBS="$EXTRA_LIBS $PULSE_LIBS"
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+ CFLAGS="$CFLAGS $PULSE_CFLAGS"],
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+ [true])
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+fi
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+
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+
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AC_ARG_WITH(oss, AS_HELP_STRING([--without-oss],
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[Compile without OSS support]))
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diff --git a/options.c b/options.c
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--- a/options.c
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+++ b/options.c
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@@ -572,10 +572,11 @@
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#ifdef OPENBSD
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add_list ("SoundDriver", "SNDIO:JACK:OSS",
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- CHECK_DISCRETE(5), "SNDIO", "Jack", "ALSA", "OSS", "null");
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+ CHECK_DISCRETE(5), "SNDIO", "PulseAudio", "Jack", "ALSA", "OSS", "null");
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+
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#else
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add_list ("SoundDriver", "Jack:ALSA:OSS",
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- CHECK_DISCRETE(5), "SNDIO", "Jack", "ALSA", "OSS", "null");
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+ CHECK_DISCRETE(5), "SNDIO", "PulseAudio", "Jack", "ALSA", "OSS", "null");
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#endif
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add_str ("JackClientName", "moc", CHECK_NONE);
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diff --git a/pulse.c b/pulse.c
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new file mode 100644
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--- /dev/null
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+++ b/pulse.c
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@@ -0,0 +1,705 @@
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+/*
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+ * MOC - music on console
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+ * Copyright (C) 2011 Marien Zwart <marienz@marienz.net>
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+ *
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+ * This program is free software; you can redistribute it and/or modify
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+ * it under the terms of the GNU General Public License as published by
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+ * the Free Software Foundation; either version 2 of the License, or
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+ * (at your option) any later version.
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+ *
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+ */
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+
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+/* PulseAudio backend.
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+ *
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+ * FEATURES:
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+ *
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+ * Does not autostart a PulseAudio server, but uses an already-started
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+ * one, which should be better than alsa-through-pulse.
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+ *
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+ * Supports control of either our stream's or our entire sink's volume
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+ * while we are actually playing. Volume control while paused is
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+ * intentionally unsupported: the PulseAudio documentation strongly
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+ * suggests not passing in an initial volume when creating a stream
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+ * (allowing the server to track this instead), and we do not know
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+ * which sink to control if we do not have a stream open.
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+ *
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+ * IMPLEMENTATION:
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+ *
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+ * Most client-side (resource allocation) errors are fatal. Failure to
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+ * create a server context or stream is not fatal (and MOC should cope
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+ * with these failures too), but server communication failures later
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+ * on are currently not handled (MOC has no great way for us to tell
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+ * it we no longer work, and I am not sure if attempting to reconnect
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+ * is worth it or even a good idea).
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+ *
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+ * The pulse "simple" API is too simple: it combines connecting to the
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+ * server and opening a stream into one operation, while I want to
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+ * connect to the server when MOC starts (and fall back to a different
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+ * backend if there is no server), and I cannot open a stream at that
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+ * time since I do not know the audio format yet.
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+ *
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+ * PulseAudio strongly recommends we use a high-latency connection,
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+ * which the MOC frontend code might not expect from its audio
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+ * backend. We'll see.
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+ *
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+ * We map MOC's percentage volumes linearly to pulse's PA_VOLUME_MUTED
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+ * (0) .. PA_VOLUME_NORM range. This is what the PulseAudio docs recommend
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+ * ( http://pulseaudio.org/wiki/WritingVolumeControlUIs ). It does mean
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+ * PulseAudio volumes above PA_VOLUME_NORM do not work well with MOC.
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+ *
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+ * Comments in audio.h claim "All functions are executed only by one
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+ * thread" (referring to the function in the hw_funcs struct). This is
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+ * a blatant lie. Most of them are invoked off the "output buffer"
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+ * thread (out_buf.c) but at least the "playing" thread (audio.c)
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+ * calls audio_close which calls our close function. We can mostly
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+ * ignore this problem because we serialize on the pulseaudio threaded
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+ * mainloop lock. But it does mean that functions that are normally
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+ * only called between open and close (like reset) are sometimes
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+ * called without us having a stream. Bulletproof, therefore:
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+ * serialize setting/unsetting our global stream using the threaded
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+ * mainloop lock, and check for that stream being non-null before
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+ * using it.
