diff --git a/src/go2rtc/.gitignore b/src/go2rtc/.gitignore
new file mode 100644
index 0000000..d597da1
--- /dev/null
+++ b/src/go2rtc/.gitignore
@@ -0,0 +1,3 @@
+# Ignore the install dir
+_install/
+go2rtc/go2rtc
diff --git a/src/go2rtc/cleanup.go2rtc b/src/go2rtc/cleanup.go2rtc
new file mode 100755
index 0000000..abb588a
--- /dev/null
+++ b/src/go2rtc/cleanup.go2rtc
@@ -0,0 +1,7 @@
+#!/bin/bash
+
+SCRIPT_DIR=$(cd `dirname $0` && pwd)
+cd $SCRIPT_DIR
+
+cd go2rtc
+rm -f go2rtc
diff --git a/src/go2rtc/compile.go2rtc b/src/go2rtc/compile.go2rtc
new file mode 100755
index 0000000..2194de0
--- /dev/null
+++ b/src/go2rtc/compile.go2rtc
@@ -0,0 +1,7 @@
+#!/bin/bash
+
+SCRIPT_DIR=$(cd `dirname $0` && pwd)
+cd $SCRIPT_DIR
+
+mkdir -p ./_install/bin || exit 1
+cp go2rtc/go2rtc ./_install/bin || exit 1
diff --git a/src/go2rtc/init.go2rtc b/src/go2rtc/init.go2rtc
new file mode 100755
index 0000000..7a628d7
--- /dev/null
+++ b/src/go2rtc/init.go2rtc
@@ -0,0 +1,12 @@
+#!/bin/bash
+
+SCRIPT_DIR=$(cd `dirname $0` && pwd)
+cd $SCRIPT_DIR
+
+rm -rf ./_install
+
+cd go2rtc
+rm -f go2rtc
+wget https://github.com/AlexxIT/go2rtc/releases/download/v1.9.4/go2rtc_linux_arm || exit 1
+mv go2rtc_linux_arm go2rtc || exit 1
+chmod 0755 go2rtc || exit 1
diff --git a/src/go2rtc/install.go2rtc b/src/go2rtc/install.go2rtc
new file mode 100755
index 0000000..acf048b
--- /dev/null
+++ b/src/go2rtc/install.go2rtc
@@ -0,0 +1,8 @@
+#!/bin/bash
+
+SCRIPT_DIR=$(cd `dirname $0` && pwd)
+cd $SCRIPT_DIR
+
+mkdir -p ../../build/yi-hack/bin/ || exit 1
+
+rsync -av ./_install/* ../../build/yi-hack/ || exit 1
diff --git a/src/static/static/yi-hack/script/service.sh b/src/static/static/yi-hack/script/service.sh
index bc5383f..64732d5 100755
--- a/src/static/static/yi-hack/script/service.sh
+++ b/src/static/static/yi-hack/script/service.sh
@@ -48,24 +48,27 @@ init_config()
fi
if [[ $(get_config RTSP) == "yes" ]] ; then
- RTSP_DAEMON="rRTSPServer"
- RTSP_AUDIO_COMPRESSION=$(get_config RTSP_AUDIO)
- if [ "$RTSP_AUDIO" == "aac" ]; then
- H264GRABBER_AUDIO="-a"
+ if [[ $(get_config RTSP_ALT) == "alternative" ]] ; then
+ RTSP_DAEMON="rtsp_server_yi"
+ elif [[ $(get_config RTSP_ALT) == "go2rtc" ]] ; then
+ RTSP_DAEMON="go2rtc"
+ else
+ RTSP_DAEMON="rRTSPServer"
fi
- if [[ $(get_config RTSP_ALT) == "yes" ]] ; then
- RTSP_DAEMON="rtsp_server_yi"
+ RTSP_AUDIO_COMPRESSION=$(get_config RTSP_AUDIO)
+ if [ "$RTSP_AUDIO_COMPRESSION" == "aac" ]; then
