tjh.dev
moc: refactor options, add patches
Add support for conditional compile options, add pulseaudio and ffmpeg4
support patches.
+84
-9
pkgs/applications/audio/moc/default.nix
+84
-9
pkgs/applications/audio/moc/default.nix
···
1
-
{ stdenv, fetchurl, ncurses, pkgconfig, alsaLib, flac, libmad, speex, ffmpeg
2
-
, libvorbis, libmpc, libsndfile, libjack2, db, libmodplug, timidity, libid3tag
3
-
, libtool
4
}:
5
6
-
stdenv.mkDerivation rec {
7
name = "moc-${version}";
8
version = "2.5.2";
9
···
12
sha256 = "026v977kwb0wbmlmf6mnik328plxg8wykfx9ryvqhirac0aq39pk";
13
};
14
15
-
nativeBuildInputs = [ pkgconfig ];
16
17
-
buildInputs = [
18
-
ncurses alsaLib flac libmad speex ffmpeg libvorbis libmpc libsndfile libjack2
19
-
db libmodplug timidity libid3tag libtool
20
];
21
22
meta = with stdenv.lib; {
23
description = "An ncurses console audio player designed to be powerful and easy to use";
24
homepage = http://moc.daper.net/;
25
license = licenses.gpl2;
26
-
maintainers = with maintainers; [ pSub jagajaga ];
27
platforms = platforms.linux;
28
};
29
}
···
1
+
{ stdenv, fetchurl, pkgconfig
2
+
, ncurses, db , popt, libtool
3
+
# Sound sub-systems
4
+
, alsaSupport ? true, alsaLib
5
+
, pulseSupport ? true, libpulseaudio, autoreconfHook
6
+
, jackSupport ? true, libjack2
7
+
, ossSupport ? true
8
+
# Audio formats
9
+
, aacSupport ? true, faad2, libid3tag
10
+
, flacSupport ? true, flac
11
+
, midiSupport ? true, timidity
12
+
, modplugSupport ? true, libmodplug
13
+
, mp3Support ? true, libmad
14
+
, musepackSupport ? true, libmpc, libmpcdec, taglib
15
+
, vorbisSupport ? true, libvorbis
16
+
, speexSupport ? true, speex
17
+
, ffmpegSupport ? true, ffmpeg
18
+
, sndfileSupport ? true, libsndfile
19
+
, wavpackSupport ? true, wavpack
20
+
# Misc
21
+
, withffmpeg4 ? false, ffmpeg_4
22
+
, curlSupport ? true, curl
23
+
, samplerateSupport ? true, libsamplerate
24
+
, withDebug ? false
25
}:
26
27
+
let
28
+
opt = stdenv.lib.optional;
29
+
mkFlag = c: f: if c then "--with-${f}" else "--without-${f}";
30
+
31
+
in stdenv.mkDerivation rec {
32
+
33
name = "moc-${version}";
34
version = "2.5.2";
35
···
38
sha256 = "026v977kwb0wbmlmf6mnik328plxg8wykfx9ryvqhirac0aq39pk";
39
};
40
41
+
patches = []
42
+
++ opt withffmpeg4 ./moc-ffmpeg4.patch
43
+
++ opt pulseSupport ./pulseaudio.patch;
44
+
45
+
nativeBuildInputs = [ pkgconfig ]
46
+
++ opt pulseSupport autoreconfHook;
47
+
48
+
buildInputs = [ ncurses db popt libtool ]
49
+
# Sound sub-systems
50
+
++ opt alsaSupport alsaLib
51
+
++ opt pulseSupport libpulseaudio
52
+
++ opt jackSupport libjack2
53
+
# Audio formats
54
+
++ opt (aacSupport || mp3Support) libid3tag
55
+
++ opt aacSupport faad2
56
+
++ opt flacSupport flac
57
+
++ opt midiSupport timidity
58
+
++ opt modplugSupport libmodplug
59
+
++ opt mp3Support libmad
60
+
++ opt musepackSupport [ libmpc libmpcdec taglib ]
61
+
++ opt vorbisSupport libvorbis
62
+
++ opt speexSupport speex
63
+
++ opt (ffmpegSupport && !withffmpeg4) ffmpeg
64
+
++ opt (ffmpegSupport && withffmpeg4) ffmpeg_4
65
+
++ opt sndfileSupport libsndfile
66
+
++ opt wavpackSupport wavpack
67
+
# Misc
68
+
++ opt curlSupport curl
69
+
++ opt samplerateSupport libsamplerate;
70
71
+
configureFlags = [
72
+
# Sound sub-systems
73
+
(mkFlag alsaSupport "alsa")
74
+
(mkFlag pulseSupport "pulse")
75
+
(mkFlag jackSupport "jack")
76
+
(mkFlag ossSupport "oss")
77
+
# Audio formats
78
+
(mkFlag aacSupport "aac")
79
+
(mkFlag flacSupport "flac")
80
+
(mkFlag midiSupport "timidity")
81
+
