1 files changed, 449 insertions, 0 deletions

diff --git a/modules/gst/gst.c b/modules/gst/gst.c
new file mode 100644
index 0000000..7af74f4
--- /dev/null
+++ b/modules/gst/gst.c
@@ -0,0 +1,449 @@
+/**
+ * @file gst.c  Gstreamer playbin pipeline
+ *
+ * Copyright (C) 2010 Creytiv.com
+ */
+#include <stdlib.h>
+#include <string.h>
+#define __USE_POSIX199309
+#include <time.h>
+#include <pthread.h>
+#include <gst/gst.h>
+#include <re.h>
+#include <rem.h>
+#include <baresip.h>
+#include "gst.h"
+
+
+#define DEBUG_MODULE "gst"
+#define DEBUG_LEVEL 5
+#include <re_dbg.h>
+
+
+/**
+ * Defines the Gstreamer state
+ *
+ * <pre>
+ *                ptime=variable             ptime=20ms
+ *  .-----------. N kHz          .---------. N kHz
+ *  |           | 1-2 channels   |         | 1-2 channels
+ *  | Gstreamer |--------------->|Packetize|-------------> [read handler]
+ *  |           |                |         |
+ *  '-----------'                '---------'
+ *
+ * </pre>
+ */
+struct ausrc_st {
+	struct ausrc *as;           /**< Inheritance             */
+	pthread_t tid;              /**< Thread ID               */
+	bool run;                   /**< Running flag            */
+	ausrc_read_h *rh;           /**< Read handler            */
+	ausrc_error_h *errh;        /**< Error handler           */
+	void *arg;                  /**< Handler argument        */
+	struct ausrc_prm prm;       /**< Read parameters         */
+	struct aubuf *aubuf;        /**< Packet buffer           */
+	uint32_t psize;             /**< Packet size in bytes    */
+
+	/* Gstreamer */
+	char *uri;
+	GstElement *pipeline, *bin, *source, *capsfilt, *sink;
+	GMainLoop *loop;
+};
+
+
+typedef struct _GstFakeSink GstFakeSink;
+static char gst_uri[256] = "http://relay1.slayradio.org:8000/";
+static struct ausrc *ausrc;
+
+
+static void *thread(void *arg)
+{
+	struct ausrc_st *st = arg;
+
+	/* Now set to playing and iterate. */
+	DEBUG_NOTICE("Setting pipeline to PLAYING\n");
+	gst_element_set_state(st->pipeline, GST_STATE_PLAYING);
+
+	while (st->run) {
+		g_main_loop_run(st->loop);
+	}
+
+	return NULL;
+}
+
+
+static gboolean bus_watch_handler(GstBus *bus, GstMessage *msg, gpointer data)
+{
+	struct ausrc_st *st = data;
+	GMainLoop *loop = st->loop;
+	GstTagList *tag_list;
+	gchar *title;
+	GError *err;
+	gchar *d;
+
+	(void)bus;
+
+	switch (GST_MESSAGE_TYPE(msg)) {
+
+	case GST_MESSAGE_EOS:
+		DEBUG_NOTICE("End-of-stream\n");
+
+		/* XXX decrementing repeat count? */
+
+		/* Re-start stream */
+		if (st->run) {
+			gst_element_set_state(st->pipeline, GST_STATE_NULL);
+			gst_element_set_state(st->pipeline, GST_STATE_PLAYING);
+		}
+		else {
+			g_main_loop_quit(loop);
+		}
+		break;
+
+	case GST_MESSAGE_ERROR:
+		gst_message_parse_error(msg, &err, &d);
+
+		DEBUG_WARNING("Error: %d(%m) message=%s\n", err->code,
+			      err->code, err->message);
+		DEBUG_WARNING("Debug: %s\n", d);
+
+		g_free(d);
+
+		/* Call error handler */
+		if (st->errh)
+			st->errh(err->code, err->message, st->arg);
+
+		g_error_free(err);
+
+		st->run = false;
+		g_main_loop_quit(loop);
+		break;
+
+	case GST_MESSAGE_TAG:
+		gst_message_parse_tag(msg, &tag_list);
+
+		if (gst_tag_list_get_string(tag_list, GST_TAG_TITLE, &title)) {
+			DEBUG_NOTICE("Title: %s\n", title);
+			g_free(title);
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	return TRUE;
+}
+
+
+static void format_check(struct ausrc_st *st, GstStructure *s)
+{
+	int rate, channels, width;
+	gboolean sign;
+
+	if (!st || !s)
+		return;
+
+	gst_structure_get_int(s, "rate", &rate);
+	gst_structure_get_int(s, "channels", &channels);
+	gst_structure_get_int(s, "width", &width);
+	gst_structure_get_boolean(s, "signed", &sign);
+
+	if ((int)st->prm.srate != rate) {
+		DEBUG_WARNING("expected %u Hz (got %u Hz)\n", st->prm.srate,
+			      rate);
+	}
+	if (st->prm.ch != channels) {
+		DEBUG_WARNING("expected %d channels (got %d)\n",
+			      st->prm.ch, channels);
+	}
+	if (16 != width) {
+		DEBUG_WARNING("expected 16-bit width (got %d)\n", width);
+	}
+	if (!sign) {
+		DEBUG_WARNING("expected signed 16-bit format\n");
+	}
+}
+
+
+static void play_packet(struct ausrc_st *st)
+{
+	uint8_t buf[st->psize];
+
+	/* timed read from audio-buffer */
+	if (aubuf_get(st->aubuf, st->prm.ptime, buf, sizeof(buf)))
+		return;
+
+	/* call read handler */
+	if (st->rh)
+		st->rh(buf, sizeof(buf), st->arg);
+}
+
+
+/* Expected format: 16-bit signed PCM */
+static void packet_handler(struct ausrc_st *st, GstBuffer *buffer)
+{
+	int err;
+
+	if (!st->run)
+		return;
+
+	/* NOTE: When streaming from files, the buffer will be filled up
+	 *       pretty quickly..