+ H264GRABBER_AUDIO="-a"
fi
if [[ "$RTSP_AUDIO_COMPRESSION" == "none" ]] ; then
RTSP_AUDIO_COMPRESSION="no"
fi
if [ ! -z $RTSP_AUDIO_COMPRESSION ]; then
- RTSP_AUDIO_COMPRESSION="-a "$RTSP_AUDIO_COMPRESSION
+ RTSP_AUDIO="-a "$RTSP_AUDIO_COMPRESSION
fi
if [ ! -z $RTSP_PORT ]; then
- RTSP_PORT="-p "$RTSP_PORT
+ P_RTSP_PORT="-p "$RTSP_PORT
fi
if [ ! -z $USERNAME ]; then
RTSP_USER="-u "$USERNAME
@@ -107,6 +110,8 @@ init_config()
start_rtsp()
{
+ # If "null" use default
+
if [ "$1" == "low" ] || [ "$1" == "high" ] || [ "$1" == "both" ]; then
RTSP_RES=$1
fi
@@ -114,32 +119,68 @@ start_rtsp()
H264GRABBER_AUDIO="-a"
fi
if [ "$2" == "no" ] || [ "$2" == "yes" ] || [ "$2" == "alaw" ] || [ "$2" == "ulaw" ] || [ "$2" == "pcm" ] || [ "$2" == "aac" ] ; then
- RTSP_AUDIO_COMPRESSION="-a "$2
+ RTSP_AUDIO_COMPRESSION=$2
+ RTSP_AUDIO="-a "$2
fi
- if [[ $RTSP_RES == "low" ]]; then
- if [ "$RTSP_ALT" == "yes" ]; then
- h264grabber -m $MODEL_SUFFIX -r low $H264GRABBER_AUDIO -f &
- sleep 1
+ if [ "$RTSP_ALT" == "go2rtc" ]; then
+ echo "streams:" > /tmp/go2rtc.yaml
+ if [ "$RTSP_RES" == "high" ] || [ "$RTSP_RES" == "both" ]; then
+ echo " ch0_0.h264:" >> /tmp/go2rtc.yaml
+ echo " - exec:h264grabber -m h52ga -r high#backchannel=0" >> /tmp/go2rtc.yaml
fi
- $RTSP_DAEMON -m $MODEL_SUFFIX -r low $RTSP_AUDIO_COMPRESSION $RTSP_PORT $RTSP_USER $RTSP_PASSWORD $B_ONVIF_AUDIO_BC &
- elif [[ $RTSP_RES == "high" ]]; then
- if [ "$RTSP_ALT" == "yes" ]; then
- h264grabber -m $MODEL_SUFFIX -r high $H264GRABBER_AUDIO -f &
- sleep 1
+ if [ "$RTSP_RES" != "low" ] && [ "$RTSP_AUDIO_COMPRESSION" == "aac" ] ; then
+ echo " - exec:h264grabber -m $MODEL_SUFFIX -r none -a#backchannel=0" >> /tmp/go2rtc.yaml
fi
- $RTSP_DAEMON -m $MODEL_SUFFIX -r high $RTSP_AUDIO_COMPRESSION $RTSP_PORT $RTSP_USER $RTSP_PASSWORD $B_ONVIF_AUDIO_BC &
- elif [[ $RTSP_RES == "both" ]]; then
- if [ "$RTSP_ALT" == "yes" ]; then
- h264grabber -m $MODEL_SUFFIX -r both $H264GRABBER_AUDIO -f &
- sleep 1
+ if [ "$RTSP_RES" == "low" ] || [ "$RTSP_RES" == "both" ]; then
+ echo " ch0_1.h264:" >> /tmp/go2rtc.yaml
+ echo " - exec:h264grabber -m $MODEL_SUFFIX -r low#backchannel=0" >> /tmp/go2rtc.yaml
fi
- $RTSP_DAEMON -m $MODEL_SUFFIX -r both $RTSP_AUDIO_COMPRESSION $RTSP_PORT $RTSP_USER $RTSP_PASSWORD $B_ONVIF_AUDIO_BC &
- fi
+ if [ "$RTSP_RES" == "low" ] && [ "$RTSP_AUDIO_COMPRESSION" == "aac" ] ; then
