diff options
| author | Seungha Yang <seungha@centricular.com> | 2020-12-10 19:16:52 +0900 |
|---|---|---|
| committer | Seungha Yang <seungha@centricular.com> | 2020-12-10 20:18:30 +0900 |
| commit | 767f46b1a0339e2729d21290c5f1940bc52ea941 (patch) | |
| tree | 323cd6e072047923f41974ada1ac334701fdfede | |
| parent | 85aeda42fe6f0096553647f4b0c71fb0ca9af832 (diff) | |
webrtc: sendonly: Add support for Windows
Add meson build script and use mfvideosrc element in case of Windows
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/29>
| -rw-r--r-- | webrtc/meson.build | 1 | ||||
| -rw-r--r-- | webrtc/sendonly/meson.build | 7 | ||||
| -rw-r--r-- | webrtc/sendonly/webrtc-unidirectional-h264.c | 11 |
3 files changed, 17 insertions, 2 deletions
diff --git a/webrtc/meson.build b/webrtc/meson.build index 05fa6a4..5efc2ec 100644 --- a/webrtc/meson.build +++ b/webrtc/meson.build @@ -17,6 +17,7 @@ endif subdir('multiparty-sendrecv') subdir('signalling') +subdir('sendonly') subdir('sendrecv') subdir('check') diff --git a/webrtc/sendonly/meson.build b/webrtc/sendonly/meson.build new file mode 100644 index 0000000..bb41eca --- /dev/null +++ b/webrtc/sendonly/meson.build @@ -0,0 +1,7 @@ +executable('webrtc-recvonly-h264', + 'webrtc-recvonly-h264.c', + dependencies : [gst_dep, gstsdp_dep, gstwebrtc_dep, libsoup_dep, json_glib_dep ]) + +executable('webrtc-unidirectional-h264', + 'webrtc-unidirectional-h264.c', + dependencies : [gst_dep, gstsdp_dep, gstwebrtc_dep, libsoup_dep, json_glib_dep ]) diff --git a/webrtc/sendonly/webrtc-unidirectional-h264.c b/webrtc/sendonly/webrtc-unidirectional-h264.c index ea145c8..48fe8a0 100644 --- a/webrtc/sendonly/webrtc-unidirectional-h264.c +++ b/webrtc/sendonly/webrtc-unidirectional-h264.c @@ -19,6 +19,12 @@ #define SOUP_HTTP_PORT 57778 #define STUN_SERVER "stun.l.google.com:19302" +#ifdef G_OS_WIN32 +#define VIDEO_SRC "mfvideosrc" +#else +#define VIDEO_SRC "v4l2src" +#endif + gchar *video_priority = NULL; gchar *audio_priority = NULL; @@ -232,11 +238,12 @@ create_receiver_entry (SoupWebsocketConnection * connection) receiver_entry->pipeline = gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://" STUN_SERVER " " - "v4l2src ! videorate ! videoscale ! video/x-raw,width=640,height=360,framerate=15/1 ! videoconvert ! queue max-size-buffers=1 ! x264enc bitrate=600 speed-preset=ultrafast tune=zerolatency key-int-max=15 ! video/x-h264,profile=constrained-baseline ! queue max-size-time=100000000 ! h264parse ! " + VIDEO_SRC + " ! videorate ! videoscale ! video/x-raw,width=640,height=360,framerate=15/1 ! videoconvert ! queue max-size-buffers=1 ! x264enc bitrate=600 speed-preset=ultrafast tune=zerolatency key-int-max=15 ! video/x-h264,profile=constrained-baseline ! queue max-size-time=100000000 ! h264parse ! " "rtph264pay config-interval=-1 name=payloader aggregate-mode=zero-latency ! " "application/x-rtp,media=video,encoding-name=H264,payload=" RTP_PAYLOAD_TYPE " ! webrtcbin. " - "autoaudiosrc is-live=1 ! queue max-size-buffers=1 leaky=downstream ! opusenc ! rtpopuspay pt=" + "autoaudiosrc is-live=1 ! queue max-size-buffers=1 leaky=downstream ! audioconvert ! audioresample ! opusenc ! rtpopuspay pt=" RTP_AUDIO_PAYLOAD_TYPE " ! webrtcbin. ", &error); if (error != NULL) { g_error ("Could not create WebRTC pipeline: %s\n", error->message); |