(mkFlag modplugSupport "modplug")
82
+
(mkFlag mp3Support "mp3")
83
+
(mkFlag musepackSupport "musepack")
84
+
(mkFlag vorbisSupport "vorbis")
85
+
(mkFlag speexSupport "speex")
86
+
(mkFlag ffmpegSupport "ffmpeg")
87
+
(mkFlag sndfileSupport "sndfile")
88
+
(mkFlag wavpackSupport "wavpack")
89
+
# Misc
90
+
(mkFlag curlSupport "curl")
91
+
(mkFlag samplerateSupport "samplerate")
92
+
("--enable-debug=" + (if withDebug then "yes" else "no"))
93
+
"--disable-cache"
94
+
"--without-rcc"
95
];
96
97
meta = with stdenv.lib; {
98
description = "An ncurses console audio player designed to be powerful and easy to use";
99
homepage = http://moc.daper.net/;
100
license = licenses.gpl2;
101
+
maintainers = with maintainers; [ aethelz pSub jagajaga ];
102
platforms = platforms.linux;
103
};
104
}
+33
pkgs/applications/audio/moc/moc-ffmpeg4.patch
+33
pkgs/applications/audio/moc/moc-ffmpeg4.patch
···
···
1
+
Index: decoder_plugins/ffmpeg/ffmpeg.c
2
+
===================================================================
3
+
--- /decoder_plugins/ffmpeg/ffmpeg.c (revisión: 2963)
4
+
+++ /decoder_plugins/ffmpeg/ffmpeg.c (copia de trabajo)
5
+
@@ -697,7 +697,7 @@
6
+
* FFmpeg/LibAV in use. For some versions this will be caught in
7
+
* *_find_stream_info() above and misreported as an unfound codec
8
+
* parameters error. */
9
+
- if (data->codec->capabilities & CODEC_CAP_EXPERIMENTAL) {
10
+
+ if (data->codec->capabilities & AV_CODEC_CAP_EXPERIMENTAL) {
11
+
decoder_error (&data->error, ERROR_FATAL, 0,
12
+
"The codec is experimental and may damage MOC: %s",
13
+
data->codec->name);
14
+
@@ -705,8 +705,8 @@
15
+
}
16
+
17
+
set_downmixing (data);
18
+
- if (data->codec->capabilities & CODEC_CAP_TRUNCATED)
19
+
- data->enc->flags |= CODEC_FLAG_TRUNCATED;
20
+
+ if (data->codec->capabilities & AV_CODEC_CAP_TRUNCATED)
21
+
+ data->enc->flags |= AV_CODEC_FLAG_TRUNCATED;
22
+
23
+
if (avcodec_open2 (data->enc, data->codec, NULL) < 0)
24
+
{
25
+
@@ -725,7 +725,7 @@
26
+
27
+
data->sample_width = sfmt_Bps (data->fmt);
28
+
29
+
- if (data->codec->capabilities & CODEC_CAP_DELAY)
30
+
+ if (data->codec->capabilities & AV_CODEC_CAP_DELAY)
31
+
data->delay = true;
32
+
data->seek_broken = is_seek_broken (data);
33
+
data->timing_broken = is_timing_broken (data->ic);
+800
pkgs/applications/audio/moc/pulseaudio.patch
+800
pkgs/applications/audio/moc/pulseaudio.patch
···
···
1
+
diff --git a/audio.c b/audio.c
2
+
--- a/audio.c
3
+
+++ b/audio.c
4
+
@@ -32,6 +32,9 @@
5
+
#include "log.h"
6
+
#include "lists.h"
7
+
8
+
+#ifdef HAVE_PULSE
9
+
+# include "pulse.h"
10
+
+#endif
11
+
#ifdef HAVE_OSS
12
+
# include "oss.h"
13
+
#endif
14
+
@@ -893,6 +896,15 @@
15
+
}
16
+
#endif
17
+
18
+
+#ifdef HAVE_PULSE
19
+
+ if (!strcasecmp(name, "pulseaudio")) {
20
+
+ pulse_funcs (funcs);
21
+
+ printf ("Trying PulseAudio...\n");
22
+
+ if (funcs->init(&hw_caps))
23
+
+ return;
24
+
+ }
25
+
+#endif
26
+
+
27
+
#ifdef HAVE_OSS
28
+
if (!strcasecmp(name, "oss")) {
29
+
oss_funcs (funcs);
30
+
diff --git a/configure.in b/configure.in
31
+
--- a/configure.in
32
+
+++ b/configure.in
33
+
@@ -162,6 +162,21 @@
34
+
AC_MSG_ERROR([BerkeleyDB (libdb) not found.]))
35
+
fi
36
+
37
+
+AC_ARG_WITH(pulse, AS_HELP_STRING(--without-pulse,
38
+
+ Compile without PulseAudio support.))
39
+
+
40
+
+if test "x$with_pulse" != "xno"
41
+
+then
42
+
+ PKG_CHECK_MODULES(PULSE, [libpulse],
43
+
+ [SOUND_DRIVERS="$SOUND_DRIVERS PULSE"
44
+
+ EXTRA_OBJS="$EXTRA_OBJS pulse.o"
45
+
+ AC_DEFINE([HAVE_PULSE], 1, [Define if you have PulseAudio.])
