From fb651bb51725ddc1d227daab011c0693d89bc1dc Mon Sep 17 00:00:00 2001 From: Walter Bender Date: Mon, 11 Apr 2011 02:58:29 +0000 Subject: new sample for recording audio --- (limited to 'pysamples') diff --git a/pysamples/grecord.py b/pysamples/grecord.py new file mode 100644 index 0000000..f5e2836 --- /dev/null +++ b/pysamples/grecord.py @@ -0,0 +1,229 @@ +#Copyright (c) 2008, Media Modifications Ltd. +#Copyright (c) 2011, Walter Bender + +# This procedure is invoked when the user-definable block on the +# "extras" palette is selected. + +# Usage: Import this code into a Python (user-definable) block; Pass +# it 'start' to start recording; 'stop' to stop recording; 'play' +# to play back your recording; or 'save' to save to the Sugar Journal. + + +def myblock(tw, arg): + ''' Record and playback a sound (Sugar only) ''' + import os + import time + + import gtk + import gst + + import gobject + gobject.threads_init() + + from TurtleArt.tautils import get_path + from TurtleArt.tagplay import play_audio_from_file + from sugar.datastore import datastore + from sugar import profile + + + class Grecord: + + def __init__(self, tw): + ''' Set up the stream. ''' + datapath = get_path(tw.parent, 'instance') + self.capture_file = os.path.join(datapath, 'output.wav') + self.save_file = os.path.join(datapath, 'output.ogg') + self._eos_cb = None + + self._can_limit_framerate = False + self._playing = False + + self._audio_transcode_handler = None + self._transcode_id = None + + self._pipeline = gst.Pipeline("Record") + self._create_audiobin() + self._pipeline.add(self._audiobin) + + bus = self._pipeline.get_bus() + bus.add_signal_watch() + bus.connect('message', self._bus_message_handler) + + def _create_audiobin(self): + 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", self.capture_file) + + 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") + + def play(self): + 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): + self.stop() + + audio_path = self.capture_file + if not os.path.exists(audio_path) or \ + os.path.getsize(audio_path) <= 0: + 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 = self.save_file + audioFilesink.set_property("location", audioOggFilepath) + + audioBus = audioline.get_bus() + audioBus.add_signal_watch() + self._audio_transcode_handler = audioBus.connect( + 'message', self._onMuxedAudioMessageCb, audioline) + 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.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): + 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 = self.capture_file + oggFilepath = self.save_file + 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 + + # We store the stream as tw.grecord so that we can use it repeatedly. + if not hasattr(tw, 'grecord'): + tw.grecord = Grecord(tw) + + # Sometime we need to pass multiple arguments, e.g., save, savename + savename = _('turtle audio recording') + if type(arg) == type([]): + cmd = arg[0] + if len(arg) > 1: + savename = str(arg[1]) + else: + cmd = arg + + if cmd == _('start'): + tw.grecord.record_audio() + elif cmd == _('stop'): + if os.path.exists(tw.grecord.save_file): + os.remove(tw.grecord.save_file) + tw.grecord.stop_recording_audio() + elif cmd == _('play'): + play_audio_from_file(tw.lc, tw.grecord.save_file) + elif cmd == _('save'): + if os.path.exists(tw.grecord.save_file): + if tw.running_sugar: + dsobject = datastore.create() + dsobject.metadata['title'] = savename + dsobject.metadata['icon-color'] = \ + profile.get_color().to_string() + dsobject.metadata['mime_type'] = 'audio/ogg' + dsobject.set_file_path(tw.grecord.save_file) + datastore.write(dsobject) + dsobject.destroy() -- cgit v0.9.1