adding record functionality

1 files changed, 235 insertions, 0 deletions

diff --git a/utils/grecord.py b/utils/grecord.py
new file mode 100644
index 0000000..3a928f3
--- /dev/null
+++ b/utils/grecord.py
@@ -0,0 +1,235 @@
+#Copyright (c) 2008, Media Modifications Ltd.
+#Copyright (c) 2011, Walter Bender
+
+#Permission is hereby granted, free of charge, to any person obtaining a copy
+#of this software and associated documentation files (the "Software"), to deal
+#in the Software without restriction, including without limitation the rights
+#to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+#copies of the Software, and to permit persons to whom the Software is
+#furnished to do so, subject to the following conditions:
+
+#The above copyright notice and this permission notice shall be included in
+#all copies or substantial portions of the Software.
+
+#THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+#IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+#FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+#AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+#LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+#OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+#THE SOFTWARE.
+
+import os
+from gettext import gettext as _
+import time
+
+import gtk
+import gst
+import pygst
+import gobject
+gobject.threads_init()
+
+
+class Grecord:
+
+    def __init__(self, parent, output_file='output'):
+        self._activity = parent
+        self._eos_cb = None
+
+        self._can_limit_framerate = False
+        self._playing = False
+
+        self._audio_transcode_handler = None
+        self._transcode_id = None
+
+        self._audio_pixbuf = None
+
+        self._pipeline = gst.Pipeline("Record")
+        self._create_audiobin(output_file)
+
+        bus = self._pipeline.get_bus()
+        bus.add_signal_watch()
+        bus.connect('message', self._bus_message_handler)
+
+    def _create_audiobin(self, output_file='output'):
+        src = gst.element_factory_make("alsasrc", "absrc")
+
+        # attempt to use direct access to the 0,0 device, solving some A/V
+        # sync issues
+        src.set_property("device", "plughw:0,0")
+        hwdev_available = src.set_state(gst.STATE_PAUSED) != \
+                          gst.STATE_CHANGE_FAILURE
+        src.set_state(gst.STATE_NULL)
+        if not hwdev_available:
+            src.set_property("device", "default")
+
+        srccaps = gst.Caps("audio/x-raw-int,rate=16000,channels=1,depth=16")
+
+        # guarantee perfect stream, important for A/V sync
+        rate = gst.element_factory_make("audiorate")
+
+        # without a buffer here, gstreamer struggles at the start of the
+        # recording and then the A/V sync is bad for the whole video
+        # (possibly a gstreamer/ALSA bug -- even if it gets caught up, it
+        # should be able to resync without problem)
+        queue = gst.element_factory_make("queue", "audioqueue")
+        queue.set_property("leaky", True) # prefer fresh data
+        queue.set_property("max-size-time", 5000000000) # 5 seconds
+        queue.set_property("max-size-buffers", 500)
+        queue.connect("overrun", self._log_queue_overrun)
+
+        enc = gst.element_factory_make("wavenc", "abenc")
+
+        sink = gst.element_factory_make("filesink", "absink")
+        sink.set_property("location",
+            os.path.join(self._activity.datapath, output_file + '.wav'))
+
+        self._audiobin = gst.Bin("audiobin")
+        self._audiobin.add(src, rate, queue, enc, sink)
+
+        src.link(rate, srccaps)
+        gst.element_link_many(rate, queue, enc, sink)
+
+    def _log_queue_overrun(self, queue):
+        cbuffers = queue.get_property("current-level-buffers")
+        cbytes = queue.get_property("current-level-bytes")
+        ctime = queue.get_property("current-level-time")
+        # logger.error("Buffer overrun in %s (%d buffers, %d bytes, %d time)"
+        #    % (queue.get_name(), cbuffers, cbytes, ctime))
+ 
+    def play(self, use_xv=True):
+        if self._get_state() == gst.STATE_PLAYING:
+            return
+
+        self._pipeline.set_state(gst.STATE_PLAYING)
+        self._playing = True
+
+    def pause(self):
+        self._pipeline.set_state(gst.STATE_PAUSED)
+        self._playing = False
+
+    def stop(self):
+        self._pipeline.set_state(gst.STATE_NULL)
+        self._playing = False
+
+    def is_playing(self):
+        return self._playing
+
+    def _get_state(self):
+        return self._pipeline.get_state()[1]
+
+    def stop_recording_audio(self, output_file='output'):
+        # We should be able to simply pause and remove the audiobin, but
+        # this seems to cause a gstreamer segfault. So we stop the whole
+        # pipeline while manipulating it.
