Merge pull request #28 from fengqikai1414/master

add image_common, audio_common and vision_opencv, refer to ros-indigo

ros/audio_common/.gitignore

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+*.pyc

ros/audio_common/audio_capture/.gitignore

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+build

ros/audio_common/audio_capture/CHANGELOG.rst

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+^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
+Changelog for package audio_capture
+^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
+
+0.2.12 (2016-02-29)
+-------------------
+
+0.2.11 (2016-02-16)
+-------------------
+
+0.2.10 (2016-01-21)
+-------------------
+
+0.2.9 (2015-12-02)
+------------------
+* [audio_capture] add error handler
+* [audio_capture] add option to publish captured audio data as wav format
+* Fixed memory leak (see `#18 <https://github.com/ros-drivers/audio_common/issues/18>`_).
+* Removed trailing whitespace.
+* Contributors: Felix Duvallet, Furushchev
+
+0.2.8 (2015-10-02)
+------------------
+* Update maintainer email
+* Contributors: trainman419
+
+0.2.7 (2014-07-25)
+------------------
+* audio_capture.cpp has to wait for generated AudioData headers
+* Contributors: v4hn
+
+0.2.6 (2014-02-26)
+------------------
+* audio_capture and play _require\_ gstreamer, it's not optional
+* Contributors: v4hn
+
+0.2.5 (2014-01-23)
+------------------
+* "0.2.5"
+* Contributors: trainman419
+
+0.2.4 (2013-09-10)
+------------------
+* Update CMakeLists.txt
+* audio_capture: install launchfiles
+* Contributors: David Gossow
+
+0.2.3 (2013-07-15)
+------------------
+* Fix install rule for audio_capture.
+* Contributors: Austin Hendrix
+
+0.2.2 (2013-04-10)
+------------------
+
+0.2.1 (2013-04-08 13:59)
+------------------------
+
+0.2.0 (2013-04-08 13:49)
+------------------------
+* Finish catkinizing audio_common.
+* Catkinize audio_play.
+* Catkinize audio_capture.
+* Fix typo in package.xml
+* Versions and more URLs.
+* Convert manifests to package.xml
+* Convert audio_capture manifest to package.xml
+* Ditch old makefiles.
+* Updates manifest
+* Updated manifests for rodep2
+* oneiric build fixes, bump version to 0.1.6
+* Removed redundant thread::thread
+* Added a rosdep.yaml file
+* Fixed to use audio_common_msgs
+* Added ability to use different festival voices
+* Updated documentation
+* Added ability to capture to file
+* Fixed ignore files
+* Added hgignore files
+* Audio_capture and audio_play working
+* Making separate audio_capture and audio_play packages
+* Moved audio_transport to audio_capture
+* Contributors: Austin Hendrix, Brian Gerkey, Nate Koenig, nkoenig

ros/audio_common/audio_capture/CMakeLists.txt

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+cmake_minimum_required(VERSION 2.8.3)
+
+project(audio_capture)
+
+find_package(catkin REQUIRED COMPONENTS roscpp audio_common_msgs)
+
+find_package(PkgConfig)
+pkg_check_modules(GST gstreamer-0.10 REQUIRED)
+
+find_package(Boost REQUIRED COMPONENTS thread)
+
+include_directories(${catkin_INCLUDE_DIRS} ${Boost_INCLUDE_DIRS} ${GST_INCLUDE_DIRS})
+
+catkin_package()
+
+add_executable(audio_capture src/audio_capture.cpp)
+target_link_libraries(audio_capture ${catkin_LIBRARIES} ${GST_LIBRARIES} ${Boost_LIBRARIES})
+add_dependencies(audio_capture ${catkin_EXPORTED_TARGETS})
+
+install(TARGETS audio_capture
+   DESTINATION ${CATKIN_PACKAGE_BIN_DESTINATION})
+
+install(DIRECTORY launch
+   DESTINATION ${CATKIN_PACKAGE_SHARE_DESTINATION})

ros/audio_common/audio_capture/launch/capture.launch

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+<launch>
+
+  <node name="audio_capture" pkg="audio_capture" type="audio_capture" output="screen">
+    <param name="bitrate" value="128"/>
+  </node>
+
+</launch>

ros/audio_common/audio_capture/launch/capture_to_file.launch

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+<launch>
+
+  <node name="audio_capture" pkg="audio_capture" type="audio_capture" output="screen">
+    <param name="bitrate" value="128"/>
+    <param name="dst" value="output.mp3"/>
+  </node>
+
+</launch>

