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;
55 #define MODES(_) _("off", OFF) _("track", TRACK) _("album", ALBUM)
57 #define DEFENUM(name, tag) tag,
62 static const char *const modes[] = {
63 #define DEFNAME(name, tag) name,
69 static const char *const dithers[] = {
70 "none", "rpdf", "tpdf", "tpdf-hf", 0
73 static const char *const shapes[] = {
74 "none", "error-feedback", "simple", "medium", "high", 0
77 static int dither = -1;
78 static int mode = ALBUM;
79 static int quality = -1;
80 static int shape = -1;
81 static gdouble fallback = 0.0;
83 static struct stream_header hdr;
85 /* Report the pads of an element ELT, as iterated by IT; WHAT is an adjective
86 * phrase describing the pads for use in the output.
88 static void report_element_pads(const char *what, GstElement *elt,
95 switch(gst_iterator_next(it, &pad)) {
96 case GST_ITERATOR_DONE:
99 cs = gst_caps_to_string(gst_pad_get_caps(pad));
100 disorder_error(0, " `%s' %s pad: %s", GST_OBJECT_NAME(elt), what, cs);
104 case GST_ITERATOR_RESYNC:
105 gst_iterator_resync(it);
107 case GST_ITERATOR_ERROR:
108 disorder_error(0, "<failed to enumerate `%s' %s pads>",
109 GST_OBJECT_NAME(elt), what);
115 gst_iterator_free(it);
118 /* Link together two elements; fail with an approximately useful error
119 * message if it didn't work.
121 static void link_elements(GstElement *left, GstElement *right)
123 /* Try to link things together. */
124 if(gst_element_link(left, right)) return;
126 /* If this didn't work, it's probably for some really hairy reason, so
127 * provide a bunch of debugging information.
129 disorder_error(0, "failed to link GStreamer elements `%s' and `%s'",
130 GST_OBJECT_NAME(left), GST_OBJECT_NAME(right));
131 report_element_pads("source", left, gst_element_iterate_src_pads(left));
132 report_element_pads("source", right, gst_element_iterate_sink_pads(right));
133 disorder_fatal(0, "can't decode `%s'", file);
136 /* The `decoderbin' element (DECODE) has deigned to announce a new PAD.
137 * Maybe we should attach the tag end of our pipeline (starting with the
140 static void decoder_pad_arrived(GstElement *decode, GstPad *pad, gpointer u)
142 GstElement *tail = u;
143 GstCaps *caps = gst_pad_get_caps(pad);
148 /* The input file could be more or less anything, so this could be any kind
149 * of pad. We're only interested if it's audio, so let's go check.
151 for(i = 0, n = gst_caps_get_size(caps); i < n; i++) {
152 s = gst_caps_get_structure(caps, i);
153 name = gst_structure_get_name(s);
154 if(strncmp(name, "audio/x-raw-", 12) == 0) goto match;
159 /* Yes, it's audio. Link the two elements together. */
160 link_elements(decode, tail);
162 /* If requested using the environemnt variable `GST_DEBUG_DUMP_DOT_DIR',
163 * write a dump of the now-completed pipeline.
165 GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(pipeline),
166 GST_DEBUG_GRAPH_SHOW_ALL,
167 "disorder-gstdecode");
170 /* Prepare the GStreamer pipeline, ready to decode the given FILE. This sets
171 * up the variables `appsink' and `pipeline'.
173 static void prepare_pipeline(void)
175 GstElement *source = gst_element_factory_make("filesrc", "file");
176 GstElement *decode = gst_element_factory_make("decodebin", "decode");
177 GstElement *resample = gst_element_factory_make("audioresample",
179 GstElement *convert = gst_element_factory_make("audioconvert", "convert");
180 GstElement *sink = gst_element_factory_make("appsink", "sink");
181 GstElement *tail = sink;
184 const struct stream_header *fmt = &config->sample_format;
186 /* Set up the global variables. */
187 pipeline = gst_pipeline_new("pipe");
188 appsink = GST_APP_SINK(sink);
190 /* Configure the various simple elements. */
191 g_object_set(source, "location", file, END);
192 g_object_set(sink, "sync", FALSE, END);
194 /* Configure the resampler and converter. Leave things as their defaults
195 * if the user hasn't made an explicit request.
197 if(quality >= 0) g_object_set(resample, "quality", quality, END);
198 if(dither >= 0) g_object_set(convert, "dithering", dither, END);
199 if(shape >= 0) g_object_set(convert, "noise-shaping", shape, END);
201 /* Set up the sink's capabilities from the configuration. */
202 caps = gst_caps_new_simple("audio/x-raw-int",
203 "width", G_TYPE_INT, fmt->bits,
204 "depth", G_TYPE_INT, fmt->bits,
205 "channels", G_TYPE_INT, fmt->channels,
206 "signed", G_TYPE_BOOLEAN, TRUE,
207 "rate", G_TYPE_INT, fmt->rate,
208 "endianness", G_TYPE_INT,
209 fmt->endian == ENDIAN_BIG ?
