gstreamer/examples/test-appsrc2.c
Jan Schmidt b6c3960f11 examples: Add test-appsrc2
Add an example of feeding both audio and video into an RTSP
pipeline via appsrc.
2018-07-11 01:25:51 +10:00

196 lines
6.9 KiB
C

/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <gst/gst.h>
#include <gst/app/app.h>
#include <gst/rtsp-server/rtsp-server.h>
typedef struct
{
GstElement *generator_pipe;
GstElement *vid_appsink;
GstElement *vid_appsrc;
GstElement *aud_appsink;
GstElement *aud_appsrc;
} MyContext;
/* called when we need to give data to an appsrc */
static void
need_data (GstElement * appsrc, guint unused, MyContext * ctx)
{
GstSample *sample;
GstFlowReturn ret;
if (appsrc == ctx->vid_appsrc)
sample = gst_app_sink_pull_sample (GST_APP_SINK (ctx->vid_appsink));
else
sample = gst_app_sink_pull_sample (GST_APP_SINK (ctx->aud_appsink));
if (sample) {
GstBuffer *buffer = gst_sample_get_buffer (sample);
GstSegment *seg = gst_sample_get_segment (sample);
GstClockTime pts, dts;
/* Convert the PTS/DTS to running time so they start from 0 */
pts = GST_BUFFER_PTS (buffer);
if (GST_CLOCK_TIME_IS_VALID (pts))
pts = gst_segment_to_running_time (seg, GST_FORMAT_TIME, pts);
dts = GST_BUFFER_DTS (buffer);
if (GST_CLOCK_TIME_IS_VALID (dts))
dts = gst_segment_to_running_time (seg, GST_FORMAT_TIME, dts);
if (buffer) {
/* Make writable so we can adjust the timestamps */
buffer = gst_buffer_copy (buffer);
GST_BUFFER_PTS (buffer) = pts;
GST_BUFFER_DTS (buffer) = dts;
g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
}
/* we don't need the appsink sample anymore */
gst_sample_unref (sample);
}
}
static void
ctx_free (MyContext * ctx)
{
gst_element_set_state (ctx->generator_pipe, GST_STATE_NULL);
gst_object_unref (ctx->generator_pipe);
gst_object_unref (ctx->vid_appsrc);
gst_object_unref (ctx->vid_appsink);
gst_object_unref (ctx->aud_appsrc);
gst_object_unref (ctx->aud_appsink);
g_free (ctx);
}
/* called when a new media pipeline is constructed. We can query the
* pipeline and configure our appsrc */
static void
media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media,
gpointer user_data)
{
GstElement *element, *appsrc, *appsink;
GstCaps *caps;
MyContext *ctx;
ctx = g_new0 (MyContext, 1);
/* This pipeline generates H264 video and PCM audio. The appsinks are kept small so that if delivery is slow,
* encoded buffers are dropped as needed. There's slightly more buffers (32) allowed for audio */
ctx->generator_pipe =
gst_parse_launch
("videotestsrc is-live=true ! x264enc speed-preset=superfast tune=zerolatency ! h264parse ! appsink name=vid max-buffers=1 drop=true "
"audiotestsrc is-live=true ! appsink name=aud max-buffers=32 drop=true",
NULL);
/* make sure the data is freed when the media is gone */
g_object_set_data_full (G_OBJECT (media), "rtsp-extra-data", ctx,
(GDestroyNotify) ctx_free);
/* get the element (bin) used for providing the streams of the media */
element = gst_rtsp_media_get_element (media);
/* Find the 2 app sources (video / audio), and configure them, connect to the
* signals to request data */
/* configure the caps of the video */
caps = gst_caps_new_simple ("video/x-h264",
"stream-format", G_TYPE_STRING, "byte-stream",
"alignment", G_TYPE_STRING, "au",
"width", G_TYPE_INT, 384, "height", G_TYPE_INT, 288,
"framerate", GST_TYPE_FRACTION, 15, 1, NULL);
ctx->vid_appsrc = appsrc =
gst_bin_get_by_name_recurse_up (GST_BIN (element), "videosrc");
ctx->vid_appsink = appsink =
gst_bin_get_by_name (GST_BIN (ctx->generator_pipe), "vid");
gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
g_object_set (G_OBJECT (appsrc), "caps", caps, NULL);
g_object_set (G_OBJECT (appsink), "caps", caps, NULL);
/* install the callback that will be called when a buffer is needed */
g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
gst_caps_unref (caps);
caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S24BE",
"layout", G_TYPE_STRING, "interleaved", "rate", G_TYPE_INT, 48000,
"channels", G_TYPE_INT, 2, NULL);
ctx->aud_appsrc = appsrc =
gst_bin_get_by_name_recurse_up (GST_BIN (element), "audiosrc");
ctx->aud_appsink = appsink =
gst_bin_get_by_name (GST_BIN (ctx->generator_pipe), "aud");
gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
g_object_set (G_OBJECT (appsrc), "caps", caps, NULL);
g_object_set (G_OBJECT (appsink), "caps", caps, NULL);
g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
gst_caps_unref (caps);
gst_element_set_state (ctx->generator_pipe, GST_STATE_PLAYING);
gst_object_unref (element);
}
int
main (int argc, char *argv[])
{
GMainLoop *loop;
GstRTSPServer *server;
GstRTSPMountPoints *mounts;
GstRTSPMediaFactory *factory;
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/* create a server instance */
server = gst_rtsp_server_new ();
/* get the mount points for this server, every server has a default object
* that be used to map uri mount points to media factories */
mounts = gst_rtsp_server_get_mount_points (server);
/* make a media factory for a test stream. The default media factory can use
* gst-launch syntax to create pipelines.
* any launch line works as long as it contains elements named pay%d. Each
* element with pay%d names will be a stream */
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_launch (factory,
"( appsrc name=videosrc ! h264parse ! rtph264pay name=pay0 pt=96 "
" appsrc name=audiosrc ! audioconvert ! rtpL24pay name=pay1 pt=97 )");
/* notify when our media is ready, This is called whenever someone asks for
* the media and a new pipeline with our appsrc is created */
g_signal_connect (factory, "media-configure", (GCallback) media_configure,
NULL);
/* attach the test factory to the /test url */
gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
/* don't need the ref to the mounts anymore */
g_object_unref (mounts);
/* attach the server to the default maincontext */
gst_rtsp_server_attach (server, NULL);
/* start serving */
g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
g_main_loop_run (loop);
return 0;
}