ros/audio_common/audio_capture/launch/capture_wave.launch

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+<launch>
+  <!-- publish audio data as wav format -->
+  <node name="audio_capture" pkg="audio_capture" type="audio_capture" output="screen">
+    <param name="format" value="wave" />
+    <param name="channels" value="1" />
+    <param name="depth" value="16" />
+    <param name="sample_rate" value="16000" />
+  </node>
+</launch>

ros/audio_common/audio_capture/mainpage.dox

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+/**
+\mainpage
+\htmlinclude manifest.html
+
+\b audio_capture is a package that records audio from a microphone and makes it available to other ROS nodes.
+
+\section codeapi Code API
+
+<!--
+Provide links to specific auto-generated API documentation within your
+package that is of particular interest to a reader. Doxygen will
+document pretty much every part of your code, so do your best here to
+point the reader to the actual API.
+
+If your codebase is fairly large or has different sets of APIs, you
+should use the doxygen 'group' tag to keep these APIs together. For
+example, the roscpp documentation has 'libros' group.
+-->
+
+
+*/

ros/audio_common/audio_capture/package.xml

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+<package>
+   <name>audio_capture</name>
+   <version>0.2.12</version>
+   <description>
+      Transports audio from a source to a destination. Audio sources can come
+      from a microphone or file. The destination can play the audio or save it
+      to an mp3 file.
+   </description>
+   <maintainer email="[email protected]">Austin Hendrix</maintainer>
+   <author>Nate Koenig</author>
+   <license>BSD</license>
+   <url type="website">http://ros.org/wiki/audio_capture</url>
+   <url type="repository">https://github.com/ros-drivers/audio_common</url>
+   <url type="bugtracker">https://github.com/ros-drivers/audio_common/issues</url>
+
+   <buildtool_depend>catkin</buildtool_depend>
+
+   <build_depend>roscpp</build_depend>
+   <build_depend>audio_common_msgs</build_depend>
+   <build_depend>libgstreamer0.10-dev</build_depend>
+   <build_depend>libgstreamer-plugins-base0.10-dev</build_depend>
+
+   <run_depend>roscpp</run_depend>
+   <run_depend>audio_common_msgs</run_depend>
+   <run_depend>libgstreamer0.10-0</run_depend>
+   <run_depend>libgstreamer-plugins-base0.10-0</run_depend>
+   <run_depend>gstreamer0.10-plugins-ugly</run_depend>
+   <run_depend>gstreamer0.10-plugins-good</run_depend>
+</package>

