2 * This file is part of DisOrder
3 * Copyright (C) 2013 Mark Wooding
5 * This program is free software: you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation, either version 3 of the License, or
8 * (at your option) any later version.
10 * This program is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 * GNU General Public License for more details.
15 * You should have received a copy of the GNU General Public License
16 * along with this program. If not, see <http://www.gnu.org/licenses/>.
18 /** @file server/gstdecode.c
19 * @brief Decode compressed audio files, and apply ReplayGain.
22 #include "disorder-server.h"
24 #include "speaker-protocol.h"
26 /* Ugh. It turns out that libxml tries to define a function called
27 * `attribute', and it's included by GStreamer for some unimaginable reason.
28 * So undefine it here. We'll want GCC attributes for special effects, but
29 * can take care of ourselves.
35 #include <gst/app/gstappsink.h>
36 #include <gst/audio/audio.h>
38 /* The only application we have for `attribute' is declaring function
39 * arguments as being unused, because we have a lot of callback functions
40 * which are meant to comply with an externally defined interface.
43 # define UNUSED __attribute__((unused))
46 #define END ((void *)0)
47 #define N(v) (sizeof(v)/sizeof(*(v)))
50 static const char *file
;
51 static GstAppSink
*appsink
;
52 static GstElement
*pipeline
;
53 static GMainLoop
*loop
;
54 static unsigned flags
= 0;
57 #define MODES(_) _("off", OFF) _("track", TRACK) _("album", ALBUM)
59 #define DEFENUM(name, tag) tag,
64 static const char *const modes
[] = {
65 #define DEFNAME(name, tag) name,
71 static const char *const dithers
[] = {
72 "none", "rpdf", "tpdf", "tpdf-hf", 0
75 static const char *const shapes
[] = {
76 "none", "error-feedback", "simple", "medium", "high", 0
79 static int dither
= -1;
80 static int mode
= ALBUM
;
81 static int quality
= -1;
82 static int shape
= -1;
83 static gdouble fallback
= 0.0;
85 static struct stream_header hdr
;
87 /* Report the pads of an element ELT, as iterated by IT; WHAT is an adjective
88 * phrase describing the pads for use in the output.
90 static void report_element_pads(const char *what
, GstElement
*elt
,
97 switch(gst_iterator_next(it
, &pad
)) {
98 case GST_ITERATOR_DONE
:
100 case GST_ITERATOR_OK
:
101 cs
= gst_caps_to_string(gst_pad_get_caps(pad
));
102 disorder_error(0, " `%s' %s pad: %s", GST_OBJECT_NAME(elt
), what
, cs
);
106 case GST_ITERATOR_RESYNC
:
107 gst_iterator_resync(it
);
109 case GST_ITERATOR_ERROR
:
110 disorder_error(0, "<failed to enumerate `%s' %s pads>",
111 GST_OBJECT_NAME(elt
), what
);
117 gst_iterator_free(it
);
120 /* Link together two elements; fail with an approximately useful error
121 * message if it didn't work.
123 static void link_elements(GstElement
*left
, GstElement
*right
)
125 /* Try to link things together. */
126 if(gst_element_link(left
, right
)) return;
128 /* If this didn't work, it's probably for some really hairy reason, so
129 * provide a bunch of debugging information.
131 disorder_error(0, "failed to link GStreamer elements `%s' and `%s'",
132 GST_OBJECT_NAME(left
), GST_OBJECT_NAME(right
));
133 report_element_pads("source", left
, gst_element_iterate_src_pads(left
));
134 report_element_pads("source", right
, gst_element_iterate_sink_pads(right
));
135 disorder_fatal(0, "can't decode `%s'", file
);
138 /* The `decoderbin' element (DECODE) has deigned to announce a new PAD.
139 * Maybe we should attach the tag end of our pipeline (starting with the
142 static void decoder_pad_arrived(GstElement
*decode
, GstPad
*pad
, gpointer u
)
144 GstElement
*tail
= u
;
145 GstCaps
*caps
= gst_pad_get_caps(pad
);
150 /* The input file could be more or less anything, so this could be any kind
151 * of pad. We're only interested if it's audio, so let's go check.