46
+
+ EXTRA_LIBS="$EXTRA_LIBS $PULSE_LIBS"
47
+
+ CFLAGS="$CFLAGS $PULSE_CFLAGS"],
48
+
+ [true])
49
+
+fi
50
+
+
51
+
+
52
+
AC_ARG_WITH(oss, AS_HELP_STRING([--without-oss],
53
+
[Compile without OSS support]))
54
+
55
+
diff --git a/options.c b/options.c
56
+
--- a/options.c
57
+
+++ b/options.c
58
+
@@ -572,10 +572,11 @@
59
+
60
+
#ifdef OPENBSD
61
+
add_list ("SoundDriver", "SNDIO:JACK:OSS",
62
+
- CHECK_DISCRETE(5), "SNDIO", "Jack", "ALSA", "OSS", "null");
63
+
+ CHECK_DISCRETE(5), "SNDIO", "PulseAudio", "Jack", "ALSA", "OSS", "null");
64
+
+
65
+
#else
66
+
add_list ("SoundDriver", "Jack:ALSA:OSS",
67
+
- CHECK_DISCRETE(5), "SNDIO", "Jack", "ALSA", "OSS", "null");
68
+
+ CHECK_DISCRETE(5), "SNDIO", "PulseAudio", "Jack", "ALSA", "OSS", "null");
69
+
#endif
70
+
71
+
add_str ("JackClientName", "moc", CHECK_NONE);
72
+
diff --git a/pulse.c b/pulse.c
73
+
new file mode 100644
74
+
--- /dev/null
75
+
+++ b/pulse.c
76
+
@@ -0,0 +1,705 @@
77
+
+/*
78
+
+ * MOC - music on console
79
+
+ * Copyright (C) 2011 Marien Zwart <marienz@marienz.net>
80
+
+ *
81
+
+ * This program is free software; you can redistribute it and/or modify
82
+
+ * it under the terms of the GNU General Public License as published by
83
+
+ * the Free Software Foundation; either version 2 of the License, or
84
+
+ * (at your option) any later version.
85
+
+ *
86
+
+ */
87
+
+
88
+
+/* PulseAudio backend.
89
+
+ *
90
+
+ * FEATURES:
91
+
+ *
92
+
+ * Does not autostart a PulseAudio server, but uses an already-started
93
+
+ * one, which should be better than alsa-through-pulse.
94
+
+ *
95
+
+ * Supports control of either our stream's or our entire sink's volume
96
+
+ * while we are actually playing. Volume control while paused is
97
+
+ * intentionally unsupported: the PulseAudio documentation strongly
98
+
+ * suggests not passing in an initial volume when creating a stream
99
+
+ * (allowing the server to track this instead), and we do not know
100
+
+ * which sink to control if we do not have a stream open.
101
+
+ *
102
+
+ * IMPLEMENTATION:
103
+
+ *
104
+
+ * Most client-side (resource allocation) errors are fatal. Failure to
105
+
+ * create a server context or stream is not fatal (and MOC should cope
106
+
+ * with these failures too), but server communication failures later
107
+
+ * on are currently not handled (MOC has no great way for us to tell
108
+
+ * it we no longer work, and I am not sure if attempting to reconnect
109
+
+ * is worth it or even a good idea).
110
+
+ *
111
+
+ * The pulse "simple" API is too simple: it combines connecting to the
112
+
+ * server and opening a stream into one operation, while I want to
113
+
+ * connect to the server when MOC starts (and fall back to a different
114
+
+ * backend if there is no server), and I cannot open a stream at that
115
+
+ * time since I do not know the audio format yet.
116
+
+ *
117
+
+ * PulseAudio strongly recommends we use a high-latency connection,
118
+
+ * which the MOC frontend code might not expect from its audio
119
+
+ * backend. We'll see.
120
+
+ *
121
+
+ * We map MOC's percentage volumes linearly to pulse's PA_VOLUME_MUTED
122
+
+ * (0) .. PA_VOLUME_NORM range. This is what the PulseAudio docs recommend
123
+
+ * ( http://pulseaudio.org/wiki/WritingVolumeControlUIs ). It does mean
124
+
+ * PulseAudio volumes above PA_VOLUME_NORM do not work well with MOC.
125
+
+ *
126
+
+ * Comments in audio.h claim "All functions are executed only by one
127
+
+ * thread" (referring to the function in the hw_funcs struct). This is
128
+
+ * a blatant lie. Most of them are invoked off the "output buffer"
129
+
+ * thread (out_buf.c) but at least the "playing" thread (audio.c)
130
+
+ * calls audio_close which calls our close function. We can mostly
131
+
+ * ignore this problem because we serialize on the pulseaudio threaded
132
+
+ * mainloop lock. But it does mean that functions that are normally
133
+
+ * only called between open and close (like reset) are sometimes
134
+
+ * called without us having a stream. Bulletproof, therefore:
135
+
+ * serialize setting/unsetting our global stream using the threaded
136
+
+ * mainloop lock, and check for that stream being non-null before
137
+
+ * using it.
138
+
+ *
139
+
+ * I am not convinced there are no further dragons lurking here: can
140
+
+ * the "playing" thread(s) close and reopen our output stream while
141
+
+ * the "output buffer" thread is sending output there? We can bail if
142
+
+ * our stream is simply closed, but we do not currently detect it
143
+
+ * being reopened and no longer using the same sample format, which
144
+
+ * might have interesting results...
145
+
+ *
146
+
+ * Also, read_mixer is called from the main server thread (handling
147
+
+ * commands). This crashed me once when it got at a stream that was in
148
+
+ * the "creating" state and therefore did not have a valid stream
149
+
+ * index yet. Fixed by only assigning to the stream global when the
150
+
+ * stream is valid.
151
+
+ */
152
+
+
153
+
+#ifdef HAVE_CONFIG_H
154
+
+# include "config.h"
155
+
+#endif
156
+
+
157
+
+#define DEBUG
158
+
+
159
+
+#include <pulse/pulseaudio.h>
160
+
+#include "common.h"
161
+
+#include "log.h"
162
+
+#include "audio.h"
163
+
+
164
+
+
165
+
+/* The pulse mainloop and context are initialized in pulse_init and
166
+
+ * destroyed in pulse_shutdown.
