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Added volume control parameter at audio_capture/play #258

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04db285
Update audio_capture.cpp
BeomsuHa Apr 8, 2025
7242888
Update audio_play.cpp
BeomsuHa Apr 8, 2025
8982632
Update play.launch
BeomsuHa Apr 8, 2025
cbb680d
Update capture.launch
BeomsuHa Apr 8, 2025
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3 changes: 3 additions & 0 deletions audio_capture/launch/capture.launch
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Original file line number Diff line number Diff line change
Expand Up @@ -12,6 +12,8 @@
<arg name="sample_format" default="S16LE"/>
<arg name="ns" default="audio"/>
<arg name="audio_topic" default="audio"/>
<arg name="capture_volume" default="50"/>


<group ns="$(arg ns)">
<node name="audio_capture" pkg="audio_capture" type="audio_capture" output="screen">
Expand All @@ -24,6 +26,7 @@
<param name="depth" value="$(arg depth)"/>
<param name="sample_rate" value="$(arg sample_rate)"/>
<param name="sample_format" value="$(arg sample_format)"/>
<param name="capture_volume" value="$(arg capture_volume)"/>
</node>
</group>

Expand Down
66 changes: 58 additions & 8 deletions audio_capture/src/audio_capture.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -17,6 +17,7 @@ namespace audio_transport
RosGstCapture()
{
_bitrate = 192;
_capture_volume = 50; // Default volume 50%

std::string dst_type;

Expand All @@ -31,6 +32,11 @@ namespace audio_transport
ros::param::param<int>("~channels", _channels, 1);
ros::param::param<int>("~depth", _depth, 16);
ros::param::param<int>("~sample_rate", _sample_rate, 16000);

// The volume level for audio capture (0-100%)
ros::param::param<int>("~capture_volume", _capture_volume, 50);
// Ensure volume is within valid range
_capture_volume = std::max(0, std::min(_capture_volume, 100));

// The destination of the audio
ros::param::param<std::string>("~dst", dst_type, "appsink");
Expand Down Expand Up @@ -83,6 +89,18 @@ namespace audio_transport
// ghcar *gst_device = device.c_str();
g_object_set(G_OBJECT(_source), "device", device.c_str(), NULL);
}

// Create volume control element
_volume = gst_element_factory_make("volume", "volume");
if (!_volume) {
ROS_ERROR_STREAM("Failed to create volume element");
exitOnMainThread(1);
}

// Set volume level (0-100% mapped to 0.0-1.0)
double volume_value = static_cast<double>(_capture_volume) / 100.0;
g_object_set(G_OBJECT(_volume), "volume", volume_value, NULL);
ROS_INFO("Setting capture volume to %d%% (%.2f)", _capture_volume, volume_value);

GstCaps *caps;
caps = gst_caps_new_simple("audio/x-raw",
Expand Down Expand Up @@ -114,18 +132,21 @@ namespace audio_transport
g_object_set( G_OBJECT(_encode), "target", 1, NULL);
g_object_set( G_OBJECT(_encode), "bitrate", _bitrate, NULL);

gst_bin_add_many( GST_BIN(_pipeline), _source, _filter, _convert, _encode, _sink, NULL);
link_ok = gst_element_link_many(_source, _filter, _convert, _encode, _sink, NULL);
// Add volume element to pipeline
gst_bin_add_many( GST_BIN(_pipeline), _source, _filter, _volume, _convert, _encode, _sink, NULL);
link_ok = gst_element_link_many(_source, _filter, _volume, _convert, _encode, _sink, NULL);
} else if (_format == "wave") {
if (dst_type == "appsink") {
g_object_set( G_OBJECT(_sink), "caps", caps, NULL);
gst_caps_unref(caps);
gst_bin_add_many( GST_BIN(_pipeline), _source, _sink, NULL);
link_ok = gst_element_link_many( _source, _sink, NULL);
// Add volume element to pipeline
gst_bin_add_many( GST_BIN(_pipeline), _source, _volume, _sink, NULL);
link_ok = gst_element_link_many( _source, _volume, _sink, NULL);
} else {
_filter = gst_element_factory_make("wavenc", "filter");
gst_bin_add_many( GST_BIN(_pipeline), _source, _filter, _sink, NULL);
link_ok = gst_element_link_many( _source, _filter, _sink, NULL);
// Add volume element to pipeline
gst_bin_add_many( GST_BIN(_pipeline), _source, _volume, _filter, _sink, NULL);
link_ok = gst_element_link_many( _source, _volume, _filter, _sink, NULL);
}
} else {
ROS_ERROR_STREAM("format must be \"wave\" or \"mp3\"");
Expand Down Expand Up @@ -159,6 +180,9 @@ namespace audio_transport
info_msg.bitrate = _bitrate;
info_msg.coding_format = _format;
_pub_info.publish(info_msg);

// Start timer to check for parameter updates
_timer = _nh.createTimer(ros::Duration(1.0), &RosGstCapture::parameterCallback, this);
}

~RosGstCapture()
Expand All @@ -168,6 +192,30 @@ namespace audio_transport
gst_object_unref(_pipeline);
g_main_loop_unref(_loop);
}

// Check for volume parameter updates
void parameterCallback(const ros::TimerEvent& event)
{
int current_volume;
if(ros::param::get("~capture_volume", current_volume))
{
if(current_volume != _capture_volume)
{
_capture_volume = std::max(0, std::min(current_volume, 100));
updateVolume();
}
}
}