153 for(i
= 0, n
= gst_caps_get_size(caps
); i
< n
; i
++) {
154 s
= gst_caps_get_structure(caps
, i
);
155 name
= gst_structure_get_name(s
);
156 if(strncmp(name
, "audio/x-raw-", 12) == 0) goto match
;
161 /* Yes, it's audio. Link the two elements together. */
162 link_elements(decode
, tail
);
164 /* If requested using the environemnt variable `GST_DEBUG_DUMP_DOT_DIR',
165 * write a dump of the now-completed pipeline.
167 GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(pipeline
),
168 GST_DEBUG_GRAPH_SHOW_ALL
,
169 "disorder-gstdecode");
172 /* Prepare the GStreamer pipeline, ready to decode the given FILE. This sets
173 * up the variables `appsink' and `pipeline'.
175 static void prepare_pipeline(void)
177 GstElement
*source
= gst_element_factory_make("filesrc", "file");
178 GstElement
*decode
= gst_element_factory_make("decodebin", "decode");
179 GstElement
*resample
= gst_element_factory_make("audioresample",
181 GstElement
*convert
= gst_element_factory_make("audioconvert", "convert");
182 GstElement
*sink
= gst_element_factory_make("appsink", "sink");
183 GstElement
*tail
= sink
;
186 const struct stream_header
*fmt
= &config
->sample_format
;
188 /* Set up the global variables. */
189 pipeline
= gst_pipeline_new("pipe");
190 appsink
= GST_APP_SINK(sink
);
192 /* Configure the various simple elements. */
193 g_object_set(source
, "location", file
, END
);
194 g_object_set(sink
, "sync", FALSE
, END
);
196 /* Configure the resampler and converter. Leave things as their defaults
197 * if the user hasn't made an explicit request.
199 if(quality
>= 0) g_object_set(resample
, "quality", quality
, END
);
200 if(dither
>= 0) g_object_set(convert
, "dithering", dither
, END
);
201 if(shape
>= 0) g_object_set(convert
, "noise-shaping", shape
, END
);
203 /* Set up the sink's capabilities from the configuration. */
204 caps
= gst_caps_new_simple("audio/x-raw-int",
205 "width", G_TYPE_INT
, fmt
->bits
,
206 "depth", G_TYPE_INT
, fmt
->bits
,
207 "channels", G_TYPE_INT
, fmt
->channels
,
208 "signed", G_TYPE_BOOLEAN
, TRUE
,
209 "rate", G_TYPE_INT
, fmt
->rate
,
210 "endianness", G_TYPE_INT
,
211 fmt
->endian
== ENDIAN_BIG ?
212 G_BIG_ENDIAN
: G_LITTLE_ENDIAN
,
214 gst_app_sink_set_caps(appsink
, caps
);
216 /* Add the various elements into the pipeline. We'll stitch them together
217 * in pieces, because the pipeline is somewhat dynamic.
219 gst_bin_add_many(GST_BIN(pipeline
),
221 resample
, convert
, sink
, END
);
223 /* Link audio conversion stages onto the front. The rest of DisOrder
224 * doesn't handle much of the full panoply of exciting audio formats.
226 link_elements(convert
, tail
); tail
= convert
;
227 link_elements(resample
, tail
); tail
= resample
;
229 /* If we're meant to do ReplayGain then insert it into the pipeline before
233 gain
= gst_element_factory_make("rgvolume", "gain");
235 "album-mode", mode
== ALBUM
,
236 "fallback-gain", fallback
,
238 gst_bin_add(GST_BIN(pipeline
), gain
);
239 link_elements(gain
, tail
); tail
= gain
;
242 /* Link the source and the decoder together. The `decodebin' is annoying
243 * and doesn't have any source pads yet, so the best we can do is make two
244 * halves of the chain, and add a hook to stitch them together later.
246 link_elements(source
, decode
);
247 g_signal_connect(decode
, "pad-added",
248 G_CALLBACK(decoder_pad_arrived
), tail
);
251 /* Respond to a message from the BUS. The only thing we need worry about
252 * here is errors from the pipeline.