210 G_BIG_ENDIAN : G_LITTLE_ENDIAN,
212 gst_app_sink_set_caps(appsink, caps);
214 /* Add the various elements into the pipeline. We'll stitch them together
215 * in pieces, because the pipeline is somewhat dynamic.
217 gst_bin_add_many(GST_BIN(pipeline),
219 resample, convert, sink, END);
221 /* Link audio conversion stages onto the front. The rest of DisOrder
222 * doesn't handle much of the full panoply of exciting audio formats.
224 link_elements(convert, tail); tail = convert;
225 link_elements(resample, tail); tail = resample;
227 /* If we're meant to do ReplayGain then insert it into the pipeline before
231 gain = gst_element_factory_make("rgvolume", "gain");
233 "album-mode", mode == ALBUM,
234 "fallback-gain", fallback,
236 gst_bin_add(GST_BIN(pipeline), gain);
237 link_elements(gain, tail); tail = gain;
240 /* Link the source and the decoder together. The `decodebin' is annoying
241 * and doesn't have any source pads yet, so the best we can do is make two
242 * halves of the chain, and add a hook to stitch them together later.
244 link_elements(source, decode);
245 g_signal_connect(decode, "pad-added",
246 G_CALLBACK(decoder_pad_arrived), tail);
249 /* Respond to a message from the BUS. The only thing we need worry about
250 * here is errors from the pipeline.
252 static void bus_message(GstBus UNUSED *bus, GstMessage *msg,
256 case GST_MESSAGE_ERROR:
257 disorder_fatal(0, "%s",
258 gst_structure_get_string(msg->structure, "debug"));
264 /* End of stream. Stop polling the main loop. */
265 static void cb_eos(GstAppSink UNUSED *sink, gpointer UNUSED u)
266 { g_main_loop_quit(loop); }
268 /* Preroll buffers are prepared when the pipeline moves to the `paused'
269 * state, so that they're ready for immediate playback. Conveniently, they
270 * also carry format information, which is what we want here. Stash the
271 * sample format information in the `stream_header' structure ready for
272 * actual buffers of interesting data.
274 static GstFlowReturn cb_preroll(GstAppSink *sink, gpointer UNUSED u)
276 GstBuffer *buf = gst_app_sink_pull_preroll(sink);
277 GstCaps *caps = GST_BUFFER_CAPS(buf);
279 #ifdef HAVE_GST_AUDIO_INFO_FROM_CAPS
281 /* Parse the audio format information out of the caps. There's a handy
282 * function to do this in later versions of gst-plugins-base, so use that
283 * if it's available. Once we no longer care about supporting such old
284 * versions we can delete the version which does the job the hard way.
289 if(!gst_audio_info_from_caps(&ai, caps))
290 disorder_fatal(0, "can't decode `%s': failed to parse audio info", file);
292 hdr.channels = ai.channels;
293 hdr.bits = ai.finfo->width;
294 hdr.endian = ai.finfo->endianness == G_BIG_ENDIAN ?
295 ENDIAN_BIG : ENDIAN_LITTLE;
301 gint rate, channels, bits, endian;
304 /* Make sure that the caps is basically the right shape. */
305 if(!GST_CAPS_IS_SIMPLE(caps)) disorder_fatal(0, "expected simple caps");
306 s = gst_caps_get_structure(caps, 0);
307 ty = gst_structure_get_name(s);
308 if(strcmp(ty, "audio/x-raw-int") != 0)
309 disorder_fatal(0, "unexpected content type `%s'", ty);
311 /* Extract fields from the structure. */
312 if(!gst_structure_get(s,
313 "rate", G_TYPE_INT, &rate,
314 "channels", G_TYPE_INT, &channels,
315 "width", G_TYPE_INT, &bits,
316 "endianness", G_TYPE_INT, &endian,
317 "signed", G_TYPE_BOOLEAN, &signedp,
319 disorder_fatal(0, "can't decode `%s': failed to parse audio caps", file);
320 hdr.rate = rate; hdr.channels = channels; hdr.bits = bits;
321 hdr.endian = endian == G_BIG_ENDIAN ? ENDIAN_BIG : ENDIAN_LITTLE;
325 gst_buffer_unref(buf);
329 /* A new buffer of sample data has arrived, so we should pass it on with
330 * appropriate framing.