+	 */
+
+	err = aubuf_write(st->aubuf, GST_BUFFER_DATA(buffer),
+			  GST_BUFFER_SIZE(buffer));
+	if (err) {
+		DEBUG_WARNING("aubuf_write: %m\n", err);
+	}
+
+	/* Empty buffer now */
+	while (st->run) {
+		const struct timespec delay = {0, st->prm.ptime*1000000/2};
+
+		play_packet(st);
+
+		if (aubuf_cur_size(st->aubuf) < st->psize)
+			break;
+
+		(void)nanosleep(&delay, NULL);
+	}
+}
+
+
+static void handoff_handler(GstFakeSink *fakesink, GstBuffer *buffer,
+			    GstPad *pad, gpointer user_data)
+{
+	struct ausrc_st *st = user_data;
+
+	(void)fakesink;
+	(void)pad;
+
+	format_check(st, gst_caps_get_structure(GST_BUFFER_CAPS(buffer), 0));
+
+	packet_handler(st, buffer);
+}
+
+
+static void set_caps(struct ausrc_st *st)
+{
+	GstCaps *caps;
+
+	/* Set the capabilities we want */
+	caps = gst_caps_new_simple("audio/x-raw-int",
+				   "rate",     G_TYPE_INT,    st->prm.srate,
+				   "channels", G_TYPE_INT,    st->prm.ch,
+				   "width",    G_TYPE_INT,    16,
+				   "signed",   G_TYPE_BOOLEAN,true,
+				   NULL);
+#if 0
+	gst_dump_caps(caps);
+#endif
+	g_object_set(G_OBJECT(st->capsfilt), "caps", caps, NULL);
+}
+
+
+/**
+ * Set up the Gstreamer pipeline. The playbin element is used to decode
+ * all kinds of different formats. The capsfilter is used to deliver the
+ * audio in a fixed format (X Hz, 1-2 channels, 16 bit signed)
+ *
+ * The pipeline looks like this:
+ *
+ * <pre>
+ *  .--------------.    .------------------------------------------.
