From b6c3960f1153d092097d2a0ce1845de377032a43 Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Wed, 11 Jul 2018 01:25:51 +1000 Subject: [PATCH] examples: Add test-appsrc2 Add an example of feeding both audio and video into an RTSP pipeline via appsrc. --- examples/.gitignore | 1 + examples/Makefile.am | 2 +- examples/test-appsrc2.c | 196 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 198 insertions(+), 1 deletion(-) create mode 100644 examples/test-appsrc2.c diff --git a/examples/.gitignore b/examples/.gitignore index d9df57c609..4a65f7420c 100644 --- a/examples/.gitignore +++ b/examples/.gitignore @@ -1,4 +1,5 @@ test-appsrc +test-appsrc2 test-cgroups test-launch test-mp4 diff --git a/examples/Makefile.am b/examples/Makefile.am index 6a70d728fe..66345c0043 100644 --- a/examples/Makefile.am +++ b/examples/Makefile.am @@ -1,6 +1,6 @@ noinst_PROGRAMS = test-video test-ogg test-mp4 test-readme \ test-launch test-sdp test-uri test-auth test-auth-digest \ - test-multicast test-multicast2 test-appsrc \ + test-multicast test-multicast2 test-appsrc test-appsrc2 \ test-video-rtx test-record test-record-auth \ test-netclock test-netclock-client \ test-onvif-backchannel test-video-disconnect diff --git a/examples/test-appsrc2.c b/examples/test-appsrc2.c new file mode 100644 index 0000000000..da2513ae8f --- /dev/null +++ b/examples/test-appsrc2.c @@ -0,0 +1,196 @@ +/* GStreamer + * Copyright (C) 2008 Wim Taymans + * + * 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 +#include + +#include + +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; +}