capture timestamped audio (ros ticket #20) about audio_common HOT 3 OPEN

# HG changeset patch
# User Mike Bosse <[email protected]>
# Date 2012-10-30 12:42:08 +1000
# Node ID 41867e966e88da6df23521e150724be6901e0197
# Parent  13166e69e1eef5c96b9f8c9ec3f2040b1389805a
added message and node to capture timestamped audio data in raw format

diff --git a/audio_capture/CMakeLists.txt b/audio_capture/CMakeLists.txt
--- a/audio_capture/CMakeLists.txt
+++ b/audio_capture/CMakeLists.txt
@@ -23,4 +23,6 @@
 include_directories(${GST_INCLUDE_DIRS})
 rosbuild_add_executable(audio_capture src/audio_capture.cpp)
 target_link_libraries(audio_capture ${GST_LIBRARIES})
+rosbuild_add_executable(audiostampedraw_capture src/audiostampedraw_capture.cpp)
+target_link_libraries(audiostampedraw_capture ${GST_LIBRARIES})
 rosbuild_link_boost(audio_capture thread)
diff --git a/audio_capture/launch/capture_stampedraw.launch b/audio_capture/launch/capture_stampedraw.launch
new file mode 100644
--- /dev/null
+++ b/audio_capture/launch/capture_stampedraw.launch
@@ -0,0 +1,6 @@
+<launch>
+
+  <node name="audio_capture" pkg="audio_capture" type="audiostampedraw_capture" output="screen">
+  </node>
+
+</launch>
diff --git a/audio_capture/src/audiostampedraw_capture.cpp b/audio_capture/src/audiostampedraw_capture.cpp
new file mode 100644
--- /dev/null
+++ b/audio_capture/src/audiostampedraw_capture.cpp
@@ -0,0 +1,145 @@
+#include <stdio.h>
+#include <gst/gst.h>
+#include <gst/app/gstappsink.h>
+#include <boost/thread.hpp>
+
+#include <ros/ros.h>
+
+#include "audio_common_msgs/AudioDataStampedRaw.h"
+
+namespace audio_transport
+{
+  class RosGstCapture
+  {
+    public: 
+      RosGstCapture()
+      {
+        _bitrate = 192;
+
+        std::string dst_type;
+
+        // The bitrate at which to encode the audio
+        ros::param::param<int>("~bitrate", _bitrate, 192);
+
+        // 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::AudioDataStampedRaw>("audio", 10, true);
+
+        _loop = g_main_loop_new(NULL, false);
+        _pipeline = gst_pipeline_new("ros_pipeline");
+
+        // 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");
+
+#if 0
+        _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);
+        gst_element_link_many(_source, _convert, _encode, _sink, NULL);
+#else
+   // remove _encode so that we have raw audio
+        gst_bin_add_many( GST_BIN(_pipeline), _source, _convert, _sink, NULL);
+        gst_element_link_many(_source, _convert, _sink, NULL);
+
+   GstCaps *caps = gst_caps_new_simple("audio/x-raw-int",
+                       "channels",G_TYPE_INT,1,
+                       "width",G_TYPE_INT,8,
+                       "depth",G_TYPE_INT,8,
+                       NULL);
+
+   g_object_set( G_OBJECT(_sink), "caps", caps, NULL);
+
+   gst_caps_unref(caps);
+#endif
+        /*}
+        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);
+        }
+        */
+
+   // setup the pipeline to use the system clock in real time mode
+   _sysclock = gst_system_clock_obtain();
+   g_object_set( G_OBJECT(_sysclock), "clock-type", GST_CLOCK_TYPE_REALTIME, NULL);
+   gst_pipeline_use_clock(GST_PIPELINE(_pipeline),_sysclock);
+
+        gst_element_set_state(GST_ELEMENT(_pipeline), GST_STATE_PLAYING);
+   //_sysclock = gst_pipeline_get_clock(GST_PIPELINE(_pipeline));
+
+        _gst_thread = boost::thread( boost::bind(g_main_loop_run, _loop) );
+      }
+
+      void publish( const audio_common_msgs::AudioDataStampedRaw &msg )
+      {
+           _pub.publish(msg);
+      }
+
+      static GstFlowReturn onNewBuffer (GstAppSink *appsink, gpointer userData)
+      {
+        RosGstCapture *server = reinterpret_cast<RosGstCapture*>(userData);
+        audio_common_msgs::AudioDataStampedRaw msg;
+
+        GstBuffer *buffer;
+        g_signal_emit_by_name(appsink, "pull-buffer", &buffer);
+
+   msg.header.stamp = ros::Time::now().fromNSec(buffer->timestamp + gst_element_get_base_time(GST_ELEMENT(server->_pipeline)) );
+
+        msg.data.resize( buffer->size );
+        memcpy( &msg.data[0], buffer->data, buffer->size);
+
+        server->publish(msg);
+
+   gst_buffer_unref(buffer);
+
+        return GST_FLOW_OK;
+      }
+
+    private:
+      ros::NodeHandle _nh;
+      ros::Publisher _pub;
+
+      boost::thread _gst_thread;
+
+      GstElement *_pipeline, *_source, *_sink, *_convert, *_encode;
+      GstClock *_sysclock;
+      GMainLoop *_loop;
+      int _bitrate;
+  };
+}
+
+int main (int argc, char **argv)
+{
+  ros::init(argc, argv, "audio_capture");
+  gst_init(&argc, &argv);
+
+  audio_transport::RosGstCapture server;
+  ros::spin();
+}
diff --git a/audio_common_msgs/msg/AudioDataStampedRaw.msg b/audio_common_msgs/msg/AudioDataStampedRaw.msg
new file mode 100644
--- /dev/null
+++ b/audio_common_msgs/msg/AudioDataStampedRaw.msg
@@ -0,0 +1,2 @@
+Header header
+uint8[] data