From 375ac7d255db443f5443021a5ca97b0ac43a30bc Mon Sep 17 00:00:00 2001 From: Walter Bender Date: Sat, 09 Apr 2011 19:58:37 +0000 Subject: adding record functionality --- (limited to 'utils') 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) + -- cgit v0.9.1