167
+
+ */
168
+
+static pa_threaded_mainloop *mainloop = NULL;
169
+
+static pa_context *context = NULL;
170
+
+
171
+
+/* The stream is initialized in pulse_open and destroyed in pulse_close. */
172
+
+static pa_stream *stream = NULL;
173
+
+
174
+
+static int showing_sink_volume = 0;
175
+
+
176
+
+/* Callbacks that do nothing but wake up the mainloop. */
177
+
+
178
+
+static void context_state_callback (pa_context *context ATTR_UNUSED,
179
+
+ void *userdata)
180
+
+{
181
+
+ pa_threaded_mainloop *m = userdata;
182
+
+
183
+
+ pa_threaded_mainloop_signal (m, 0);
184
+
+}
185
+
+
186
+
+static void stream_state_callback (pa_stream *stream ATTR_UNUSED,
187
+
+ void *userdata)
188
+
+{
189
+
+ pa_threaded_mainloop *m = userdata;
190
+
+
191
+
+ pa_threaded_mainloop_signal (m, 0);
192
+
+}
193
+
+
194
+
+static void stream_write_callback (pa_stream *stream ATTR_UNUSED,
195
+
+ size_t nbytes ATTR_UNUSED, void *userdata)
196
+
+{
197
+
+ pa_threaded_mainloop *m = userdata;
198
+
+
199
+
+ pa_threaded_mainloop_signal (m, 0);
200
+
+}
201
+
+
202
+
+/* Initialize pulse mainloop and context. Failure to connect to the
203
+
+ * pulse daemon is nonfatal, everything else is fatal (as it
204
+
+ * presumably means we ran out of resources).
205
+
+ */
206
+
+static int pulse_init (struct output_driver_caps *caps)
207
+
+{
208
+
+ pa_context *c;
209
+
+ pa_proplist *proplist;
210
+
+
211
+
+ assert (!mainloop);
212
+
+ assert (!context);
213
+
+
214
+
+ mainloop = pa_threaded_mainloop_new ();
215
+
+ if (!mainloop)
216
+
+ fatal ("Cannot create PulseAudio mainloop");
217
+
+
218
+
+ if (pa_threaded_mainloop_start (mainloop) < 0)
219
+
+ fatal ("Cannot start PulseAudio mainloop");
220
+
+
221
+
+ /* TODO: possibly add more props.
222
+
+ *
223
+
+ * There are a few we could set in proplist.h but nothing I
224
+
+ * expect to be very useful.
225
+
+ *
226
+
+ * http://pulseaudio.org/wiki/ApplicationProperties recommends
227
+
+ * setting at least application.name, icon.name and media.role.
228
+
+ *
229
+
+ * No need to set application.name here, the name passed to
230
+
+ * pa_context_new_with_proplist overrides it.
231
+
+ */
232
+
+ proplist = pa_proplist_new ();
233
+
+ if (!proplist)
234
+
+ fatal ("Cannot allocate PulseAudio proplist");
235
+
+
236
+
+ pa_proplist_sets (proplist,
237
+
+ PA_PROP_APPLICATION_VERSION, PACKAGE_VERSION);
238
+
+ pa_proplist_sets (proplist, PA_PROP_MEDIA_ROLE, "music");
239
+
+ pa_proplist_sets (proplist, PA_PROP_APPLICATION_ID, "net.daper.moc");
240
+
+
241
+
+ pa_threaded_mainloop_lock (mainloop);
242
+
+
243
+
+ c = pa_context_new_with_proplist (
244
+
+ pa_threaded_mainloop_get_api (mainloop),
245
+
+ PACKAGE_NAME, proplist);
246
+
+ pa_proplist_free (proplist);
247
+
+
248
+
+ if (!c)
249
+
+ fatal ("Cannot allocate PulseAudio context");
250
+
+
251
+
+ pa_context_set_state_callback (c, context_state_callback, mainloop);
252
+
+
253
+
+ /* Ignore return value, rely on state being set properly */
254
+
+ pa_context_connect (c, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL);
255
+
+
256
+
+ while (1) {
257
+
+ pa_context_state_t state = pa_context_get_state (c);
258
+
+
259
+
+ if (state == PA_CONTEXT_READY)
260
+
+ break;
261
+
+
262
+
+ if (!PA_CONTEXT_IS_GOOD (state)) {
263
+
+ error ("PulseAudio connection failed: %s",
264
+
+ pa_strerror (pa_context_errno (c)));
265
+
+
266
+
+ goto unlock_and_fail;
267
+
+ }
268
+
+
269
+
+ debug ("waiting for context to become ready...");
270
+
+ pa_threaded_mainloop_wait (mainloop);
271
+
+ }
272
+
+
273
+
+ /* Only set the global now that the context is actually ready */
274
+
+ context = c;
275
+
+
276
+
+ pa_threaded_mainloop_unlock (mainloop);
277
+
+
278
+
+ /* We just make up the hardware capabilities, since pulse is
279
+
+ * supposed to be abstracting these out. Assume pulse will
280
+
+ * deal with anything we want to throw at it, and that we will
281
+
+ * only want mono or stereo audio.