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+ *
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+ * I am not convinced there are no further dragons lurking here: can
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+ * the "playing" thread(s) close and reopen our output stream while
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+ * the "output buffer" thread is sending output there? We can bail if
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+ * our stream is simply closed, but we do not currently detect it
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+ * being reopened and no longer using the same sample format, which
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+ * might have interesting results...
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+ *
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+ * Also, read_mixer is called from the main server thread (handling
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+ * commands). This crashed me once when it got at a stream that was in
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+ * the "creating" state and therefore did not have a valid stream
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+ * index yet. Fixed by only assigning to the stream global when the
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+ * stream is valid.
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+ */
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+
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+#ifdef HAVE_CONFIG_H
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+# include "config.h"
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+#endif
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+
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+#define DEBUG
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+
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+#include <pulse/pulseaudio.h>
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+#include "common.h"
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+#include "log.h"
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+#include "audio.h"
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+
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+
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+/* The pulse mainloop and context are initialized in pulse_init and
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+ * destroyed in pulse_shutdown.
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+ */
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+static pa_threaded_mainloop *mainloop = NULL;
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+static pa_context *context = NULL;
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+
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+/* The stream is initialized in pulse_open and destroyed in pulse_close. */
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+static pa_stream *stream = NULL;
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+
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+static int showing_sink_volume = 0;
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+
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+/* Callbacks that do nothing but wake up the mainloop. */
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+
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+static void context_state_callback (pa_context *context ATTR_UNUSED,
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+ void *userdata)
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+{
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+ pa_threaded_mainloop *m = userdata;
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+
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+ pa_threaded_mainloop_signal (m, 0);
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+}
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+
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+static void stream_state_callback (pa_stream *stream ATTR_UNUSED,
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+ void *userdata)
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+{
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+ pa_threaded_mainloop *m = userdata;
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+
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+ pa_threaded_mainloop_signal (m, 0);
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+}
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+
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+static void stream_write_callback (pa_stream *stream ATTR_UNUSED,
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+ size_t nbytes ATTR_UNUSED, void *userdata)
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+{
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+ pa_threaded_mainloop *m = userdata;
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+
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+ pa_threaded_mainloop_signal (m, 0);
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+}
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+
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+/* Initialize pulse mainloop and context. Failure to connect to the
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+ * pulse daemon is nonfatal, everything else is fatal (as it
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+ * presumably means we ran out of resources).
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+ */
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+static int pulse_init (struct output_driver_caps *caps)
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+{
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+ pa_context *c;
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+ pa_proplist *proplist;
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+
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+ assert (!mainloop);
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+ assert (!context);
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+
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+ mainloop = pa_threaded_mainloop_new ();
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+ if (!mainloop)
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+ fatal ("Cannot create PulseAudio mainloop");
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+
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+ if (pa_threaded_mainloop_start (mainloop) < 0)
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+ fatal ("Cannot start PulseAudio mainloop");
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+
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+ /* TODO: possibly add more props.
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+ *
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+ * There are a few we could set in proplist.h but nothing I
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+ * expect to be very useful.
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+ *
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+ * http://pulseaudio.org/wiki/ApplicationProperties recommends
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+ * setting at least application.name, icon.name and media.role.
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+ *
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+ * No need to set application.name here, the name passed to
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+ * pa_context_new_with_proplist overrides it.