+ echo " - exec:h264grabber -m $MODEL_SUFFIX -r none -a#backchannel=0" >> /tmp/go2rtc.yaml
+ fi
+
+ echo "" >> /tmp/go2rtc.yaml
+ echo "api:" >> /tmp/go2rtc.yaml
+ echo " listen: \"\"" >> /tmp/go2rtc.yaml
+ echo "" >> /tmp/go2rtc.yaml
+ echo "webrtc:" >> /tmp/go2rtc.yaml
+ echo " listen: \"\"" >> /tmp/go2rtc.yaml
+ echo "" >> /tmp/go2rtc.yaml
+ echo "rtsp:" >> /tmp/go2rtc.yaml
+ echo " listen: \":$RTSP_PORT\"" >> /tmp/go2rtc.yaml
+ if [ ! -z $USERNAME ]; then
+ echo " username: \"$USERNAME\"" >> /tmp/go2rtc.yaml
+ echo " password: \"$PASSWORD\"" >> /tmp/go2rtc.yaml
+ fi
+
+ $RTSP_DAEMON -c /tmp/go2rtc.yaml -d
+ else
- WD_COUNT=$(ps | grep wd_rtsp.sh | grep -v grep | grep -c ^)
- if [ $WD_COUNT -eq 0 ]; then
- (sleep 30; $YI_HACK_PREFIX/script/wd_rtsp.sh >/dev/null) &
+ if [[ $RTSP_RES == "low" ]]; then
+ if [ "$RTSP_ALT" == "yes" ]; then
+ h264grabber -m $MODEL_SUFFIX -r low $H264GRABBER_AUDIO -f &
+ sleep 1
+ fi
+ $RTSP_DAEMON -m $MODEL_SUFFIX -r low $RTSP_AUDIO $P_RTSP_PORT $RTSP_USER $RTSP_PASSWORD $B_ONVIF_AUDIO_BC &
+ elif [[ $RTSP_RES == "high" ]]; then
+ if [ "$RTSP_ALT" == "yes" ]; then
+ h264grabber -m $MODEL_SUFFIX -r high $H264GRABBER_AUDIO -f &
+ sleep 1
+ fi
+ $RTSP_DAEMON -m $MODEL_SUFFIX -r high $RTSP_AUDIO $P_RTSP_PORT $RTSP_USER $RTSP_PASSWORD $B_ONVIF_AUDIO_BC &
+ elif [[ $RTSP_RES == "both" ]]; then
+ if [ "$RTSP_ALT" == "yes" ]; then
+ h264grabber -m $MODEL_SUFFIX -r both $H264GRABBER_AUDIO -f &
+ sleep 1
+ fi
+ $RTSP_DAEMON -m $MODEL_SUFFIX -r both $RTSP_AUDIO $P_RTSP_PORT $RTSP_USER $RTSP_PASSWORD $B_ONVIF_AUDIO_BC &
+ fi
+
+ WD_COUNT=$(ps | grep wd_rtsp.sh | grep -v grep | grep -c ^)
+ if [ $WD_COUNT -eq 0 ]; then
+ (sleep 30; $YI_HACK_PREFIX/script/wd_rtsp.sh >/dev/null) &
+ fi
fi
}
@@ -151,13 +192,15 @@ stop_rtsp()
start_onvif()
{
- if [[ "$2" == "none" ]]; then
+ # If "null" use default
+
+ if [[ "$2" == "null" ]]; then
ONVIF_WM_SNAPSHOT=$(get_config ONVIF_WM_SNAPSHOT)
WATERMARK="&watermark="$ONVIF_WM_SNAPSHOT
elif [[ "$2" == "yes" ]]; then
WATERMARK="&watermark=yes"
fi
- if [[ "$1" == "none" ]]; then
+ if [[ "$1" == "null" ]]; then
ONVIF_PROFILE=$(get_config ONVIF_PROFILE)
elif [[ "$1" == "low" ]] || [[ "$1" == "high" ]] || [[ "$1" == "both" ]]; then
ONVIF_PROFILE=$1
@@ -289,7 +332,7 @@ stop_wsdd()
start_ftpd()
{
- if [[ "$1" == "none" ]] ; then
+ if [[ "$1" == "null" ]] ; then
if [[ $(get_config BUSYBOX_FTPD) == "yes" ]] ; then
FTPD_DAEMON="busybox"
else
@@ -308,7 +351,7 @@ start_ftpd()