282
+
+ */
283
+
+ caps->min_channels = 1;
284
+
+ caps->max_channels = 2;
285
+
+ caps->formats = (SFMT_S8 | SFMT_S16 | SFMT_S32 |
286
+
+ SFMT_FLOAT | SFMT_BE | SFMT_LE);
287
+
+
288
+
+ return 1;
289
+
+
290
+
+unlock_and_fail:
291
+
+
292
+
+ pa_context_unref (c);
293
+
+
294
+
+ pa_threaded_mainloop_unlock (mainloop);
295
+
+
296
+
+ pa_threaded_mainloop_stop (mainloop);
297
+
+ pa_threaded_mainloop_free (mainloop);
298
+
+ mainloop = NULL;
299
+
+
300
+
+ return 0;
301
+
+}
302
+
+
303
+
+static void pulse_shutdown (void)
304
+
+{
305
+
+ pa_threaded_mainloop_lock (mainloop);
306
+
+
307
+
+ pa_context_disconnect (context);
308
+
+ pa_context_unref (context);
309
+
+ context = NULL;
310
+
+
311
+
+ pa_threaded_mainloop_unlock (mainloop);
312
+
+
313
+
+ pa_threaded_mainloop_stop (mainloop);
314
+
+ pa_threaded_mainloop_free (mainloop);
315
+
+ mainloop = NULL;
316
+
+}
317
+
+
318
+
+static int pulse_open (struct sound_params *sound_params)
319
+
+{
320
+
+ pa_sample_spec ss;
321
+
+ pa_buffer_attr ba;
322
+
+ pa_stream *s;
323
+
+
324
+
+ assert (!stream);
325
+
+ /* Initialize everything to -1, which in practice gets us
326
+
+ * about 2 seconds of latency (which is fine). This is not the
327
+
+ * same as passing NULL for this struct, which gets us an
328
+
+ * unnecessarily short alsa-like latency.
329
+
+ */
330
+
+ ba.fragsize = (uint32_t) -1;
331
+
+ ba.tlength = (uint32_t) -1;
332
+
+ ba.prebuf = (uint32_t) -1;
333
+
+ ba.minreq = (uint32_t) -1;
334
+
+ ba.maxlength = (uint32_t) -1;
335
+
+
336
+
+ ss.channels = sound_params->channels;
337
+
+ ss.rate = sound_params->rate;
338
+
+ switch (sound_params->fmt) {
339
+
+ case SFMT_U8:
340
+
+ ss.format = PA_SAMPLE_U8;
341
+
+ break;
342
+
+ case SFMT_S16 | SFMT_LE:
343
+
+ ss.format = PA_SAMPLE_S16LE;
344
+
+ break;
345
+
+ case SFMT_S16 | SFMT_BE:
346
+
+ ss.format = PA_SAMPLE_S16BE;
347
+
+ break;
348
+
+ case SFMT_FLOAT | SFMT_LE:
349
+
+ ss.format = PA_SAMPLE_FLOAT32LE;
350
+
+ break;
351
+
+ case SFMT_FLOAT | SFMT_BE:
352
+
+ ss.format = PA_SAMPLE_FLOAT32BE;
353
+
+ break;
354
+
+ case SFMT_S32 | SFMT_LE:
355
+
+ ss.format = PA_SAMPLE_S32LE;
356
+
+ break;
357
+
+ case SFMT_S32 | SFMT_BE:
358
+
+ ss.format = PA_SAMPLE_S32BE;
359
+
+ break;
360
+
+
361
+
+ default:
362
+
+ fatal ("pulse: got unrequested format");
363
+
+ }
364
+
+
365
+
+ debug ("opening stream");
366
+
+
367
+
+ pa_threaded_mainloop_lock (mainloop);
368
+
+
369
+
+ /* TODO: figure out if there are useful stream properties to set.
370
+
+ *
371
+
+ * I do not really see any in proplist.h that we can set from
372
+
+ * here (there are media title/artist/etc props but we do not
373
+
+ * have that data available here).
374
+
+ */
375
+
+ s = pa_stream_new (context, "music", &ss, NULL);
376
+
+ if (!s)
377
+
+ fatal ("pulse: stream allocation failed");
378
+
+
379
+
+ pa_stream_set_state_callback (s, stream_state_callback, mainloop);
380
+
+ pa_stream_set_write_callback (s, stream_write_callback, mainloop);
381
+
+
382
+
+ /* Ignore return value, rely on failed stream state instead. */
383
+
+ pa_stream_connect_playback (
384
+
+ s, NULL, &ba,
385
+
+ PA_STREAM_INTERPOLATE_TIMING |
386
+
+ PA_STREAM_AUTO_TIMING_UPDATE |
387
+
+ PA_STREAM_ADJUST_LATENCY,
388
+
+ NULL, NULL);
389
+
+
390
+
+ while (1) {
391
+
+ pa_stream_state_t state = pa_stream_get_state (s);
392
+
+
393
+
+ if (state == PA_STREAM_READY)
394
+
+ break;
395
+
+
396
+
+ if (!PA_STREAM_IS_GOOD (state)) {
397
+
+ error ("PulseAudio stream connection failed");
398
+
+
399
+
+ goto fail;
400
+
+ }
401
+
+
402
+
+ debug ("waiting for stream to become ready...");
403
+
+ pa_threaded_mainloop_wait (mainloop);
404
+
+ }
405
+
+
406
+
+ /* Only set the global stream now that it is actually ready */
407
+
+ stream = s;
408
+
+
409
+
+ pa_threaded_mainloop_unlock (mainloop);
410
+
+
411
+
+ return 1;
412
+
+
413
+
+fail:
414
+
+ pa_stream_unref (s);
415
+
+
416
+
+ pa_threaded_mainloop_unlock (mainloop);
417
+
+ return 0;
418
+
+}
419
+
+
420
+
+static void pulse_close (void)
421
+
+{
422
+
+ debug ("closing stream");
423
+
+
424
+
+ pa_threaded_mainloop_lock (mainloop);
425
+
+
426
+
+ pa_stream_disconnect (stream);
427
+
+ pa_stream_unref (stream);
428
+
+ stream = NULL;
429
+
+
430
+
+ pa_threaded_mainloop_unlock (mainloop);
431
+
+}
432
+
+
433
+
+static int pulse_play (const char *buff, const size_t size)
434
+
+{
435
+
+ size_t offset = 0;
436
+
+
437
+
+ debug ("Got %d bytes to play", (int)size);
438
+
+
439
+
+ pa_threaded_mainloop_lock (mainloop);
440
+
+
441
+
+ /* The buffer is usually writable when we get here, and there
442
+
+ * are usually few (if any) writes after the first one. So
443
+
+ * there is no point in doing further writes directly from the
444
+
+ * callback: we can just do all writes from this thread.