332 static GstFlowReturn cb_buffer(GstAppSink *sink, gpointer UNUSED u)
334 GstBuffer *buf = gst_app_sink_pull_buffer(sink);
336 /* Make sure we actually have a grip on the sample format here. */
337 if(!hdr.rate) disorder_fatal(0, "format unset");
339 /* Write out a frame of audio data. */
340 hdr.nbytes = GST_BUFFER_SIZE(buf);
341 if(fwrite(&hdr, sizeof(hdr), 1, fp) != 1 ||
342 fwrite(GST_BUFFER_DATA(buf), 1, hdr.nbytes, fp) != hdr.nbytes)
343 disorder_fatal(errno, "output");
345 /* And we're done. */
346 gst_buffer_unref(buf);
350 static GstAppSinkCallbacks callbacks = {
352 .new_preroll = cb_preroll,
353 .new_buffer = cb_buffer
356 /* Decode the audio file. We're already set up for everything. */
357 static void decode(void)
359 GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
361 /* Set up the message bus and main loop. */
362 gst_bus_add_signal_watch(bus);
363 loop = g_main_loop_new(0, FALSE);
364 g_signal_connect(bus, "message", G_CALLBACK(bus_message), 0);
366 /* Tell the sink to call us when interesting things happen. */
367 gst_app_sink_set_callbacks(appsink, &callbacks, 0, 0);
369 /* Set the ball rolling. */
370 gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING);
372 /* And wait for the miracle to come. */
373 g_main_loop_run(loop);
375 /* Shut down the pipeline. This isn't strictly necessary, since we're
376 * about to exit very soon, but it's kind of polite.
378 gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_NULL);
381 static int getenum(const char *what, const char *s, const char *const *tags)
385 for(i = 0; tags[i]; i++)
386 if(strcmp(s, tags[i]) == 0) return i;
387 disorder_fatal(0, "unknown %s `%s'", what, s);
390 static double getfloat(const char *what, const char *s)
397 if(*q || errno) disorder_fatal(0, "invalid %s `%s'", what, s);
401 static int getint(const char *what, const char *s, int min, int max)
407 i = strtol(s, &q, 10);
408 if(*q || errno || min > i || i > max)
409 disorder_fatal(0, "invalid %s `%s'", what, s);
413 static const struct option options[] = {
414 { "help", no_argument, 0, 'h' },
415 { "version", no_argument, 0, 'V' },
416 { "config", required_argument, 0, 'c' },
417 { "dither", required_argument, 0, 'd' },
418 { "fallback-gain", required_argument, 0, 'f' },
419 { "noise-shape", required_argument, 0, 'n' },
420 { "quality", required_argument, 0, 'q' },
421 { "replay-gain", required_argument, 0, 'r' },
425 static void help(void)
428 " disorder-gstdecode [OPTIONS] PATH\n"
430 " --help, -h Display usage message\n"
431 " --version, -V Display version number\n"
432 " --config PATH, -c PATH Set configuration file\n"
433 " --dither TYPE, -d TYPE TYPE is `none', `rpdf', `tpdf', or "
435 " --fallback-gain DB, -f DB For tracks without ReplayGain data\n"
436 " --noise-shape TYPE, -n TYPE TYPE is `none', `error-feedback',\n"
437 " `simple', `medium' or `high'\n"
438 " --quality QUAL, -q QUAL Resampling quality: 0 poor, 10 good\n"
439 " --replay-gain MODE, -r MODE MODE is `off', `track' or `album'\n"
441 "Alternative audio decoder for DisOrder. Only intended to be\n"
442 "used by speaker process, not for normal users.\n");
448 int main(int argc, char *argv[])
455 if(!setlocale(LC_CTYPE, "")) disorder_fatal(errno, "calling setlocale");
457 /* Parse command line. */
458 while((n = getopt_long(argc, argv, "hVc:d:f:n:q:r:", options, 0)) >= 0) {
461 case 'V': version("disorder-gstdecode");
462 case 'c': configfile = optarg; break;
463 case 'd': dither = getenum("dither type", optarg, dithers); break;
464 case 'f': fallback = getfloat("fallback gain", optarg); break;
465 case 'n': shape = getenum("noise-shaping type", optarg, shapes); break;
466 case 'q': quality = getint("resample quality", optarg, 0, 10); break;
467 case 'r': mode = getenum("ReplayGain mode", optarg, modes); break;
468 default: disorder_fatal(0, "invalid option");
471 if(optind >= argc) disorder_fatal(0, "missing filename");
472 file = argv[optind++];
473 if(optind < argc) disorder_fatal(0, "excess arguments");
474 if(config_read(1, 0)) disorder_fatal(0, "cannot read configuration");
476 /* Set up the GStreamer machinery. */
480 /* Set up the output file. */
481 if((e = getenv("DISORDER_RAW_FD")) != 0) {
482 if((fp = fdopen(atoi(e), "wb")) == 0) disorder_fatal(errno, "fdopen");
489 /* And now we're done. */