summaryrefslogtreecommitdiff
path: root/src/audio.c
blob: 1c537dd7901e3d0c87e086bb34ddbb9cba65d84c (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
#include "audio.h"
#include "varint.h"
#include "cmumble.h"
#include <string.h>

#define SAMPLERATE 48000
#define CHANNELS 1

void
cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
		   const guint8 *data, gsize size)
{
	GstBuffer *gstbuf;

	gstbuf = gst_app_buffer_new(g_memdup(data, size), size, g_free, NULL);
	gst_app_src_push_buffer(user->src, gstbuf);
}

static GstFlowReturn
pull_buffer(GstAppSink *sink, gpointer user_data)
{
	struct cmumble *cm = user_data;
	GstBuffer *buf;
	uint8_t data[1024];
	uint32_t write = 0, pos = 0;
	MumbleProto__UDPTunnel tunnel;
	static int seq = 0;

	/* FIXME: Make this more generic/disable pulling
	 * the pipeline completely if not connected?
	 */
	if (cm->con.conn == NULL)
		return GST_FLOW_OK;

	buf = gst_app_sink_pull_buffer(cm->audio.sink);

	if (++seq <= 2) {
		gst_buffer_unref(buf);
		return GST_FLOW_OK;
	}
	if (GST_BUFFER_SIZE(buf) > 127) {
		g_printerr("GOT TOO BIG BUFFER\n");
		return GST_FLOW_ERROR;
	}

	data[pos++] = (udp_voice_celt_alpha << 5) | (udp_normal_talking);

	encode_varint(&data[pos], &write, ++cm->sequence, sizeof(data)-pos);
	pos += write;

	data[pos++] = 0x00 /*: 0x80 */ | (GST_BUFFER_SIZE(buf) & 0x7F);
	memcpy(&data[pos], GST_BUFFER_DATA(buf), GST_BUFFER_SIZE(buf));
	pos += GST_BUFFER_SIZE(buf);

	gst_buffer_unref(buf);

	mumble_proto__udptunnel__init(&tunnel);
	tunnel.packet.data = data;
	tunnel.packet.len = pos;
	cmumble_send_msg(cm, &tunnel.base);

	return GST_FLOW_OK;
}

static int
setup_recording_gst_pipeline(struct cmumble *cm)
{
	GstElement *pipeline, *cutter, *sink;
	GError *error = NULL;
	GstCaps *caps;

	char *desc = "autoaudiosrc ! cutter name=cutter ! audioresample ! audioconvert ! "
		"audio/x-raw-int,channels=1,depth=16,rate=48000,signed=TRUE,width=16 ! "
		"celtenc ! appsink name=sink";

	pipeline = gst_parse_launch(desc, &error);
	if (error) {
		g_printerr("Failed to create pipeline: %s\n", error->message);
		return -1;
	}
	sink = gst_bin_get_by_name(GST_BIN(pipeline), "sink");
	cm->audio.sink = GST_APP_SINK(sink);
	cm->audio.record_pipeline = pipeline;

	cutter = gst_bin_get_by_name(GST_BIN(pipeline), "cutter");
	g_object_set(G_OBJECT(cutter),
		     "threshold_dB", -45.0, "leaky", TRUE, NULL);

	gst_app_sink_set_emit_signals(cm->audio.sink, TRUE);
	gst_app_sink_set_drop(cm->audio.sink, FALSE);;
	g_signal_connect(sink, "new-buffer", G_CALLBACK(pull_buffer), cm);

	caps = gst_caps_new_simple("audio/x-celt",
				   "rate", G_TYPE_INT, SAMPLERATE,
				   "channels", G_TYPE_INT, 1,
				   "frame-size", G_TYPE_INT, SAMPLERATE/100,
				   NULL);
	gst_app_sink_set_caps(cm->audio.sink, caps);
	gst_caps_unref(caps);

	gst_element_set_state(pipeline, GST_STATE_PLAYING);

	cm->sequence = 0;

	return 0;
}

static void
set_pulse_states(gpointer data, gpointer user_data)
{
	GstElement *elm = data;
	struct cmumble_user *user = user_data;
	GstStructure *props;
	gchar *name;

	if (g_strcmp0(G_OBJECT_TYPE_NAME(elm), "GstPulseSink") != 0 ||
	    g_object_class_find_property(G_OBJECT_GET_CLASS(elm),
					 "stream-properties") == NULL)
		goto out;

	/* configure pulseaudio to use:
	 * load-module module-device-manager "do_routing=1"
	 * or new users may join to default output which is not headset?
	 * Also consider setting device.intended_roles = "phone" for your
	 * wanted default output (if you dont have a usb headset dev). */

	name = g_strdup_printf("cmumble [%s]", user->name);

	props = gst_structure_new("props",
				  "application.name", G_TYPE_STRING, name,
				  "media.role", G_TYPE_STRING, "phone",
				  NULL);

	g_object_set(elm, "stream-properties", props, NULL);
	gst_structure_free(props);
	g_free(name);

out:
	g_object_unref(G_OBJECT(elm));
}

int
cmumble_audio_create_playback_pipeline(struct cmumble *cm,
				       struct cmumble_user *user)
{
	GstElement *pipeline, *sink_bin;
	GError *error = NULL;
	char *desc = "appsrc name=src ! celtdec ! audioconvert ! autoaudiosink name=sink";

	pipeline = gst_parse_launch(desc, &error);
	if (error) {
		g_printerr("Failed to create pipeline: %s\n", error->message);
		return -1;
	}

	user->pipeline = pipeline;
	user->src = GST_APP_SRC(gst_bin_get_by_name(GST_BIN(pipeline), "src"));

	/* Important! */
	gst_base_src_set_live(GST_BASE_SRC(user->src), TRUE); 
	gst_base_src_set_do_timestamp(GST_BASE_SRC(user->src), TRUE);
	gst_base_src_set_format(GST_BASE_SRC(user->src), GST_FORMAT_TIME);

	gst_app_src_set_stream_type(user->src, GST_APP_STREAM_TYPE_STREAM); 

	gst_element_set_state(pipeline, GST_STATE_PLAYING);

	sink_bin = gst_bin_get_by_name(GST_BIN(pipeline), "sink");
	GstIterator *iter = gst_bin_iterate_sinks(GST_BIN(sink_bin));
	gst_iterator_foreach(iter, set_pulse_states, user);
	gst_iterator_free(iter);

	/* Setup Celt Decoder */
	cmumble_audio_push(cm, user,
			   cm->audio.celt_header_packet, sizeof(CELTHeader));
	/* fake vorbiscomment buffer */
	cmumble_audio_push(cm, user, NULL, 0);

	return 0;
}

static int
setup_playback_gst_pipeline(struct cmumble *cm)
{
	cm->audio.celt_mode = celt_mode_create(SAMPLERATE,
						SAMPLERATE / 100, NULL);
	celt_header_init(&cm->audio.celt_header, cm->audio.celt_mode, CHANNELS);
	celt_header_to_packet(&cm->audio.celt_header,
			      cm->audio.celt_header_packet, sizeof(CELTHeader));

	return 0;
}

int
cmumble_audio_init(struct cmumble *cm)
{
	if (setup_playback_gst_pipeline(cm) < 0)
		return -1;

	if (setup_recording_gst_pipeline(cm) < 0)
		return -1;

	return 0;
}

int
cmumble_audio_fini(struct cmumble *cm)
{

	return 0;
}