445
+
+ */
446
+
+
447
+
+ /* Break out of the loop if some other thread manages to close
448
+
+ * our stream underneath us.
449
+
+ */
450
+
+ while (stream) {
451
+
+ size_t towrite = MIN(pa_stream_writable_size (stream),
452
+
+ size - offset);
453
+
+ debug ("writing %d bytes", (int)towrite);
454
+
+
455
+
+ /* We have no working way of dealing with errors
456
+
+ * (see below). */
457
+
+ if (pa_stream_write(stream, buff + offset, towrite,
458
+
+ NULL, 0, PA_SEEK_RELATIVE))
459
+
+ error ("pa_stream_write failed");
460
+
+
461
+
+ offset += towrite;
462
+
+
463
+
+ if (offset >= size)
464
+
+ break;
465
+
+
466
+
+ pa_threaded_mainloop_wait (mainloop);
467
+
+ }
468
+
+
469
+
+ pa_threaded_mainloop_unlock (mainloop);
470
+
+
471
+
+ debug ("Done playing!");
472
+
+
473
+
+ /* We should always return size, calling code does not deal
474
+
+ * well with anything else. Only read the rest if you want to
475
+
+ * know why.
476
+
+ *
477
+
+ * The output buffer reader thread (out_buf.c:read_thread)
478
+
+ * repeatedly loads some 64k/0.1s of audio into a buffer on
479
+
+ * the stack, then calls audio_send_pcm repeatedly until this
480
+
+ * entire buffer has been processed (similar to the loop in
481
+
+ * this function). audio_send_pcm applies the softmixer and
482
+
+ * equalizer, then feeds the result to this function, passing
483
+
+ * through our return value.
484
+
+ *
485
+
+ * So if we return less than size the equalizer/softmixer is
486
+
+ * re-applied to the remaining data, which is silly. Also,
487
+
+ * audio_send_pcm checks for our return value being zero and
488
+
+ * calls fatal() if it is, so try to always process *some*
489
+
+ * data. Also, out_buf.c uses the return value of this
490
+
+ * function from the last run through its inner loop to update
491
+
+ * its time attribute, which means it will be interestingly
492
+
+ * off if that loop ran more than once.
493
+
+ *
494
+
+ * Oh, and alsa.c seems to think it can return -1 to indicate
495
+
+ * failure, which will cause out_buf.c to rewind its buffer
496
+
+ * (to before its start, usually).
497
+
+ */
498
+
+ return size;
499
+
+}
500
+
+
501
+
+static void volume_cb (const pa_cvolume *v, void *userdata)
502
+
+{
503
+
+ int *result = userdata;
504
+
+
505
+
+ if (v)
506
+
+ *result = 100 * pa_cvolume_avg (v) / PA_VOLUME_NORM;
507
+
+
508
+
+ pa_threaded_mainloop_signal (mainloop, 0);
509
+
+}
510
+
+
511
+
+static void sink_volume_cb (pa_context *c ATTR_UNUSED,
512
+
+ const pa_sink_info *i, int eol ATTR_UNUSED,
513
+
+ void *userdata)
514
+
+{
515
+
+ volume_cb (i ? &i->volume : NULL, userdata);
516
+
+}
517
+
+
518
+
+static void sink_input_volume_cb (pa_context *c ATTR_UNUSED,
519
+
+ const pa_sink_input_info *i,
520
+
+ int eol ATTR_UNUSED,
521
+
+ void *userdata ATTR_UNUSED)
522
+
+{
523
+
+ volume_cb (i ? &i->volume : NULL, userdata);
524
+
+}
525
+
+
526
+
+static int pulse_read_mixer (void)
527
+
+{
528
+
+ pa_operation *op;
529
+
+ int result = 0;
530
+
+
531
+
+ debug ("read mixer");
532
+
+
533
+
+ pa_threaded_mainloop_lock (mainloop);
534
+
+
535
+
+ if (stream) {
536
+
+ if (showing_sink_volume)
537
+
+ op = pa_context_get_sink_info_by_index (
538
+
+ context, pa_stream_get_device_index (stream),
539
+
+ sink_volume_cb, &result);
540
+
+ else
541
+
+ op = pa_context_get_sink_input_info (
542
+
+ context, pa_stream_get_index (stream),
543
+
+ sink_input_volume_cb, &result);
544
+
+
545
+
+ while (pa_operation_get_state (op) == PA_OPERATION_RUNNING)
546
+
+ pa_threaded_mainloop_wait (mainloop);
547
