/*
This file is part of GNUnet.
- Copyright (C) 2013 Christian Grothoff (and other contributing authors)
+ Copyright (C) 2013 GNUnet e.V.
- GNUnet is free software; you can redistribute it and/or modify
- it under the terms of the GNU General Public License as published
- by the Free Software Foundation; either version 3, or (at your
- option) any later version.
+ GNUnet is free software: you can redistribute it and/or modify it
+ under the terms of the GNU Affero General Public License as published
+ by the Free Software Foundation, either version 3 of the License,
+ or (at your option) any later version.
GNUnet is distributed in the hope that it will be useful, but
WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- General Public License for more details.
+ Affero General Public License for more details.
- You should have received a copy of the GNU General Public License
- along with GNUnet; see the file COPYING. If not, write to the
- Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- Boston, MA 02111-1307, USA.
-*/
+ You should have received a copy of the GNU Affero General Public License
+ along with this program. If not, see <http://www.gnu.org/licenses/>.
+
+ SPDX-License-Identifier: AGPL3.0-or-later
+ */
/**
* @file conversation/gnunet-helper-audio-playback-gst.c
* @brief program to playback audio data to the speaker (GStreamer version)
/**
* Max number of microseconds to buffer in audiosink.
- * Default is 200000
+ * Default is 1000
*/
#define BUFFER_TIME 1000
/**
* Min number of microseconds to buffer in audiosink.
- * Default is 10000
+ * Default is 1000
*/
#define LATENCY_TIME 1000
/**
* Tokenizer for the data we get from stdin
*/
-struct GNUNET_SERVER_MessageStreamTokenizer *stdin_mst;
+struct GNUNET_MessageStreamTokenizer *stdin_mst;
/**
* Main pipeline.
*/
static int abort_read;
+
static void
-sink_child_added (GstChildProxy *child_proxy, GObject *object, gchar *name, gpointer user_data)
+sink_child_added(GstChildProxy *child_proxy,
+ GObject *object,
+ gchar *name,
+ gpointer user_data)
{
- if (GST_IS_AUDIO_BASE_SRC (object))
- g_object_set (object, "buffer-time", (gint64) BUFFER_TIME, "latency-time", (gint64) LATENCY_TIME, NULL);
+ if (GST_IS_AUDIO_BASE_SRC(object))
+ g_object_set(object,
+ "buffer-time", (gint64)BUFFER_TIME,
+ "latency-time", (gint64)LATENCY_TIME,
+ NULL);
}
+
static void
-ogg_pad_added (GstElement *element, GstPad *pad, gpointer data)
+ogg_pad_added(GstElement *element,
+ GstPad *pad,
+ gpointer data)
{
GstPad *sinkpad;
- GstElement *decoder = (GstElement *) data;
+ GstElement *decoder = (GstElement *)data;
/* We can now link this pad with the opus-decoder sink pad */
- sinkpad = gst_element_get_static_pad (decoder, "sink");
+ sinkpad = gst_element_get_static_pad(decoder, "sink");
- gst_pad_link (pad, sinkpad);
+ gst_pad_link(pad, sinkpad);
- gst_element_link_many (decoder, conv, resampler, sink, NULL);
+ gst_element_link_many(decoder, conv, resampler, sink, NULL);
- gst_object_unref (sinkpad);
+ gst_object_unref(sinkpad);
}
+
static void
-quit ()
+quit()
{
if (NULL != source)
- gst_app_src_end_of_stream (GST_APP_SRC (source));
+ gst_app_src_end_of_stream(GST_APP_SRC(source));
if (NULL != pipeline)
- gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_element_set_state(pipeline, GST_STATE_NULL);
abort_read = 1;
}
+
