class LiveDecoder(FileDecoder):
"""Live source Decoder based on Gstreamer
capturing audio from alsasrc
Construct a new LiveDecoder capturing audio from alsasrc
Parameters
----------
num_buffers : int, optional
Number of buffers to output before sending End Of Stream signal
(-1 = unlimited).
(Allowed values: >= -1, Default value: -1)
input_src : str, optional
Gstreamer source element
default to 'alsasrc'
possible values : 'autoaudiosrc', 'alsasrc', 'osssrc'
Examples
--------
>>> import timeside
>>> from timeside.core import get_processor
>>> live_decoder = get_processor('live_decoder')(num_buffers=5)
>>> waveform = get_processor('waveform_analyzer')()
>>> mp3_encoder = timeside.plugins.encoder.mp3.Mp3Encoder('/tmp/test_live.mp3',
... overwrite=True)
>>> pipe = (live_decoder | waveform | mp3_encoder)
>>> pipe.run() # doctest: +SKIP
>>> # Show the audio as captured by the decoder
>>> import matplotlib.pyplot as plt # doctest: +SKIP
>>> plt.plot(a.results['waveform_analyzer'].time, # doctest: +SKIP
a.results['waveform_analyzer'].data) # doctest: +SKIP
>>> plt.show() # doctest: +SKIP
"""
implements(IDecoder)
# IProcessor methods
@staticmethod
@interfacedoc
def id():
return "live_decoder"
def __init__(self, num_buffers=-1, input_src='alsasrc'):
super(Decoder, self).__init__()
self.num_buffers = num_buffers
self.uri = None
self.uri_start = 0
self.uri_duration = None
self.is_segment = False
self.input_src = input_src
self._sha1 = ''
def setup(self, channels=None, samplerate=None, blocksize=None):
self.eod = False
self.last_buffer = None
# a lock to wait wait for gstreamer thread to be ready
self.discovered_cond = threading.Condition(threading.Lock())
self.discovered = False
# the output data format we want
if blocksize:
self.output_blocksize = blocksize
if samplerate:
self.output_samplerate = int(samplerate)
if channels:
self.output_channels = int(channels)
# Create the pipe with standard Gstreamer uridecodbin
self.pipe = '''%s num-buffers=%d name=src
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
''' % (self.input_src, self.num_buffers)
self.pipeline = gst.parse_launch(self.pipe)
if self.output_channels:
caps_channels = int(self.output_channels)
else:
caps_channels = "[ 1, 2 ]"
if self.output_samplerate:
caps_samplerate = int(self.output_samplerate)
else:
caps_samplerate = "{ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 96000 }"
sink_caps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
width=(int)32,
rate=(int)%s""" % (caps_channels, caps_samplerate))
self.src = self.pipeline.get_by_name('src')
self.conv = self.pipeline.get_by_name('audioconvert')
self.conv.get_pad("sink").connect("notify::caps", self._notify_caps_cb)
self.sink = self.pipeline.get_by_name('sink')
self.sink.set_property("caps", sink_caps)
self.sink.set_property('max-buffers', GST_APPSINK_MAX_BUFFERS)
self.sink.set_property("drop", False)
self.sink.set_property('emit-signals', True)
self.sink.connect("new-buffer", self._on_new_buffer_cb)
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect('message', self._on_message_cb)
self.queue = Queue.Queue(QUEUE_SIZE)
self.mainloop = gobject.MainLoop()
self.mainloopthread = MainloopThread(self.mainloop)
self.mainloopthread.start()
#self.mainloopthread = get_loop_thread()
##self.mainloop = self.mainloopthread.mainloop
# start pipeline
self.pipeline.set_state(gst.STATE_PLAYING)
self.discovered_cond.acquire()
while not self.discovered:
# print 'waiting'
self.discovered_cond.wait()
self.discovered_cond.release()
if not hasattr(self, 'input_samplerate'):
if hasattr(self, 'error_msg'):
raise IOError(self.error_msg)
else:
raise IOError('no known audio stream found')
@interfacedoc
def process(self):
buf = self.queue.get()
if buf == gst.MESSAGE_EOS:
return self.last_buffer, True
frames, eod = buf
return frames, eod
def release(self):
# TODO : check if stack support is needed here
#if self.stack:
# self.stack = False
# self.from_stack = True
pass
class LiveDecoder(FileDecoder):
"""Live source Decoder based on Gstreamer
capturing audio from alsasrc
Construct a new LiveDecoder capturing audio from alsasrc
Parameters
----------
num_buffers : int, optional
Number of buffers to output before sending End Of Stream signal
(-1 = unlimited).