254 static void bus_message(GstBus UNUSED
*bus
, GstMessage
*msg
,
258 case GST_MESSAGE_ERROR
:
259 disorder_fatal(0, "%s",
260 gst_structure_get_string(msg
->structure
, "debug"));
266 /* End of stream. Stop polling the main loop. */
267 static void cb_eos(GstAppSink UNUSED
*sink
, gpointer UNUSED u
)
268 { g_main_loop_quit(loop
); }
270 /* Preroll buffers are prepared when the pipeline moves to the `paused'
271 * state, so that they're ready for immediate playback. Conveniently, they
272 * also carry format information, which is what we want here. Stash the
273 * sample format information in the `stream_header' structure ready for
274 * actual buffers of interesting data.
276 static GstFlowReturn
cb_preroll(GstAppSink
*sink
, gpointer UNUSED u
)
278 GstBuffer
*buf
= gst_app_sink_pull_preroll(sink
);
279 GstCaps
*caps
= GST_BUFFER_CAPS(buf
);
281 #ifdef HAVE_GST_AUDIO_INFO_FROM_CAPS
283 /* Parse the audio format information out of the caps. There's a handy
284 * function to do this in later versions of gst-plugins-base, so use that
285 * if it's available. Once we no longer care about supporting such old
286 * versions we can delete the version which does the job the hard way.
291 if(!gst_audio_info_from_caps(&ai
, caps
))
292 disorder_fatal(0, "can't decode `%s': failed to parse audio info", file
);
294 hdr
.channels
= ai
.channels
;
295 hdr
.bits
= ai
.finfo
->width
;
296 hdr
.endian
= ai
.finfo
->endianness
== G_BIG_ENDIAN ?
297 ENDIAN_BIG
: ENDIAN_LITTLE
;
303 gint rate
, channels
, bits
, endian
;
306 /* Make sure that the caps is basically the right shape. */
307 if(!GST_CAPS_IS_SIMPLE(caps
)) disorder_fatal(0, "expected simple caps");
308 s
= gst_caps_get_structure(caps
, 0);
309 ty
= gst_structure_get_name(s
);
310 if(strcmp(ty
, "audio/x-raw-int") != 0)
311 disorder_fatal(0, "unexpected content type `%s'", ty
);
313 /* Extract fields from the structure. */
314 if(!gst_structure_get(s
,
315 "rate", G_TYPE_INT
, &rate
,
316 "channels", G_TYPE_INT
, &channels
,
317 "width", G_TYPE_INT
, &bits
,
318 "endianness", G_TYPE_INT
, &endian
,
319 "signed", G_TYPE_BOOLEAN
, &signedp
,
321 disorder_fatal(0, "can't decode `%s': failed to parse audio caps", file
);
322 hdr
.rate
= rate
; hdr
.channels
= channels
; hdr
.bits
= bits
;
323 hdr
.endian
= endian
== G_BIG_ENDIAN ? ENDIAN_BIG
: ENDIAN_LITTLE
;
327 gst_buffer_unref(buf
);
331 /* A new buffer of sample data has arrived, so we should pass it on with
332 * appropriate framing.
334 static GstFlowReturn
cb_buffer(GstAppSink
*sink
, gpointer UNUSED u
)
336 GstBuffer
*buf
= gst_app_sink_pull_buffer(sink
);
338 /* Make sure we actually have a grip on the sample format here. */
339 if(!hdr
.rate
) disorder_fatal(0, "format unset");
341 /* Write out a frame of audio data. */
342 hdr
.nbytes
= GST_BUFFER_SIZE(buf
);
343 if((!(flags
&f_stream
) && fwrite(&hdr
, sizeof(hdr
), 1, fp
) != 1) ||
344 fwrite(GST_BUFFER_DATA(buf
), 1, hdr
.nbytes
, fp
) != hdr
.nbytes
)
345 disorder_fatal(errno
, "output");
347 /* And we're done. */
348 gst_buffer_unref(buf
);
352 static GstAppSinkCallbacks callbacks
= {
354 .new_preroll
= cb_preroll
,
355 .new_buffer
= cb_buffer
358 /* Decode the audio file. We're already set up for everything. */
359 static void decode(void)
361 GstBus
*bus
= gst_pipeline_get_bus(GST_PIPELINE(pipeline
));
363 /* Set up the message bus and main loop. */
364 gst_bus_add_signal_watch(bus
);
365 loop
= g_main_loop_new(0, FALSE
);
366 g_signal_connect(bus
, "message", G_CALLBACK(bus_message
), 0);
368 /* Tell the sink to call us when interesting things happen. */
369 gst_app_sink_set_callbacks(appsink
, &callbacks
, 0, 0);
371 /* Set the ball rolling. */
372 gst_element_set_state(GST_ELEMENT(pipeline
), GST_STATE_PLAYING
);
374 /* And wait for the miracle to come. */
375 g_main_loop_run(loop
);
377 /* Shut down the pipeline. This isn't strictly necessary, since we're
378 * about to exit very soon, but it's kind of polite.