static gboolean
-bus_call (GstBus *bus, GstMessage *msg, gpointer data)
+bus_call(GstBus *bus, GstMessage *msg, gpointer data)
{
- GNUNET_log (GNUNET_ERROR_TYPE_DEBUG, "Bus message\n");
- switch (GST_MESSAGE_TYPE (msg))
- {
- case GST_MESSAGE_EOS:
- GNUNET_log (GNUNET_ERROR_TYPE_INFO, "End of stream\n");
- quit ();
- break;
-
- case GST_MESSAGE_ERROR:
+ GNUNET_log(GNUNET_ERROR_TYPE_DEBUG,
+ "Bus message\n");
+ switch (GST_MESSAGE_TYPE(msg))
+ {
+ case GST_MESSAGE_EOS:
+ GNUNET_log(GNUNET_ERROR_TYPE_INFO,
+ "End of stream\n");
+ quit();
+ break;
+
+ case GST_MESSAGE_ERROR:
{
gchar *debug;
GError *error;
-
- gst_message_parse_error (msg, &error, &debug);
- g_free (debug);
-
- GNUNET_log (GNUNET_ERROR_TYPE_ERROR, "Error: %s\n", error->message);
- g_error_free (error);
-
- quit ();
+
+ gst_message_parse_error(msg, &error, &debug);
+ g_free(debug);
+
+ GNUNET_log(GNUNET_ERROR_TYPE_ERROR,
+ "Error: %s\n",
+ error->message);
+ g_error_free(error);
+
+ quit();
+ break;
+ }
+
+ default:
break;
}
- default:
- break;
- }
return TRUE;
}
static void
-signalhandler (int s)
+signalhandler(int s)
{
- quit ();
+ quit();
}
+
static int
-feed_buffer_to_gst (const char *audio, size_t b_len)
+feed_buffer_to_gst(const char *audio, size_t b_len)
{
GstBuffer *b;
gchar *bufspace;
GstFlowReturn flow;
- GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
- "Feeding %u bytes to GStreamer\n",
- (unsigned int) b_len);
+ GNUNET_log(GNUNET_ERROR_TYPE_DEBUG,
+ "Feeding %u bytes to GStreamer\n",
+ (unsigned int)b_len);
- bufspace = g_memdup (audio, b_len);
- b = gst_buffer_new_wrapped (bufspace, b_len);
+ bufspace = g_memdup(audio, b_len);
+ b = gst_buffer_new_wrapped(bufspace, b_len);
if (NULL == b)
- {
- GNUNET_log (GNUNET_ERROR_TYPE_DEBUG, "Failed to wrap a buffer\n");
- g_free (bufspace);
- return GNUNET_SYSERR;
- }
- flow = gst_app_src_push_buffer (GST_APP_SRC (source), b);
+ {
+ GNUNET_log(GNUNET_ERROR_TYPE_DEBUG,
+ "Failed to wrap a buffer\n");
+ g_free(bufspace);
+ return GNUNET_SYSERR;
+ }
+ flow = gst_app_src_push_buffer(GST_APP_SRC(source), b);
/* They all return GNUNET_OK, because currently player stops when
* data stops coming. This might need to be changed for the player
* to also stop when pipeline breaks.
*/
switch (flow)
- {
- case GST_FLOW_OK:
- GNUNET_log (GNUNET_ERROR_TYPE_DEBUG, "Fed %u bytes to the pipeline\n",
- (unsigned int) b_len);
- break;
- case GST_FLOW_FLUSHING:
- /* buffer was dropped, because pipeline state is not PAUSED or PLAYING */
- GNUNET_log (GNUNET_ERROR_TYPE_INFO, "Dropped a buffer\n");
- break;
- case GST_FLOW_EOS:
- /* end of stream */
- GNUNET_log (GNUNET_ERROR_TYPE_INFO, "EOS\n");
- break;
- default:
- GNUNET_log (GNUNET_ERROR_TYPE_WARNING, "Unexpected push result\n");
- break;
- }
+ {
+ case GST_FLOW_OK:
+ GNUNET_log(GNUNET_ERROR_TYPE_DEBUG,
+ "Fed %u bytes to the pipeline\n",
+ (unsigned int)b_len);
+ break;
+
+ case GST_FLOW_FLUSHING:
+ /* buffer was dropped, because pipeline state is not PAUSED or PLAYING */
+ GNUNET_log(GNUNET_ERROR_TYPE_INFO,
+ "Dropped a buffer\n");
+ break;
+
+ case GST_FLOW_EOS:
+ /* end of stream */
+ GNUNET_log(GNUNET_ERROR_TYPE_INFO,
+ "EOS\n");
+ break;
+
+ default:
+ GNUNET_log(GNUNET_ERROR_TYPE_WARNING,
+ "Unexpected push result\n");
+ break;
+ }
return GNUNET_OK;
}
+
/**
* Message callback
+ *
+ * @param msg message we received.