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+ */
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+ proplist = pa_proplist_new ();
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+ if (!proplist)
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+ fatal ("Cannot allocate PulseAudio proplist");
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+
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+ pa_proplist_sets (proplist,
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+ PA_PROP_APPLICATION_VERSION, PACKAGE_VERSION);
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+ pa_proplist_sets (proplist, PA_PROP_MEDIA_ROLE, "music");
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+ pa_proplist_sets (proplist, PA_PROP_APPLICATION_ID, "net.daper.moc");
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+
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+ pa_threaded_mainloop_lock (mainloop);
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+
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+ c = pa_context_new_with_proplist (
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+ pa_threaded_mainloop_get_api (mainloop),
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+ PACKAGE_NAME, proplist);
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+ pa_proplist_free (proplist);
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+
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+ if (!c)
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+ fatal ("Cannot allocate PulseAudio context");
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+
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+ pa_context_set_state_callback (c, context_state_callback, mainloop);
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+
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+ /* Ignore return value, rely on state being set properly */
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+ pa_context_connect (c, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL);
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+
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+ while (1) {
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+ pa_context_state_t state = pa_context_get_state (c);
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+
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+ if (state == PA_CONTEXT_READY)
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+ break;
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+
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+ if (!PA_CONTEXT_IS_GOOD (state)) {
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+ error ("PulseAudio connection failed: %s",
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+ pa_strerror (pa_context_errno (c)));
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+
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+ goto unlock_and_fail;
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+ }
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+
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+ debug ("waiting for context to become ready...");
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+ pa_threaded_mainloop_wait (mainloop);
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+ }
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+
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+ /* Only set the global now that the context is actually ready */
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+ context = c;
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+
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+ pa_threaded_mainloop_unlock (mainloop);
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+
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+ /* We just make up the hardware capabilities, since pulse is
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+ * supposed to be abstracting these out. Assume pulse will
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+ * deal with anything we want to throw at it, and that we will
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+ * only want mono or stereo audio.
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+ */
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+ caps->min_channels = 1;
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+ caps->max_channels = 2;
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+ caps->formats = (SFMT_S8 | SFMT_S16 | SFMT_S32 |
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+ SFMT_FLOAT | SFMT_BE | SFMT_LE);
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+
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+ return 1;
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+
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+unlock_and_fail:
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+
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+ pa_context_unref (c);
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+
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+ pa_threaded_mainloop_unlock (mainloop);
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+
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+ pa_threaded_mainloop_stop (mainloop);
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+ pa_threaded_mainloop_free (mainloop);
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+ mainloop = NULL;
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+
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+ return 0;
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+}
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+
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+static void pulse_shutdown (void)
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+{
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+ pa_threaded_mainloop_lock (mainloop);
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+
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+ pa_context_disconnect (context);
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+ pa_context_unref (context);
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+ context = NULL;
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+
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+ pa_threaded_mainloop_unlock (mainloop);
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+
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+ pa_threaded_mainloop_stop (mainloop);
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+ pa_threaded_mainloop_free (mainloop);
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+ mainloop = NULL;
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+}
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+
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+static int pulse_open (struct sound_params *sound_params)
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+{
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+ pa_sample_spec ss;
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+ pa_buffer_attr ba;
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+ pa_stream *s;
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+
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+ assert (!stream);
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+ /* Initialize everything to -1, which in practice gets us
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+ * about 2 seconds of latency (which is fine). This is not the
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+ * same as passing NULL for this struct, which gets us an
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+ * unnecessarily short alsa-like latency.
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+ */
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+ ba.fragsize = (uint32_t) -1;
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+ ba.tlength = (uint32_t) -1;
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+ ba.prebuf = (uint32_t) -1;
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+ ba.minreq = (uint32_t) -1;
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+ ba.maxlength = (uint32_t) -1;
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+
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+ ss.channels = sound_params->channels;
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+ ss.rate = sound_params->rate;
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+ switch (sound_params->fmt) {
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+ case SFMT_U8:
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+ ss.format = PA_SAMPLE_U8;
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+ break;
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+ case SFMT_S16 | SFMT_LE:
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+ ss.format = PA_SAMPLE_S16LE;
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+ break;
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+ case SFMT_S16 | SFMT_BE:
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+ ss.format = PA_SAMPLE_S16BE;
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+ break;
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+ case SFMT_FLOAT | SFMT_LE:
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+ ss.format = PA_SAMPLE_FLOAT32LE;
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+ break;
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+ case SFMT_FLOAT | SFMT_BE:
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+ ss.format = PA_SAMPLE_FLOAT32BE;
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+ break;
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+ case SFMT_S32 | SFMT_LE:
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+ ss.format = PA_SAMPLE_S32LE;
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+ break;
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+ case SFMT_S32 | SFMT_BE:
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+ ss.format = PA_SAMPLE_S32BE;
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+ break;
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+
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+ default:
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+ fatal ("pulse: got unrequested format");
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+ }
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+
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+ debug ("opening stream");
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+
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+ pa_threaded_mainloop_lock (mainloop);
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+
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+ /* TODO: figure out if there are useful stream properties to set.