+ *  |    playbin   |    |mybin    .------------.   .------------.  |
+ *  |----.    .----|    |-----.   | capsfilter |   |  fakesink  |  |
+ *  |sink|    |src |--->|ghost|   |----.   .---|   |----.   .---|  |    handoff
+ *  |----'    '----|    |pad  |-->|sink|   |src|-->|sink|   |src|--+--> handler
+ *  |              |    |-----'   '------------'   '------------'  |
+ *  '--------------'    '------------------------------------------'
+ * </pre>
+ *
+ * @param st Audio source state
+ *
+ * @return 0 if success, otherwise errorcode
+ */
+static int gst_setup(struct ausrc_st *st)
+{
+	GstBus *bus;
+	GstPad *pad;
+
+	st->loop = g_main_loop_new(NULL, FALSE);
+
+	st->pipeline = gst_pipeline_new("pipeline");
+	if (!st->pipeline) {
+		DEBUG_WARNING("failed to create pipeline element\n");
+		return ENOMEM;
+	}
+
+	/********************* Player BIN **************************/
+
+	st->source = gst_element_factory_make("playbin", "source");
+	if (!st->source) {
+		DEBUG_WARNING("failed to create playbin source element\n");
+		return ENOMEM;
+	}
+
+	/********************* My BIN **************************/
+
+	st->bin = gst_bin_new("mybin");
+
+	st->capsfilt = gst_element_factory_make("capsfilter", NULL);
+	if (!st->capsfilt) {
+		DEBUG_WARNING("failed to create capsfilter element\n");
+		return ENOMEM;
+	}
+
+	set_caps(st);
+
+	st->sink = gst_element_factory_make("fakesink", "sink");
+	if (!st->sink) {
+		DEBUG_WARNING("failed to create sink element\n");
+		return ENOMEM;
+	}
+
+	gst_bin_add_many(GST_BIN(st->bin), st->capsfilt, st->sink, NULL);
+	gst_element_link_many(st->capsfilt, st->sink, NULL);
+
+	/* add ghostpad */
+	pad = gst_element_get_pad(st->capsfilt, "sink");
+	gst_element_add_pad(st->bin, gst_ghost_pad_new("sink", pad));
+	gst_object_unref(GST_OBJECT(pad));
+
+	/* put all elements in a bin */
+	gst_bin_add_many(GST_BIN(st->pipeline), st->source, NULL);
+
+	/* Override audio-sink handoff handler */
+	g_object_set(G_OBJECT(st->sink), "signal-handoffs", TRUE, NULL);
+	g_signal_connect(st->sink, "handoff", G_CALLBACK(handoff_handler), st);
+	g_object_set(G_OBJECT(st->source), "audio-sink", st->bin, NULL);
+
+	/********************* Misc **************************/
+
+	/* Bus watch */
+	bus = gst_pipeline_get_bus(GST_PIPELINE(st->pipeline));
+	gst_bus_add_watch(bus, bus_watch_handler, st);
+	gst_object_unref(bus);
+
+	/* Set URI */
+	g_object_set(G_OBJECT(st->source), "uri", st->uri, NULL);
+
+	return 0;
+}
+
+
+static void gst_destructor(void *arg)
+{
+	struct ausrc_st *st = arg;
+
+	if (st->run) {
+		st->run = false;
+		g_main_loop_quit(st->loop);
+		pthread_join(st->tid, NULL);
+	}
+
+	gst_element_set_state(st->pipeline, GST_STATE_NULL);
+	gst_object_unref(GST_OBJECT(st->pipeline));
+
+	mem_deref(st->uri);
+	mem_deref(st->aubuf);
+
+	mem_deref(st->as);
+}
+
+
+static int gst_alloc(struct ausrc_st **stp, struct ausrc *as,
+		     struct media_ctx **ctx,
+		     struct ausrc_prm *prm, const char *device,
+		     ausrc_read_h *rh, ausrc_error_h *errh, void *arg)
+{
+	struct ausrc_st *st;
+	unsigned sampc;
+	int err;
+
+	(void)ctx;
+
+	if (!device)
+		device = gst_uri;
+
+	if (!prm)
+		return EINVAL;
+
+	prm->fmt = AUFMT_S16LE;
+
+	st = mem_zalloc(sizeof(*st), gst_destructor);
+	if (!st)
+		return ENOMEM;
+
+	st->as   = mem_ref(as);
+	st->rh   = rh;
+	st->errh = errh;
+	st->arg  = arg;
+
+	err = str_dup(&st->uri, device);
+	if (err)
+		goto out;
+
+	st->prm   = *prm;
+
+	sampc = prm->srate * prm->ch * prm->ptime / 1000;
+
+	st->psize = 2 * sampc;
+
+	err = aubuf_alloc(&st->aubuf, st->psize, 0);
+	if (err)
+		goto out;
+
+	err = gst_setup(st);
+	if (err)
+		goto out;
+
+	st->run  = true;
+	err = pthread_create(&st->tid, NULL, thread, st);
+	if (err) {
+		st->run = false;
+		goto out;
+	}
+
+ out:
+	if (err)
+		mem_deref(st);
+	else
+		*stp = st;
+
+	return err;
+}
+
+
+static int mod_gst_init(void)
+{
+	gchar *s;
+
+	gst_init(0, NULL);
+
+	s = gst_version_string();
+
+	DEBUG_NOTICE("init: %s\n", s);
+
+	g_free(s);
+
+	return ausrc_register(&ausrc, "gst", gst_alloc);
+}
+
+
+static int mod_gst_close(void)
+{
+	gst_deinit();
+	ausrc = mem_deref(ausrc);
+	return 0;
+}
+
+
+EXPORT_SYM const struct mod_export DECL_EXPORTS(gst) = {
+	"gst",
+	"sound",
+	mod_gst_init,
+	mod_gst_close
+};