diff options
Diffstat (limited to 'pysamples/grecord.py')
| -rw-r--r-- | pysamples/grecord.py | 54 |
1 files changed, 28 insertions, 26 deletions
diff --git a/pysamples/grecord.py b/pysamples/grecord.py index 3d7efdb..8e5fffc 100644 --- a/pysamples/grecord.py +++ b/pysamples/grecord.py @@ -20,6 +20,7 @@ def myblock(tw, arg): from gi.repository import GObject GObject.threads_init() + Gst.init(None) from TurtleArt.tautils import get_path from TurtleArt.tagplay import play_audio_from_file @@ -45,7 +46,7 @@ def myblock(tw, arg): self._audio_transcode_handler = None self._transcode_id = None - self._pipeline = gst.Pipeline("Record") + self._pipeline = Gst.Pipeline("Record") self._create_audiobin() self._pipeline.add(self._audiobin) @@ -55,42 +56,43 @@ def myblock(tw, arg): def _create_audiobin(self): ''' Assemble all the pieces we need. ''' - src = gst.element_factory_make("alsasrc", "absrc") + src = Gst.ElementFactory.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) + 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") + caps_str = "audio/x-raw-int,rate=16000,channels=1,depth=16" + srccaps = Gst.Caps.from_string(caps_str) # guarantee perfect stream, important for A/V sync - rate = gst.element_factory_make("audiorate") + rate = Gst.ElementFactory.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 = Gst.ElementFactory.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") + enc = Gst.ElementFactory.make("wavenc", "abenc") - sink = gst.element_factory_make("filesink", "absink") + sink = Gst.ElementFactory.make("filesink", "absink") sink.set_property("location", self.capture_file) - self._audiobin = gst.Bin("audiobin") + 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) + Gst.element_link_many(rate, queue, enc, sink) def _log_queue_overrun(self, queue): ''' We use a buffer, which may overflow. ''' @@ -108,15 +110,15 @@ def myblock(tw, arg): def start_recording_audio(self): ''' Start the stream in order to start recording. ''' - if self._get_state() == gst.STATE_PLAYING: + if self._get_state() == Gst.State.PLAYING: return - self._pipeline.set_state(gst.STATE_PLAYING) + self._pipeline.set_state(Gst.State.PLAYING) self._recording = True def stop_recording_audio(self): ''' Stop recording and then convert the results into a .ogg file using a new stream. ''' - self._pipeline.set_state(gst.STATE_NULL) + self._pipeline.set_state(Gst.State.NULL) self._recording = False if not os.path.exists(self.capture_file) or \ @@ -128,7 +130,7 @@ def myblock(tw, arg): os.remove(self.save_file) line = 'filesrc location=' + self.capture_file + ' name=audioFilesrc ! wavparse name=audioWavparse ! audioconvert name=audioAudioconvert ! vorbisenc name=audioVorbisenc ! oggmux name=audioOggmux ! filesink name=audioFilesink' - audioline = gst.parse_launch(line) + audioline = Gst.parse_launch(line) vorbis_enc = audioline.get_by_name('audioVorbisenc') @@ -141,12 +143,12 @@ def myblock(tw, arg): 'message', self._onMuxedAudioMessageCb, audioline) self._transcode_id = GObject.timeout_add( 200, self._transcodeUpdateCb, audioline) - audioline.set_state(gst.STATE_PLAYING) + audioline.set_state(Gst.State.PLAYING) def _transcodeUpdateCb(self, pipe): ''' Where are we in the transcoding process? ''' position, duration = self._query_position(pipe) - if position != gst.CLOCK_TIME_NONE: + if position != Gst.CLOCK_TIME_NONE: value = position * 100.0 / duration value = value/100.0 return True @@ -154,27 +156,27 @@ def myblock(tw, arg): def _query_position(self, pipe): ''' Where are we in the stream? ''' try: - position, format = pipe.query_position(gst.FORMAT_TIME) + position, format = pipe.query_position(Gst.FORMAT_TIME) except: - position = gst.CLOCK_TIME_NONE + position = Gst.CLOCK_TIME_NONE try: - duration, format = pipe.query_duration(gst.FORMAT_TIME) + duration, format = pipe.query_duration(Gst.FORMAT_TIME) except: - duration = gst.CLOCK_TIME_NONE + duration = Gst.CLOCK_TIME_NONE return (position, duration) def _onMuxedAudioMessageCb(self, bus, message, pipe): ''' Clean up at end of stream.''' - if message.type != gst.MESSAGE_EOS: + if message.type != Gst.MessageType.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.set_state(Gst.State.NULL) pipe.get_bus().remove_signal_watch() pipe.get_bus().disable_sync_message_emission() @@ -184,12 +186,12 @@ def myblock(tw, arg): def _bus_message_handler(self, bus, message): ''' Handle any messages associated with the stream. ''' t = message.type - if t == gst.MESSAGE_EOS: + if t == Gst.MessageType.EOS: if self._eos_cb: cb = self._eos_cb self._eos_cb = None cb() - elif t == gst.MESSAGE_ERROR: + elif t == Gst.MessageType.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 = |