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+ *
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+ * I do not really see any in proplist.h that we can set from
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+ * here (there are media title/artist/etc props but we do not
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+ * have that data available here).
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+ */
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+ s = pa_stream_new (context, "music", &ss, NULL);
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+ if (!s)
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+ fatal ("pulse: stream allocation failed");
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+
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+ pa_stream_set_state_callback (s, stream_state_callback, mainloop);
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+ pa_stream_set_write_callback (s, stream_write_callback, mainloop);
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+
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+ /* Ignore return value, rely on failed stream state instead. */
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+ pa_stream_connect_playback (
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+ s, NULL, &ba,
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+ PA_STREAM_INTERPOLATE_TIMING |
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+ PA_STREAM_AUTO_TIMING_UPDATE |
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+ PA_STREAM_ADJUST_LATENCY,
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+ NULL, NULL);
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+
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+ while (1) {
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+ pa_stream_state_t state = pa_stream_get_state (s);
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+
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+ if (state == PA_STREAM_READY)
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+ break;
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+
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+ if (!PA_STREAM_IS_GOOD (state)) {
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+ error ("PulseAudio stream connection failed");
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+
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+ goto fail;
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+ }
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+
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+ debug ("waiting for stream to become ready...");
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+ pa_threaded_mainloop_wait (mainloop);
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+ }
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+
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+ /* Only set the global stream now that it is actually ready */
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+ stream = s;
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+
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+ pa_threaded_mainloop_unlock (mainloop);
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+
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+ return 1;
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+
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+fail:
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+ pa_stream_unref (s);
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+
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+ pa_threaded_mainloop_unlock (mainloop);
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+ return 0;
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+}
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+
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+static void pulse_close (void)
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+{
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+ debug ("closing stream");
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+
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+ pa_threaded_mainloop_lock (mainloop);
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+
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+ pa_stream_disconnect (stream);
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+ pa_stream_unref (stream);
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+ stream = NULL;
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+
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+ pa_threaded_mainloop_unlock (mainloop);
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+}
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+
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+static int pulse_play (const char *buff, const size_t size)
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+{
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+ size_t offset = 0;
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+
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+ debug ("Got %d bytes to play", (int)size);
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+
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+ pa_threaded_mainloop_lock (mainloop);
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+
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+ /* The buffer is usually writable when we get here, and there
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+ * are usually few (if any) writes after the first one. So
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+ * there is no point in doing further writes directly from the
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+ * callback: we can just do all writes from this thread.
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+ */
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+
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+ /* Break out of the loop if some other thread manages to close
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+ * our stream underneath us.
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+ */
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+ while (stream) {
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+ size_t towrite = MIN(pa_stream_writable_size (stream),
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+ size - offset);
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+ debug ("writing %d bytes", (int)towrite);
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+
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+ /* We have no working way of dealing with errors
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+ * (see below). */
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+ if (pa_stream_write(stream, buff + offset, towrite,
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+ NULL, 0, PA_SEEK_RELATIVE))
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+ error ("pa_stream_write failed");
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+
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+ offset += towrite;
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+
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+ if (offset >= size)
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+ break;
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+
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+ pa_threaded_mainloop_wait (mainloop);
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+ }
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+
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+ pa_threaded_mainloop_unlock (mainloop);
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+
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+ debug ("Done playing!");
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+
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+ /* We should always return size, calling code does not deal
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+ * well with anything else. Only read the rest if you want to
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+ * know why.
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+ *
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+ * The output buffer reader thread (out_buf.c:read_thread)
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+ * repeatedly loads some 64k/0.1s of audio into a buffer on
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+ * the stack, then calls audio_send_pcm repeatedly until this
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+ * 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
|