+
+
548
+
+ pa_operation_unref (op);
549
+
+ }
550
+
+
551
+
+ pa_threaded_mainloop_unlock (mainloop);
552
+
+
553
+
+ return result;
554
+
+}
555
+
+
556
+
+static void pulse_set_mixer (int vol)
557
+
+{
558
+
+ pa_cvolume v;
559
+
+ pa_operation *op;
560
+
+
561
+
+ /* Setting volume for one channel does the right thing. */
562
+
+ pa_cvolume_set(&v, 1, vol * PA_VOLUME_NORM / 100);
563
+
+
564
+
+ pa_threaded_mainloop_lock (mainloop);
565
+
+
566
+
+ if (stream) {
567
+
+ if (showing_sink_volume)
568
+
+ op = pa_context_set_sink_volume_by_index (
569
+
+ context, pa_stream_get_device_index (stream),
570
+
+ &v, NULL, NULL);
571
+
+ else
572
+
+ op = pa_context_set_sink_input_volume (
573
+
+ context, pa_stream_get_index (stream),
574
+
+ &v, NULL, NULL);
575
+
+
576
+
+ pa_operation_unref (op);
577
+
+ }
578
+
+
579
+
+ pa_threaded_mainloop_unlock (mainloop);
580
+
+}
581
+
+
582
+
+static int pulse_get_buff_fill (void)
583
+
+{
584
+
+ /* This function is problematic. MOC uses it to for the "time
585
+
+ * remaining" in the UI, but calls it more than once per
586
+
+ * second (after each chunk of audio played, not for each
587
+
+ * playback time update). We have to be fairly accurate here
588
+
+ * for that time remaining to not jump weirdly. But PulseAudio
589
+
+ * cannot give us a 100% accurate value here, as it involves a
590
+
+ * server roundtrip. And if we call this a lot it suggests
591
+
+ * switching to a mode where the value is interpolated, making
592
+
+ * it presumably more inaccurate (see the flags we pass to
593
+
+ * pa_stream_connect_playback).
594
+
+ *
595
+
+ * MOC also contains what I believe to be a race: it calls
596
+
+ * audio_get_buff_fill "soon" (after playing the first chunk)
597
+
+ * after starting playback of the next song, at which point we
598
+
+ * still have part of the previous song buffered. This means
599
+
+ * our position into the new song is negative, which fails an
600
+
+ * assert (in out_buf.c:out_buf_time_get). There is no sane
601
+
+ * way for us to detect this condition. I believe no other
602
+
+ * backend triggers this because the assert sits after an
603
+
+ * implicit float -> int seconds conversion, which means we
604
+
+ * have to be off by at least an entire second to get a
605
+
+ * negative value, and none of the other backends have buffers
606
+
+ * that large (alsa buffers are supposedly a few 100 ms).
607
+
+ */
608
+
+ pa_usec_t buffered_usecs = 0;
609
+
+ int buffered_bytes = 0;
610
+
+
611
+
+ pa_threaded_mainloop_lock (mainloop);
612
+
+
613
+
+ /* Using pa_stream_get_timing_info and returning the distance
614
+
+ * between write_index and read_index would be more obvious,
615
+
+ * but because of how the result is actually used I believe
616
+
+ * using the latency value is slightly more correct, and it
617
+
+ * makes the following crash-avoidance hack more obvious.
618
+
+ */
619
+
+
620
+
+ /* This function will frequently fail the first time we call
621
+
+ * it (pulse does not have the requested data yet). We ignore
622
+
+ * that and just return 0.
623
+
+ *
624
+
+ * Deal with stream being NULL too, just in case this is
625
+
+ * called in a racy fashion similar to how reset() is.
626
+
+ */
627
+
+ if (stream &&
628
+
+ pa_stream_get_latency (stream, &buffered_usecs, NULL) >= 0) {
629
+
+ /* Crash-avoidance HACK: floor our latency to at most
630
+
+ * 1 second. It is usually more, but reporting that at
631
+
+ * the start of playback crashes MOC, and we cannot
632
+
+ * sanely detect when reporting it is safe.