380 gst_element_set_state(GST_ELEMENT(pipeline
), GST_STATE_NULL
);
383 static int getenum(const char *what
, const char *s
, const char *const *tags
)
387 for(i
= 0; tags
[i
]; i
++)
388 if(strcmp(s
, tags
[i
]) == 0) return i
;
389 disorder_fatal(0, "unknown %s `%s'", what
, s
);
392 static double getfloat(const char *what
, const char *s
)
399 if(*q
|| errno
) disorder_fatal(0, "invalid %s `%s'", what
, s
);
403 static int getint(const char *what
, const char *s
, int min
, int max
)
409 i
= strtol(s
, &q
, 10);
410 if(*q
|| errno
|| min
> i
|| i
> max
)
411 disorder_fatal(0, "invalid %s `%s'", what
, s
);
415 static const struct option options
[] = {
416 { "help", no_argument
, 0, 'h' },
417 { "version", no_argument
, 0, 'V' },
418 { "config", required_argument
, 0, 'c' },
419 { "dither", required_argument
, 0, 'd' },
420 { "fallback-gain", required_argument
, 0, 'f' },
421 { "noise-shape", required_argument
, 0, 'n' },
422 { "quality", required_argument
, 0, 'q' },
423 { "replay-gain", required_argument
, 0, 'r' },
424 { "stream", no_argument
, 0, 's' },
428 static void help(void)
431 " disorder-gstdecode [OPTIONS] PATH\n"
433 " --help, -h Display usage message\n"
434 " --version, -V Display version number\n"
435 " --config PATH, -c PATH Set configuration file\n"
436 " --dither TYPE, -d TYPE TYPE is `none', `rpdf', `tpdf', or "
438 " --fallback-gain DB, -f DB For tracks without ReplayGain data\n"
439 " --noise-shape TYPE, -n TYPE TYPE is `none', `error-feedback',\n"
440 " `simple', `medium' or `high'\n"
441 " --quality QUAL, -q QUAL Resampling quality: 0 poor, 10 good\n"
442 " --replay-gain MODE, -r MODE MODE is `off', `track' or `album'\n"
443 " --stream, -s Output raw samples, without framing\n"
445 "Alternative audio decoder for DisOrder. Only intended to be\n"
446 "used by speaker process, not for normal users.\n");
452 int main(int argc
, char *argv
[])
459 if(!setlocale(LC_CTYPE
, "")) disorder_fatal(errno
, "calling setlocale");
461 /* Parse command line. */
462 while((n
= getopt_long(argc
, argv
, "hVc:d:f:n:q:r:s", options
, 0)) >= 0) {
465 case 'V': version("disorder-gstdecode");
466 case 'c': configfile
= optarg
; break;
467 case 'd': dither
= getenum("dither type", optarg
, dithers
); break;
468 case 'f': fallback
= getfloat("fallback gain", optarg
); break;
469 case 'n': shape
= getenum("noise-shaping type", optarg
, shapes
); break;
470 case 'q': quality
= getint("resample quality", optarg
, 0, 10); break;
471 case 'r': mode
= getenum("ReplayGain mode", optarg
, modes
); break;
472 case 's': flags
|= f_stream
; break;
473 default: disorder_fatal(0, "invalid option");
476 if(optind
>= argc
) disorder_fatal(0, "missing filename");
477 file
= argv
[optind
++];
478 if(optind
< argc
) disorder_fatal(0, "excess arguments");
479 if(config_read(1, 0)) disorder_fatal(0, "cannot read configuration");
481 /* Set up the GStreamer machinery. */
485 /* Set up the output file. */
486 if((e
= getenv("DISORDER_RAW_FD")) != 0) {
487 if((fp
= fdopen(atoi(e
), "wb")) == 0) disorder_fatal(errno
, "fdopen");
494 /* And now we're done. */