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fix GStreamer capitalization

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This commit is contained in:
aler9
2022-03-15 11:42:52 +01:00
parent e7efb0c405
commit bf74ea8ec7
5 changed files with 11 additions and 11 deletions
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6
examples/client-publish-aac/main.go
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@@ -9,9 +9,9 @@ import (
) )
// This example shows how to // This example shows how to
// 1. generate RTP/AAC packets with Gstreamer // 1. generate RTP/AAC packets with GStreamer
// 2. connect to a RTSP server, announce an AAC track // 2. connect to a RTSP server, announce an AAC track
// 3. route the packets from Gstreamer to the server // 3. route the packets from GStreamer to the server
func main() { func main() {
// open a listener to receive RTP/AAC packets // open a listener to receive RTP/AAC packets
@@ -21,7 +21,7 @@ func main() {
} }
defer pc.Close() defer pc.Close()
log.Println("Waiting for a RTP/AAC stream on UDP port 9000 - you can send one with Gstreamer:\n" + log.Println("Waiting for a RTP/AAC stream on UDP port 9000 - you can send one with GStreamer:\n" +
"gst-launch-1.0 audiotestsrc freq=300 ! audioconvert ! audioresample ! audio/x-raw,rate=48000" + "gst-launch-1.0 audiotestsrc freq=300 ! audioconvert ! audioresample ! audio/x-raw,rate=48000" +
" ! avenc_aac bitrate=128000" + " ! avenc_aac bitrate=128000" +
" ! rtpmp4gpay ! udpsink host=127.0.0.1 port=9000") " ! rtpmp4gpay ! udpsink host=127.0.0.1 port=9000")

6
examples/client-publish-h264/main.go
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@@ -10,9 +10,9 @@ import (
) )
// This example shows how to // This example shows how to
// 1. generate RTP/H264 packets with Gstreamer // 1. generate RTP/H264 packets with GStreamer
// 2. connect to a RTSP server, announce an H264 track // 2. connect to a RTSP server, announce an H264 track
// 3. route the packets from Gstreamer to the server // 3. route the packets from GStreamer to the server
func main() { func main() {
// open a listener to receive RTP/H264 packets // open a listener to receive RTP/H264 packets
@@ -22,7 +22,7 @@ func main() {
} }
defer pc.Close() defer pc.Close()
log.Println("Waiting for a RTP/H264 stream on UDP port 9000 - you can send one with Gstreamer:\n" + log.Println("Waiting for a RTP/H264 stream on UDP port 9000 - you can send one with GStreamer:\n" +
"gst-launch-1.0 videotestsrc ! video/x-raw,width=1920,height=1080" + "gst-launch-1.0 videotestsrc ! video/x-raw,width=1920,height=1080" +
" ! x264enc speed-preset=veryfast tune=zerolatency bitrate=600000" + " ! x264enc speed-preset=veryfast tune=zerolatency bitrate=600000" +
" ! rtph264pay ! udpsink host=127.0.0.1 port=9000") " ! rtph264pay ! udpsink host=127.0.0.1 port=9000")

2
examples/client-publish-options/main.go
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@@ -12,7 +12,7 @@ import (
// This example shows how to // This example shows how to
// 1. set additional client options // 1. set additional client options
// 2. generate RTP/H264 frames from a file with Gstreamer // 2. generate RTP/H264 frames from a file with GStreamer
// 3. connect to a RTSP server, announce an H264 track // 3. connect to a RTSP server, announce an H264 track
// 4. write the frames to the server // 4. write the frames to the server

6
examples/client-publish-opus/main.go
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@@ -9,9 +9,9 @@ import (
) )
// This example shows how to // This example shows how to
// 1. generate RTP/Opus packets with Gstreamer // 1. generate RTP/Opus packets with GStreamer
// 2. connect to a RTSP server, announce an Opus track // 2. connect to a RTSP server, announce an Opus track
// 3. route the packets from Gstreamer to the server // 3. route the packets from GStreamer to the server
func main() { func main() {
// open a listener to receive RTP/Opus packets // open a listener to receive RTP/Opus packets
@@ -21,7 +21,7 @@ func main() {
} }
defer pc.Close() defer pc.Close()
log.Println("Waiting for a RTP/Opus stream on UDP port 9000 - you can send one with Gstreamer:\n" + log.Println("Waiting for a RTP/Opus stream on UDP port 9000 - you can send one with GStreamer:\n" +
"gst-launch-1.0 audiotestsrc freq=300 ! audioconvert ! audioresample ! audio/x-raw,rate=48000" + "gst-launch-1.0 audiotestsrc freq=300 ! audioconvert ! audioresample ! audio/x-raw,rate=48000" +
" ! opusenc" + " ! opusenc" +
" ! rtpopuspay ! udpsink host=127.0.0.1 port=9000") " ! rtpopuspay ! udpsink host=127.0.0.1 port=9000")

2
examples/client-publish-pause/main.go
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@@ -11,7 +11,7 @@ import (
) )
// This example shows how to // This example shows how to
// 1. generate RTP/H264 frames from a file with Gstreamer // 1. generate RTP/H264 frames from a file with GStreamer
// 2. connect to a RTSP server, announce an H264 track // 2. connect to a RTSP server, announce an H264 track
// 3. write the frames to the server for 5 seconds // 3. write the frames to the server for 5 seconds
// 4. pause for 5 seconds // 4. pause for 5 seconds