+        # http://dev.laptop.org/ticket/10183
+        self._pipeline.set_state(gst.STATE_NULL)
+        self._pipeline.remove(self._audiobin)
+        self.play()
+
+        if not self._audio_pixbuf:
+            # FIXME: inform model of failure?
+            return
+
+        audio_path = os.path.join(self._activity.datapath, output_file + '.wav')
+        if not os.path.exists(audio_path) or os.path.getsize(audio_path) <= 0:
+            # FIXME: inform model of failure?
+            return
+
+        line = 'filesrc location=' + audio_path + ' name=audioFilesrc ! wavparse name=audioWavparse ! audioconvert name=audioAudioconvert ! vorbisenc name=audioVorbisenc ! oggmux name=audioOggmux ! filesink name=audioFilesink'
+        audioline = gst.parse_launch(line)
+
+        vorbis_enc = audioline.get_by_name('audioVorbisenc')
+
+        audioFilesink = audioline.get_by_name('audioFilesink')
+        audioOggFilepath = os.path.join(self._activity.datapath,
+                                        output_file + '.ogg')
+        audioFilesink.set_property("location", audioOggFilepath)
+
+        audioBus = audioline.get_bus()
+        audioBus.add_signal_watch()
+        self._audio_transcode_handler = audioBus.connect(
+            'message', self._onMuxedAudioMessageCb, audioline, output_file)
+        self._transcode_id = gobject.timeout_add(200, self._transcodeUpdateCb, audioline)
+        audioline.set_state(gst.STATE_PLAYING)
+
+    def blockedCb(self, x, y, z):
+        pass
+
+    def record_audio(self):
+        self._audio_pixbuf = None
+
+        # we should be able to add the audiobin on the fly, but unfortunately
+        # this results in several seconds of silence being added at the start
+        # of the recording. So we stop the whole pipeline while adjusting it.
+        # SL#2040
+        self._pipeline.set_state(gst.STATE_NULL)
+        self._pipeline.add(self._audiobin)
+        self.play()
+
+    def _transcodeUpdateCb( self, pipe ):
+        position, duration = self._query_position( pipe )
+        if position != gst.CLOCK_TIME_NONE:
+            value = position * 100.0 / duration
+            value = value/100.0
+        return True
+
+    def _query_position(self, pipe):
+        try:
+            position, format = pipe.query_position(gst.FORMAT_TIME)
+        except:
+            position = gst.CLOCK_TIME_NONE
+
+        try:
+            duration, format = pipe.query_duration(gst.FORMAT_TIME)
+        except:
+            duration = gst.CLOCK_TIME_NONE
+
+        return (position, duration)
+
+    def _onMuxedAudioMessageCb(self, bus, message, pipe, output_file='output'):
+        if message.type != gst.MESSAGE_EOS:
+            return True
+
+        gobject.source_remove(self._audio_transcode_handler)
+        self._audio_transcode_handler = None
+        gobject.source_remove(self._transcode_id)
+        self._transcode_id = None
+        pipe.set_state(gst.STATE_NULL)
+        pipe.get_bus().remove_signal_watch()
+        pipe.get_bus().disable_sync_message_emission()
+
+        wavFilepath = os.path.join(self._activity.datapath,
+                                   output_file + '.wav')
+        oggFilepath = os.path.join(self._activity.datapath,
+                                   output_file + '.ogg')
+        os.remove( wavFilepath )
+        return False
+
+    def _bus_message_handler(self, bus, message):
+        t = message.type
+        if t == gst.MESSAGE_EOS:
+            if self._eos_cb:
+                cb = self._eos_cb
+                self._eos_cb = None
+                cb()
+        elif t == gst.MESSAGE_ERROR:
+            #todo: if we come out of suspend/resume with errors, then get us back up and running...
+            #todo: handle "No space left on the resource.gstfilesink.c"
+            #err, debug = message.parse_error()
+            pass
+
+    def abandonMedia(self):
+        self.stop()
+
+        if self._audio_transcode_handler:
+            gobject.source_remove(self._audio_transcode_handler)
+            self._audio_transcode_handler = None
+        if self._transcode_id:
+            gobject.source_remove(self._transcode_id)
+            self._transcode_id = None
+
+        wav_path = os.path.join(Instance.instancePath, "output.wav")
+        if os.path.exists(wav_path):
+            os.remove(wav_path)
+