ros/audio_common/audio_capture/src/audio_capture.cpp

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+#include <stdio.h>
+#include <gst/gst.h>
+#include <gst/app/gstappsink.h>
+#include <boost/thread.hpp>
+
+#include <ros/ros.h>
+
+#include "audio_common_msgs/AudioData.h"
+
+namespace audio_transport
+{
+  class RosGstCapture
+  {
+    public:
+      RosGstCapture()
+      {
+        _bitrate = 192;
+
+        std::string dst_type;
+
+        // Need to encoding or publish raw wave data
+        ros::param::param<std::string>("~format", _format, "mp3");
+
+        // The bitrate at which to encode the audio
+        ros::param::param<int>("~bitrate", _bitrate, 192);
+
+        // only available for raw data
+        ros::param::param<int>("~channels", _channels, 1);
+        ros::param::param<int>("~depth", _depth, 16);
+        ros::param::param<int>("~sample_rate", _sample_rate, 16000);
+
+        // The destination of the audio
+        ros::param::param<std::string>("~dst", dst_type, "appsink");
+
+        // The source of the audio
+        //ros::param::param<std::string>("~src", source_type, "alsasrc");
+
+        _pub = _nh.advertise<audio_common_msgs::AudioData>("audio", 10, true);
+
+        _loop = g_main_loop_new(NULL, false);
+        _pipeline = gst_pipeline_new("ros_pipeline");
+        _bus = gst_pipeline_get_bus(GST_PIPELINE(_pipeline));
+        gst_bus_add_signal_watch(_bus);
+        g_signal_connect(_bus, "message::error",
+                         G_CALLBACK(onMessage), this);
+        g_object_unref(_bus);
+
+        // We create the sink first, just for convenience
+        if (dst_type == "appsink")
+        {
+          _sink = gst_element_factory_make("appsink", "sink");
+          g_object_set(G_OBJECT(_sink), "emit-signals", true, NULL);
+          g_object_set(G_OBJECT(_sink), "max-buffers", 100, NULL);
+          g_signal_connect( G_OBJECT(_sink), "new-buffer",
+                            G_CALLBACK(onNewBuffer), this);
+        }
+        else
+        {
+          printf("file sink\n");
+          _sink = gst_element_factory_make("filesink", "sink");
+          g_object_set( G_OBJECT(_sink), "location", dst_type.c_str(), NULL);
+        }
+
+        _source = gst_element_factory_make("alsasrc", "source");
+        _convert = gst_element_factory_make("audioconvert", "convert");
+
+        gboolean link_ok;
+
+        if (_format == "mp3"){
+          _encode = gst_element_factory_make("lame", "encoder");
+          g_object_set( G_OBJECT(_encode), "preset", 1001, NULL);
+          g_object_set( G_OBJECT(_encode), "bitrate", _bitrate, NULL);
+
+          gst_bin_add_many( GST_BIN(_pipeline), _source, _convert, _encode, _sink, NULL);
+          link_ok = gst_element_link_many(_source, _convert, _encode, _sink, NULL);
+        } else if (_format == "wave") {
+          GstCaps *caps;
+          caps = gst_caps_new_simple("audio/x-raw-int",
+                                     "channels", G_TYPE_INT, _channels,
+                                     "width",    G_TYPE_INT, _depth,
+                                     "depth",    G_TYPE_INT, _depth,
+                                     "rate",     G_TYPE_INT, _sample_rate,
+                                     "signed",   G_TYPE_BOOLEAN, TRUE,
+                                     NULL);
+
+          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);
+        } else {
+          ROS_ERROR_STREAM("format must be \"wave\" or \"mp3\"");
+          exitOnMainThread(1);
+        }
+        /*}
+        else
+        {
+          _sleep_time = 10000;
+          _source = gst_element_factory_make("filesrc", "source");
+          g_object_set(G_OBJECT(_source), "location", source_type.c_str(), NULL);
+
+          gst_bin_add_many( GST_BIN(_pipeline), _source, _sink, NULL);
+          gst_element_link_many(_source, _sink, NULL);
+        }
+        */
+
+        if (!link_ok) {
+          ROS_ERROR_STREAM("Unsupported media type.");
+          exitOnMainThread(1);
+        }
+
+        gst_element_set_state(GST_ELEMENT(_pipeline), GST_STATE_PLAYING);
+
+        _gst_thread = boost::thread( boost::bind(g_main_loop_run, _loop) );
+      }
+
+      ~RosGstCapture()
+      {
+        g_main_loop_quit(_loop);
+        gst_element_set_state(_pipeline, GST_STATE_NULL);
+        gst_object_unref(_pipeline);
+        g_main_loop_unref(_loop);
+      }
+
+      void exitOnMainThread(int code)
+      {
+        exit(code);
+      }
+
+      void publish( const audio_common_msgs::AudioData &msg )
+      {
+        _pub.publish(msg);
+      }
+
+      static GstFlowReturn onNewBuffer (GstAppSink *appsink, gpointer userData)
+      {
+        RosGstCapture *server = reinterpret_cast<RosGstCapture*>(userData);
+
+        GstBuffer *buffer;
+        g_signal_emit_by_name(appsink, "pull-buffer", &buffer);
+
+        audio_common_msgs::AudioData msg;
+        msg.data.resize( buffer->size );
+        memcpy( &msg.data[0], buffer->data, buffer->size);
+
+        server->publish(msg);
+
+        return GST_FLOW_OK;
+      }
+
+      static gboolean onMessage (GstBus *bus, GstMessage *message, gpointer userData)
+      {
+        RosGstCapture *server = reinterpret_cast<RosGstCapture*>(userData);
+        GError *err;
+        gchar *debug;
+
+        gst_message_parse_error(message, &err, &debug);
+        ROS_ERROR_STREAM("gstreamer: " << err->message);
+        g_error_free(err);
+        g_free(debug);
+        g_main_loop_quit(server->_loop);
+        server->exitOnMainThread(1);
+        return FALSE;
+      }
+
+    private:
+      ros::NodeHandle _nh;
+      ros::Publisher _pub;
+
+      boost::thread _gst_thread;
+
+      GstElement *_pipeline, *_source, *_sink, *_convert, *_encode;
+      GstBus *_bus;
+      int _bitrate, _channels, _depth, _sample_rate;
+      GMainLoop *_loop;
+      std::string _format;
+  };
+}
+
+int main (int argc, char **argv)
+{
+  ros::init(argc, argv, "audio_capture");
+  gst_init(&argc, &argv);
+
+  audio_transport::RosGstCapture server;
+  ros::spin();
+}