// Update volume in GStreamer pipeline
void updateVolume()
{
if(_volume) {
double volume_value = static_cast<double>(_capture_volume) / 100.0;
ROS_INFO("Updating capture volume to %d%% (%.2f)", _capture_volume, volume_value);
g_object_set(G_OBJECT(_volume), "volume", volume_value, NULL);
}
}

void exitOnMainThread(int code)
{
Expand Down Expand Up @@ -242,12 +290,14 @@ namespace audio_transport
ros::Publisher _pub;
ros::Publisher _pub_stamped;
ros::Publisher _pub_info;
ros::Subscriber _volume_sub;
ros::Timer _timer;

boost::thread _gst_thread;

GstElement *_pipeline, *_source, *_filter, *_sink, *_convert, *_encode;
GstElement *_pipeline, *_source, *_filter, *_sink, *_convert, *_encode, *_volume;
GstBus *_bus;
int _bitrate, _channels, _depth, _sample_rate;
int _bitrate, _channels, _depth, _sample_rate, _capture_volume;
GMainLoop *_loop;
std::string _format, _sample_format;
};
Expand Down
3 changes: 3 additions & 0 deletions audio_play/launch/play.launch
View file
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Original file line number Diff line number Diff line change
Expand Up @@ -12,6 +12,8 @@
<arg name="sample_format" default="S16LE"/>
<arg name="ns" default="audio"/>
<arg name="audio_topic" default="audio" />
<arg name="play_volume" default="50"/>


<group ns="$(arg ns)">
<node name="audio_play" pkg="audio_play" type="audio_play" output="screen">
Expand All @@ -24,6 +26,7 @@
<param name="depth" value="$(arg depth)"/>
<param name="sample_rate" value="$(arg sample_rate)"/>
<param name="sample_format" value="$(arg sample_format)"/>
<param name="play_volume" value="$(arg play_volume)"/>
</node>
</group>
</launch>
62 changes: 59 additions & 3 deletions audio_play/src/audio_play.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -33,6 +33,10 @@ namespace audio_transport
ros::param::param<int>("~depth", depth, 16);
ros::param::param<int>("~sample_rate", sample_rate, 16000);
ros::param::param<std::string>("~sample_format", sample_format, "S16LE");

// The volume level for audio playback (0-100%)
ros::param::param<int>("~play_volume", _play_volume, 50);
_play_volume = std::max(0, std::min(_play_volume, 100));

_sub = _nh.subscribe("audio", 10, &RosGstPlay::onAudio, this);

Expand Down Expand Up @@ -61,14 +65,28 @@ namespace audio_transport
_convert = gst_element_factory_make("audioconvert", "convert");
audiopad = gst_element_get_static_pad(_convert, "sink");
_resample = gst_element_factory_make("audioresample", "resample");

// Create volume control element
_volume = gst_element_factory_make("volume", "volume");
if (!_volume) {
ROS_ERROR_STREAM("Failed to create volume element");
return;
}

// Set volume level (0-100% mapped to 0.0-1.0)
double volume_value = static_cast<double>(_play_volume) / 100.0;
g_object_set(G_OBJECT(_volume), "volume", volume_value, NULL);
ROS_INFO("Setting play volume to %d%% (%.2f)", _play_volume, volume_value);

_sink = gst_element_factory_make("alsasink", "sink");
g_object_set(G_OBJECT(_sink), "sync", FALSE, NULL);
if (!device.empty()) {
g_object_set(G_OBJECT(_sink), "device", device.c_str(), NULL);
}
gst_bin_add_many( GST_BIN(_audio), _convert, _resample, _sink, NULL);
gst_element_link_many(_convert, _resample, _sink, NULL);

// Add volume element to pipeline
gst_bin_add_many( GST_BIN(_audio), _convert, _resample, _volume, _sink, NULL);
gst_element_link_many(_convert, _resample, _volume, _sink, NULL);
gst_element_add_pad(_audio, gst_ghost_pad_new("sink", audiopad));
}
else
Expand Down Expand Up @@ -126,6 +144,41 @@ namespace audio_transport
//gst_element_set_state(GST_ELEMENT(_playbin), GST_STATE_PLAYING);

_gst_thread = boost::thread( boost::bind(g_main_loop_run, _loop) );

// Start timer to check for parameter updates
_timer = _nh.createTimer(ros::Duration(1.0), &RosGstPlay::parameterCallback, this);
}

~RosGstPlay()
{
g_main_loop_quit(_loop);
gst_element_set_state(_pipeline, GST_STATE_NULL);
gst_object_unref(_pipeline);
g_main_loop_unref(_loop);
}

// Check for volume parameter updates
void parameterCallback(const ros::TimerEvent& event)
{
int current_volume;
if(ros::param::get("~play_volume", current_volume))
{
if(current_volume != _play_volume)
{
_play_volume = std::max(0, std::min(current_volume, 100));
updateVolume();
}
}
}

// Update volume in GStreamer pipeline
void updateVolume()
{
if(_volume) {
double volume_value = static_cast<double>(_play_volume) / 100.0;
ROS_INFO("Updating play volume to %d%% (%.2f)", _play_volume, volume_value);
g_object_set(G_OBJECT(_volume), "volume", volume_value, NULL);
}
}

private:
Expand Down Expand Up @@ -176,11 +229,14 @@ namespace audio_transport

ros::NodeHandle _nh;
ros::Subscriber _sub;
ros::Subscriber _volume_sub;
ros::Timer _timer;
boost::thread _gst_thread;

GstElement *_pipeline, *_source, *_sink, *_decoder, *_convert, *_audio, *_resample, *_filter;
GstElement *_pipeline, *_source, *_sink, *_decoder, *_convert, *_audio, *_resample, *_filter, *_volume;
GstElement *_playbin;
GMainLoop *_loop;
int _play_volume; // Volume control (0-100%)
};
}

Expand Down