633
+
+ */
634
+
+ if (buffered_usecs > 1000000)
635
+
+ buffered_usecs = 1000000;
636
+
+
637
+
+ buffered_bytes = pa_usec_to_bytes (
638
+
+ buffered_usecs,
639
+
+ pa_stream_get_sample_spec (stream));
640
+
+ }
641
+
+
642
+
+ pa_threaded_mainloop_unlock (mainloop);
643
+
+
644
+
+ debug ("buffer fill: %d usec / %d bytes",
645
+
+ (int) buffered_usecs, (int) buffered_bytes);
646
+
+
647
+
+ return buffered_bytes;
648
+
+}
649
+
+
650
+
+static void flush_callback (pa_stream *s ATTR_UNUSED, int success,
651
+
+ void *userdata)
652
+
+{
653
+
+ int *result = userdata;
654
+
+
655
+
+ *result = success;
656
+
+
657
+
+ pa_threaded_mainloop_signal (mainloop, 0);
658
+
+}
659
+
+
660
+
+static int pulse_reset (void)
661
+
+{
662
+
+ pa_operation *op;
663
+
+ int result = 0;
664
+
+
665
+
+ debug ("reset requested");
666
+
+
667
+
+ pa_threaded_mainloop_lock (mainloop);
668
+
+
669
+
+ /* We *should* have a stream here, but MOC is racy, so bulletproof */
670
+
+ if (stream) {
671
+
+ op = pa_stream_flush (stream, flush_callback, &result);
672
+
+
673
+
+ while (pa_operation_get_state (op) == PA_OPERATION_RUNNING)
674
+
+ pa_threaded_mainloop_wait (mainloop);
675
+
+
676
+
+ pa_operation_unref (op);
677
+
+ } else
678
+
+ logit ("pulse_reset() called without a stream");
679
+
+
680
+
+ pa_threaded_mainloop_unlock (mainloop);
681
+
+
682
+
+ return result;
683
+
+}
684
+
+
685
+
+static int pulse_get_rate (void)
686
+
+{
687
+
+ /* This is called once right after open. Do not bother making
688
+
+ * this fast. */
689
+
+
690
+
+ int result;
691
+
+
692
+
+ pa_threaded_mainloop_lock (mainloop);
693
+
+
694
+
+ if (stream)
695
+
+ result = pa_stream_get_sample_spec (stream)->rate;
696
+
+ else {
697
+
+ error ("get_rate called without a stream");
698
+
+ result = 0;
699
+
+ }
700
+
+
701
+
+ pa_threaded_mainloop_unlock (mainloop);
702
+
+
703
+
+ return result;
704
+
+}
705
+
+
706
+
+static void pulse_toggle_mixer_channel (void)
707
+
+{
708
+
+ showing_sink_volume = !showing_sink_volume;
709
+
+}
710
+
+
711
+
+static void sink_name_cb (pa_context *c ATTR_UNUSED,
712
+
+ const pa_sink_info *i, int eol ATTR_UNUSED,
713
+
+ void *userdata)
714
+
+{
715
+
+ char **result = userdata;
716
+
+
717
+
+ if (i && !*result)
718
+
+ *result = xstrdup (i->name);
719
+
+
720
+
+ pa_threaded_mainloop_signal (mainloop, 0);
721
+
+}
722
+
+
723
+
+static void sink_input_name_cb (pa_context *c ATTR_UNUSED,
724
+
+ const pa_sink_input_info *i,
725
+
+ int eol ATTR_UNUSED,
726
+
+ void *userdata)
727
+
+{
728
+
+ char **result = userdata;
729
+
+
730
+
+ if (i && !*result)
731
+
+ *result = xstrdup (i->name);
732
+
+
733
+
+ pa_threaded_mainloop_signal (mainloop, 0);
734
+
+}
735
+
+
736
+
+static char *pulse_get_mixer_channel_name (void)
737
+
+{
738
+
+ char *result = NULL;
739
+
+ pa_operation *op;
740
+
+
741
+
+ pa_threaded_mainloop_lock (mainloop);
742
+
+
743
+
+ if (stream) {
744
+
+ if (showing_sink_volume)
745
+
+ op = pa_context_get_sink_info_by_index (
746
+
+ context, pa_stream_get_device_index (stream),
747
+
+ sink_name_cb, &result);
748
+
+ else
749
+
+ op = pa_context_get_sink_input_info (
750
+
+ context, pa_stream_get_index (stream),
751
+
+ sink_input_name_cb, &result);
752
+
+
753
+
+ while (pa_operation_get_state (op) == PA_OPERATION_RUNNING)
754
+
+ pa_threaded_mainloop_wait (mainloop);
755
+
+
756
+
+ pa_operation_unref (op);
757
+
+ }
758
+
+
759
+
+ pa_threaded_mainloop_unlock (mainloop);
760
+
+
761
+
+ if (!result)
762
+
+ result = xstrdup ("disconnected");
763
+
+
764
+
+ return result;
765
+
+}
766
+
+
767
+
+void pulse_funcs (struct hw_funcs *funcs)
768
+
+{
769
+
+ funcs->init = pulse_init;
770
+
+ funcs->shutdown = pulse_shutdown;
771
+
+ funcs->open = pulse_open;
772
+
+ funcs->close = pulse_close;
773
+
+ funcs->play = pulse_play;
774
+
+ funcs->read_mixer = pulse_read_mixer;
775
+
+ funcs->set_mixer = pulse_set_mixer;
776
+
+ funcs->get_buff_fill = pulse_get_buff_fill;
777
+
+ funcs->reset = pulse_reset;
778
+
+ funcs->get_rate = pulse_get_rate;
779
+
+ funcs->toggle_mixer_channel = pulse_toggle_mixer_channel;
780
+
+ funcs->get_mixer_channel_name = pulse_get_mixer_channel_name;
781
+
+}
782
+
diff --git a/pulse.h b/pulse.h
783
+
new file mode 100644
784
+
--- /dev/null
785
+
+++ b/pulse.h
786
+
@@ -0,0 +1,14 @@
787
+
+#ifndef PULSE_H
788
+
+#define PULSE_H
789
+
+
790
+
+#ifdef __cplusplus
791
+
+extern "C" {
792
+
+#endif
793
+
+
794
+
+void pulse_funcs (struct hw_funcs *funcs);
795
+
+
796
+
+#ifdef __cplusplus
797
+
+}
798
+
+#endif
799
+
+
800
+
+#endif