diff options
Diffstat (limited to 'src/audio.c')
| -rw-r--r-- | src/audio.c | 205 |
1 files changed, 205 insertions, 0 deletions
diff --git a/src/audio.c b/src/audio.c new file mode 100644 index 0000000..519dc04 --- /dev/null +++ b/src/audio.c @@ -0,0 +1,205 @@ +#include "audio.h" +#include "varint.h" +#include "cmumble.h" +#include <string.h> + +#define SAMPLERATE 48000 +#define CHANNELS 1 + +void +cmumble_audio_push(struct context *ctx, struct user *user, + const uint8_t *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 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->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) | (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 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->audio.sink = GST_APP_SINK(sink); + ctx->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(ctx->audio.sink, TRUE); + gst_app_sink_set_drop(ctx->audio.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->audio.sink, caps); + gst_caps_unref(caps); + + gst_element_set_state(pipeline, GST_STATE_PLAYING); + + ctx->sequence = 0; + + return 0; +} + +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)); +} + +int +cmumble_audio_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 */ + cmumble_audio_push(ctx, user, + ctx->audio.celt_header_packet, sizeof(CELTHeader)); + /* fake vorbiscomment buffer */ + cmumble_audio_push(ctx, user, NULL, 0); + + return 0; +} + +static int +setup_playback_gst_pipeline(struct context *ctx) +{ + ctx->audio.celt_mode = celt_mode_create(SAMPLERATE, + SAMPLERATE / 100, NULL); + celt_header_init(&ctx->audio.celt_header, ctx->audio.celt_mode, CHANNELS); + celt_header_to_packet(&ctx->audio.celt_header, + ctx->audio.celt_header_packet, sizeof(CELTHeader)); + + return 0; +} + +int +cmumble_audio_init(struct context *ctx) +{ + if (setup_playback_gst_pipeline(ctx) < 0) + return -1; + + if (setup_recording_gst_pipeline(ctx) < 0) + return -1; + + return 0; +} + +int +cmumble_audio_fini(struct context *ctx) +{ + + return 0; +} + |