+ * @return #GNUNET_OK on success,
+ * #GNUNET_NO to stop further processing due to disconnect (no error)
+ * #GNUNET_SYSERR to stop further processing due to error
*/
static int
-stdin_receiver (void *cls,
- void *client,
- const struct GNUNET_MessageHeader *msg)
+stdin_receiver(void *cls,
+ const struct GNUNET_MessageHeader *msg)
{
struct AudioMessage *audio;
size_t b_len;
- switch (ntohs (msg->type))
- {
- case GNUNET_MESSAGE_TYPE_CONVERSATION_AUDIO:
- audio = (struct AudioMessage *) msg;
-
- b_len = ntohs (audio->header.size) - sizeof (struct AudioMessage);
- feed_buffer_to_gst ((const char *) &audio[1], b_len);
- break;
- default:
- break;
- }
+ switch (ntohs(msg->type))
+ {
+ case GNUNET_MESSAGE_TYPE_CONVERSATION_AUDIO:
+ audio = (struct AudioMessage *)msg;
+
+ b_len = ntohs(audio->header.size) - sizeof(struct AudioMessage);
+ feed_buffer_to_gst((const char *)&audio[1], b_len);
+ break;
+
+ default:
+ break;
+ }
return GNUNET_OK;
}
int
-main (int argc, char **argv)
+main(int argc, char **argv)
{
GstBus *bus;
guint bus_watch_id;
uint64_t toff;
typedef void (*SignalHandlerPointer) (int);
-
+
SignalHandlerPointer inthandler, termhandler;
#ifdef DEBUG_READ_PURE_OGG
- int read_pure_ogg = getenv ("GNUNET_READ_PURE_OGG") ? 1 : 0;
+ int read_pure_ogg = getenv("GNUNET_READ_PURE_OGG") ? 1 : 0;
#endif
- inthandler = signal (SIGINT, signalhandler);
- termhandler = signal (SIGTERM, signalhandler);
+ inthandler = signal(SIGINT,
+ &signalhandler);
+ termhandler = signal(SIGTERM,
+ &signalhandler);
#ifdef WINDOWS
- setmode (0, _O_BINARY);
+ setmode(0, _O_BINARY);
#endif
/* Initialisation */
- gst_init (&argc, &argv);
+ gst_init(&argc, &argv);
- GNUNET_assert (GNUNET_OK ==
- GNUNET_log_setup ("gnunet-helper-audio-playback",
- "WARNING",
- NULL));
+ GNUNET_assert(GNUNET_OK ==
+ GNUNET_log_setup("gnunet-helper-audio-playback-gst",
+ "WARNING",
+ NULL));
- GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
- "Audio sink starts\n");
+ GNUNET_log(GNUNET_ERROR_TYPE_DEBUG,
+ "Audio sink starts\n");
- stdin_mst = GNUNET_SERVER_mst_create (&stdin_receiver, NULL);
+ stdin_mst = GNUNET_MST_create(&stdin_receiver,
+ NULL);
/* Create gstreamer elements */
- pipeline = gst_pipeline_new ("audio-player");
- source = gst_element_factory_make ("appsrc", "audio-input");
- demuxer = gst_element_factory_make ("oggdemux", "ogg-demuxer");
- decoder = gst_element_factory_make ("opusdec", "opus-decoder");
- conv = gst_element_factory_make ("audioconvert", "converter");
- resampler= gst_element_factory_make ("audioresample", "resampler");
- sink = gst_element_factory_make ("autoaudiosink", "audiosink");
+ pipeline = gst_pipeline_new("audio-player");
+ source = gst_element_factory_make("appsrc", "audio-input");
+ demuxer = gst_element_factory_make("oggdemux", "ogg-demuxer");
+ decoder = gst_element_factory_make("opusdec", "opus-decoder");
+ conv = gst_element_factory_make("audioconvert", "converter");
+ resampler = gst_element_factory_make("audioresample", "resampler");
+ sink = gst_element_factory_make("autoaudiosink", "audiosink");
if (!pipeline || !source || !conv || !resampler || !decoder || !demuxer || !sink)
- {
- GNUNET_log (GNUNET_ERROR_TYPE_ERROR,
- "One element could not be created. Exiting.\n");
- return -1;
- }
+ {
+ GNUNET_log(GNUNET_ERROR_TYPE_ERROR,
+ "One element could not be created. Exiting.\n");
+ return -1;
+ }
- g_signal_connect (sink, "child-added", G_CALLBACK (sink_child_added), NULL);
- g_signal_connect (demuxer, "pad-added", G_CALLBACK (ogg_pad_added), decoder);
+ g_signal_connect(sink,
+ "child-added",
+ G_CALLBACK(sink_child_added),
+ NULL);
+ g_signal_connect(demuxer,
+ "pad-added",
+ G_CALLBACK(ogg_pad_added),
+ decoder);