stop_ftpd()
{
- if [[ "$1" == "none" ]] ; then
+ if [[ "$1" == "null" ]] ; then
if [[ $(get_config BUSYBOX_FTPD) == "yes" ]] ; then
FTPD_DAEMON="busybox"
else
@@ -335,10 +378,10 @@ ps_program()
fi
}
-NAME="none"
-ACTION="none"
-PARAM1="none"
-PARAM2="none"
+NAME="null"
+ACTION="null"
+PARAM1="null"
+PARAM2="null"
RES=""
if [ $# -lt 2 ]; then
diff --git a/src/www/httpd/htdocs/js/modules/configurations.js b/src/www/httpd/htdocs/js/modules/configurations.js
index 31ecd50..edb96a0 100644
--- a/src/www/httpd/htdocs/js/modules/configurations.js
+++ b/src/www/httpd/htdocs/js/modules/configurations.js
@@ -27,7 +27,7 @@ APP.configurations = (function($) {
$.each(response, function(key, state) {
if (key == "HOSTNAME" || key == "TIMEZONE" || key == "TIMELAPSE_DT" || key == "TIMELAPSE_VDT" || key == "NTP_SERVER" || key == "HTTPD_PORT" || key == "RTSP_PORT" || key == "USERNAME")
$('input[type="text"][data-key="' + key + '"]').prop('value', state);
- else if (key == "RTSP_STREAM" || key == "RTSP_AUDIO" || key=="SWAP_SWAPPINESS")
+ else if (key == "RTSP_ALT" || key == "RTSP_STREAM" || key == "RTSP_AUDIO" || key=="SWAP_SWAPPINESS")
$('select[data-key="' + key + '"]').prop('value', state);
else if (key == "PASSWORD" || key == "SSH_PASSWORD")
$('input[type="password"][data-key="' + key + '"]').prop('value', state);
@@ -69,6 +69,7 @@ APP.configurations = (function($) {
configs["TIMELAPSE_VDT"] = $('input[type="text"][data-key="TIMELAPSE_VDT"]').prop('value');
configs["NTP_SERVER"] = $('input[type="text"][data-key="NTP_SERVER"]').prop('value');
configs["HTTPD_PORT"] = $('input[type="text"][data-key="HTTPD_PORT"]').prop('value');
+ configs["RTSP_ALT"] = $('select[data-key="RTSP_ALT"]').prop('value');
configs["RTSP_STREAM"] = $('select[data-key="RTSP_STREAM"]').prop('value');
configs["RTSP_AUDIO"] = $('select[data-key="RTSP_AUDIO"]').prop('value');
configs["RTSP_PORT"] = $('input[type="text"][data-key="RTSP_PORT"]').prop('value');
diff --git a/src/www/httpd/htdocs/pages/configurations.html b/src/www/httpd/htdocs/pages/configurations.html
index 0cf258a..6cdcaab 100644
--- a/src/www/httpd/htdocs/pages/configurations.html
+++ b/src/www/httpd/htdocs/pages/configurations.html
@@ -76,15 +76,19 @@ General
- | RTSP alternative server |
+ RTSP server program |
-
+
+
+
- Use alternative RTSP server instead of the live555 based one.
+ Select the RTSP server program.
+
+ Note: go2rtc supports only aac audio.
|