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_max_buffers(appsink, 16);
370 gst_app_sink_set_drop(appsink, FALSE);
371 gst_app_sink_set_callbacks(appsink, &callbacks, 0, 0);
373 /* Set the ball rolling. */
374 gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING);
376 /* And wait for the miracle to come. */
377 g_main_loop_run(loop);
379 /* Shut down the pipeline. This isn't strictly necessary, since we're
380 * about to exit very soon, but it's kind of polite.
382 gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_NULL);
385 static int getenum(const char *what, const char *s, const char *const *tags)
389 for(i = 0; tags[i]; i++)
390 if(strcmp(s, tags[i]) == 0) return i;
391 disorder_fatal(0, "unknown %s `%s'", what, s);
394 static double getfloat(const char *what, const char *s)
401 if(*q || errno) disorder_fatal(0, "invalid %s `%s'", what, s);
405 static int getint(const char *what, const char *s, int min, int max)
411 i = strtol(s, &q, 10);
412 if(*q || errno || min > i || i > max)
413 disorder_fatal(0, "invalid %s `%s'", what, s);
417 static const struct option options[] = {
418 { "help", no_argument, 0, 'h' },
419 { "version", no_argument, 0, 'V' },
420 { "config", required_argument, 0, 'c' },
421 { "dither", required_argument, 0, 'd' },
422 { "fallback-gain", required_argument, 0, 'f' },
423 { "noise-shape", required_argument, 0, 'n' },
424 { "quality", required_argument, 0, 'q' },
425 { "replay-gain", required_argument, 0, 'r' },
426 { "stream", no_argument, 0, 's' },
430 static void help(void)
433 " disorder-gstdecode [OPTIONS] PATH\n"
435 " --help, -h Display usage message\n"
436 " --version, -V Display version number\n"
437 " --config PATH, -c PATH Set configuration file\n"
438 " --dither TYPE, -d TYPE TYPE is `none', `rpdf', `tpdf', or "
440 " --fallback-gain DB, -f DB For tracks without ReplayGain data\n"
441 " --noise-shape TYPE, -n TYPE TYPE is `none', `error-feedback',\n"
442 " `simple', `medium' or `high'\n"
443 " --quality QUAL, -q QUAL Resampling quality: 0 poor, 10 good\n"
444 " --replay-gain MODE, -r MODE MODE is `off', `track' or `album'\n"
445 " --stream, -s Output raw samples, without framing\n"
447 "Alternative audio decoder for DisOrder. Only intended to be\n"
448 "used by speaker process, not for normal users.\n");
454 int main(int argc, char *argv[])
461 if(!setlocale(LC_CTYPE, "")) disorder_fatal(errno, "calling setlocale");
463 /* Parse command line. */
464 while((n = getopt_long(argc, argv, "hVc:d:f:n:q:r:s", options, 0)) >= 0) {
467 case 'V': version("disorder-gstdecode");
468 case 'c': configfile = optarg; break;
469 case 'd': dither = getenum("dither type", optarg, dithers); break;
470 case 'f': fallback = getfloat("fallback gain", optarg); break;
471 case 'n': shape = getenum("noise-shaping type", optarg, shapes); break;
472 case 'q': quality = getint("resample quality", optarg, 0, 10); break;
473 case 'r': mode = getenum("ReplayGain mode", optarg, modes); break;
474 case 's': flags |= f_stream; break;
475 default: disorder_fatal(0, "invalid option");
478 if(optind >= argc) disorder_fatal(0, "missing filename");
479 file = argv[optind++];
480 if(optind < argc) disorder_fatal(0, "excess arguments");
481 if(config_read(1, 0)) disorder_fatal(0, "cannot read configuration");
483 /* Set up the GStreamer machinery. */
487 /* Set up the output file. */
488 if((e = getenv("DISORDER_RAW_FD")) != 0) {
489 if((fp = fdopen(atoi(e), "wb")) == 0) disorder_fatal(errno, "fdopen");
496 /* And now we're done. */