/* Keep a reference to it, we operate on it */
- gst_object_ref (GST_OBJECT (source));
+ gst_object_ref(GST_OBJECT(source));
/* Set up the pipeline */
/* we feed appsrc as fast as possible, it just blocks when it's full */
- g_object_set (G_OBJECT (source),
+ g_object_set(G_OBJECT(source),
/* "format", GST_FORMAT_TIME,*/
- "block", TRUE,
- "is-live", TRUE,
- NULL);
+ "block", TRUE,
+ "is-live", TRUE,
+ NULL);
- g_object_set (G_OBJECT (decoder),
+ g_object_set(G_OBJECT(decoder),
/* "plc", FALSE,*/
/* "apply-gain", TRUE,*/
- "use-inband-fec", TRUE,
- NULL);
+ "use-inband-fec", TRUE,
+ NULL);
/* we add a message handler */
- bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
- bus_watch_id = gst_bus_add_watch (bus, bus_call, pipeline);
- gst_object_unref (bus);
+ bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
+ bus_watch_id = gst_bus_add_watch(bus, bus_call, pipeline);
+ gst_object_unref(bus);
/* we add all elements into the pipeline */
/* audio-input | ogg-demuxer | opus-decoder | converter | resampler | audiosink */
- gst_bin_add_many (GST_BIN (pipeline), source, demuxer, decoder, conv,
- resampler, sink, NULL);
+ gst_bin_add_many(GST_BIN(pipeline), source, demuxer, decoder, conv,
+ resampler, sink, NULL);
/* we link the elements together */
- gst_element_link_many (source, demuxer, NULL);
+ gst_element_link_many(source, demuxer, NULL);
/* Set the pipeline to "playing" state*/
- GNUNET_log (GNUNET_ERROR_TYPE_INFO, "Now playing\n");
- gst_element_set_state (pipeline, GST_STATE_PLAYING);
+ GNUNET_log(GNUNET_ERROR_TYPE_INFO, "Now playing\n");
+ gst_element_set_state(pipeline, GST_STATE_PLAYING);
- GNUNET_log (GNUNET_ERROR_TYPE_INFO, "Running...\n");
+ GNUNET_log(GNUNET_ERROR_TYPE_INFO, "Running...\n");
/* Iterate */
toff = 0;
while (!abort_read)
- {
- char readbuf[MAXLINE];
- int ret;
-
- ret = read (0, readbuf, sizeof (readbuf));
- if (0 > ret)
{
- GNUNET_log (GNUNET_ERROR_TYPE_ERROR,
- _("Read error from STDIN: %d %s\n"),
- ret, strerror (errno));
- break;
- }
- toff += ret;
- GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
- "Received %d bytes of audio data (total: %llu)\n",
- (int) ret,
- toff);
- if (0 == ret)
- break;
+ char readbuf[MAXLINE];
+ int ret;
+
+ ret = read(0, readbuf, sizeof(readbuf));
+ if (0 > ret)
+ {
+ GNUNET_log(GNUNET_ERROR_TYPE_ERROR,
+ _("Read error from STDIN: %d %s\n"),
+ ret, strerror(errno));
+ break;
+ }
+ toff += ret;
+ GNUNET_log(GNUNET_ERROR_TYPE_DEBUG,
+ "Received %d bytes of audio data (total: %llu)\n",
+ (int)ret,
+ (unsigned long long)toff);
+ if (0 == ret)
+ break;
#ifdef DEBUG_READ_PURE_OGG
- if (read_pure_ogg)
- {
- feed_buffer_to_gst (readbuf, ret);
- }
- else
+ if (read_pure_ogg)
+ {
+ feed_buffer_to_gst(readbuf, ret);
+ }
+ else
#endif
- GNUNET_SERVER_mst_receive (stdin_mst, NULL,
- readbuf, ret,
- GNUNET_NO, GNUNET_NO);
- }
- GNUNET_SERVER_mst_destroy (stdin_mst);
+ GNUNET_MST_from_buffer(stdin_mst,
+ readbuf,
+ ret,
+ GNUNET_NO,
+ GNUNET_NO);
+ }
+ GNUNET_MST_destroy(stdin_mst);
- signal (SIGINT, inthandler);
- signal (SIGINT, termhandler);
+ signal(SIGINT, inthandler);
+ signal(SIGINT, termhandler);
- GNUNET_log (GNUNET_ERROR_TYPE_INFO, "Returned, stopping playback\n");
- quit ();
+ GNUNET_log(GNUNET_ERROR_TYPE_INFO,
+ "Returned, stopping playback\n");
+ quit();
- GNUNET_log (GNUNET_ERROR_TYPE_INFO, "Deleting pipeline\n");
- gst_object_unref (GST_OBJECT (source));
+ GNUNET_log(GNUNET_ERROR_TYPE_INFO,
+ "Deleting pipeline\n");
+ gst_object_unref(GST_OBJECT(source));
source = NULL;
- gst_object_unref (GST_OBJECT (pipeline));
+ gst_object_unref(GST_OBJECT(pipeline));
pipeline = NULL;
- g_source_remove (bus_watch_id);
+ g_source_remove(bus_watch_id);
return 0;
}
projects / oweals / gnunet.git / blobdiff