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Diffstat (limited to 'src/cmumble.c')
-rw-r--r--src/cmumble.c193
1 files changed, 5 insertions, 188 deletions
diff --git a/src/cmumble.c b/src/cmumble.c
index 3a40e21..3d060ce 100644
--- a/src/cmumble.c
+++ b/src/cmumble.c
@@ -23,55 +23,6 @@ find_user(struct context *ctx, uint32_t session)
}
static void
-appsrc_push(GstAppSrc *src, const void *mem, size_t size)
-{
- GstBuffer *gstbuf;
-
- gstbuf = gst_app_buffer_new(g_memdup(mem, size), size, g_free, NULL);
- gst_app_src_push_buffer(src, gstbuf);
-}
-
-static GstFlowReturn
-pull_buffer(GstAppSink *sink, gpointer user_data)
-{
- struct context *ctx = user_data;
- GstBuffer *buf;
- uint8_t data[1024];
- uint32_t write = 0, pos = 0;
- MumbleProto__UDPTunnel tunnel;
- static int seq = 0;
-
- buf = gst_app_sink_pull_buffer(ctx->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) | (0 << 4);
-
- encode_varint(&data[pos], &write, ++ctx->sequence, 1024-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(ctx, &tunnel.base);
-
- return GST_FLOW_OK;
-}
-
-static void
recv_udp_tunnel(MumbleProto__UDPTunnel *tunnel, struct context *ctx)
{
int64_t session, sequence;
@@ -100,7 +51,7 @@ recv_udp_tunnel(MumbleProto__UDPTunnel *tunnel, struct context *ctx)
if (frame_len == 0 || frame_len > len-pos)
break;
- appsrc_push(user->src, &data[pos], frame_len);
+ cmumble_audio_push(ctx, user, &data[pos], frame_len);
pos += frame_len;
sequence++;
@@ -177,9 +128,6 @@ recv_user_remove(MumbleProto__UserRemove *remove, struct context *ctx)
}
}
-static int
-user_create_playback_pipeline(struct context *ctx, struct user *user);
-
static void
recv_user_state(MumbleProto__UserState *state, struct context *ctx)
{
@@ -200,7 +148,8 @@ recv_user_state(MumbleProto__UserState *state, struct context *ctx)
user->name = g_strdup(state->name);
user->user_id = state->user_id;
- user_create_playback_pipeline(ctx, user);
+
+ cmumble_audio_create_playback_pipeline(ctx, user);
g_print("receive user: %s\n", user->name);
ctx->users = g_list_prepend(ctx->users, user);
}
@@ -239,135 +188,6 @@ static const struct {
.SuggestConfig = NULL,
};
-static void
-set_pulse_states(gpointer data, gpointer user_data)
-{
- GstElement *elm = data;
- struct 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));
-}
-
-static int
-user_create_playback_pipeline(struct context *ctx, struct 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 */
- appsrc_push(user->src, ctx->celt_header_packet, sizeof(CELTHeader));
- /* fake vorbiscomment buffer */
- appsrc_push(user->src, NULL, 0);
-
- return 0;
-}
-
-static int
-setup_playback_gst_pipeline(struct context *ctx)
-{
-#define SAMPLERATE 48000
-#define CHANNELS 1
- ctx->celt_mode = celt_mode_create(SAMPLERATE,
- SAMPLERATE / 100, NULL);
- celt_header_init(&ctx->celt_header, ctx->celt_mode, CHANNELS);
- celt_header_to_packet(&ctx->celt_header,
- ctx->celt_header_packet, sizeof(CELTHeader));
-
- return 0;
-}
-
-static int
-setup_recording_gst_pipeline(struct context *ctx)
-{
- 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");
- ctx->sink = GST_APP_SINK(sink);
- ctx->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(ctx->sink, TRUE);
- gst_app_sink_set_drop(ctx->sink, FALSE);;
- g_signal_connect(sink, "new-buffer", G_CALLBACK(pull_buffer), ctx);
-
- 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(ctx->sink, caps);
- gst_caps_unref(caps);
-
- gst_element_set_state(pipeline, GST_STATE_PLAYING);
-
- ctx->sequence = 0;
-
- return 0;
-}
-
int main(int argc, char **argv)
{
char *host = "localhost";
@@ -411,12 +231,8 @@ int main(int argc, char **argv)
gst_init(&argc, &argv);
- if (setup_playback_gst_pipeline(&ctx) < 0)
+ if (cmumble_audio_init(&ctx) < 0)
return 1;
-
- if (setup_recording_gst_pipeline(&ctx) < 0)
- return 1;
-
cmumble_io_init(&ctx);
g_main_loop_run(ctx.loop);
@@ -424,6 +240,7 @@ int main(int argc, char **argv)
g_main_loop_unref(ctx.loop);
cmumble_io_fini(&ctx);
+ cmumble_audio_init(&ctx);
cmumble_connection_fini(&ctx);
return 0;
generated by cgit (git 2.34.1) at 2024-05-17 08:15:33 +0000