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#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;
mumble_udptunnel_t 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);
cmumble_init_udptunnel(&tunnel);
tunnel.packet.data = data;
tunnel.packet.len = pos;
cmumble_send_udptunnel(cm, &tunnel);
return GST_FLOW_OK;
}
static gboolean
idle(gpointer user_data)
{
struct cmumble *cm = user_data;
GstAppSink *sink;
while ((sink = g_async_queue_try_pop(cm->async_queue)) != NULL)
pull_buffer(sink, cm);
return FALSE;
}
static GstFlowReturn
new_buffer(GstAppSink *sink, gpointer user_data)
{
struct cmumble *cm = user_data;
g_async_queue_push(cm->async_queue, sink);
g_idle_add(idle, cm);
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(new_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";
GstIterator *it;
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);
/* FIXME: Use a recursive name for sink-actual-sink-pluse instead? like:
* gst_bin_get_by_name(GST_BIN(pipeline), "sink-actual-sink-pulse"); */
sink_bin = gst_bin_get_by_name(GST_BIN(pipeline), "sink");
it = gst_bin_iterate_sinks(GST_BIN(sink_bin));
gst_iterator_foreach(it, set_pulse_states, user);
gst_iterator_free(it);
/* 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));
celt_mode_info(cm->audio.celt_mode, CELT_GET_BITSTREAM_VERSION,
&cm->audio.celt_bitstream_version);
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;
}
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