(Allowed values: >= -1, Default value: -1)
input_src : str, optional
Gstreamer source element
default to 'alsasrc'
possible values : 'autoaudiosrc', 'alsasrc', 'osssrc'
Examples
--------
>>> import timeside
>>> from timeside.core import get_processor
>>> live_decoder = get_processor('live_decoder')(num_buffers=5)
>>> waveform = get_processor('waveform_analyzer')()
>>> mp3_encoder = timeside.plugins.encoder.mp3.Mp3Encoder('/tmp/test_live.mp3',
... overwrite=True)
>>> pipe = (live_decoder | waveform | mp3_encoder)
>>> pipe.run() # doctest: +SKIP
>>> # Show the audio as captured by the decoder
>>> import matplotlib.pyplot as plt # doctest: +SKIP
>>> plt.plot(a.results['waveform_analyzer'].time, # doctest: +SKIP
a.results['waveform_analyzer'].data) # doctest: +SKIP
>>> plt.show() # doctest: +SKIP
"""
implements(IDecoder)
# IProcessor methods
@staticmethod
@interfacedoc
def id():
return "live_decoder"
def __init__(self, num_buffers=-1, input_src='alsasrc'):
super(Decoder, self).__init__()
self.num_buffers = num_buffers
self.uri = None
self.uri_start = 0
self.uri_duration = None
self.is_segment = False
self.input_src = input_src
self._sha1 = ''
def setup(self, channels=None, samplerate=None, blocksize=None):
self.eod = False
self.last_buffer = None
# a lock to wait wait for gstreamer thread to be ready
self.discovered_cond = threading.Condition(threading.Lock())
self.discovered = False
# the output data format we want
if blocksize:
self.output_blocksize = blocksize
if samplerate:
self.output_samplerate = int(samplerate)
if channels:
self.output_channels = int(channels)
# Create the pipe with standard Gstreamer uridecodbin
self.pipe = '''%s num-buffers=%d name=src
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
''' % (self.input_src, self.num_buffers)
self.pipeline = gst.parse_launch(self.pipe)
if self.output_channels:
caps_channels = int(self.output_channels)
else:
caps_channels = "[ 1, 2 ]"
if self.output_samplerate:
caps_samplerate = int(self.output_samplerate)
else:
caps_samplerate = "{ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 96000 }"
sink_caps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
width=(int)32,
rate=(int)%s""" % (caps_channels, caps_samplerate))
self.src = self.pipeline.get_by_name('src')
self.conv = self.pipeline.get_by_name('audioconvert')
self.conv.get_pad("sink").connect("notify::caps", self._notify_caps_cb)
self.sink = self.pipeline.get_by_name('sink')
self.sink.set_property("caps", sink_caps)
self.sink.set_property('max-buffers', GST_APPSINK_MAX_BUFFERS)
self.sink.set_property("drop", False)
self.sink.set_property('emit-signals', True)
self.sink.connect("new-buffer", self._on_new_buffer_cb)
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect('message', self._on_message_cb)
self.queue = Queue.Queue(QUEUE_SIZE)
self.mainloop = gobject.MainLoop()
self.mainloopthread = MainloopThread(self.mainloop)
self.mainloopthread.start()
#self.mainloopthread = get_loop_thread()
##self.mainloop = self.mainloopthread.mainloop
# start pipeline
self.pipeline.set_state(gst.STATE_PLAYING)
self.discovered_cond.acquire()
while not self.discovered:
# print 'waiting'
self.discovered_cond.wait()
self.discovered_cond.release()
if not hasattr(self, 'input_samplerate'):
if hasattr(self, 'error_msg'):
raise IOError(self.error_msg)
else:
raise IOError('no known audio stream found')
@interfacedoc
def process(self):
buf = self.queue.get()
if buf == gst.MESSAGE_EOS:
return self.last_buffer, True
frames, eod = buf
return frames, eod
def release(self):
# TODO : check if stack support is needed here
#if self.stack:
# self.stack = False
# self.from_stack = True
pass
def setup(self, channels=None, samplerate=None, blocksize=None):
self.eod = False
self.last_buffer = None
# a lock to wait wait for gstreamer thread to be ready
self.discovered_cond = threading.Condition(threading.Lock())
self.discovered = False
# the output data format we want
if blocksize:
self.output_blocksize = blocksize
if samplerate:
self.output_samplerate = int(samplerate)
if channels:
self.output_channels = int(channels)
# Create the pipe with standard Gstreamer uridecodbin
self.pipe = '''%s num-buffers=%d name=src
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
''' % (self.input_src, self.num_buffers)
self.pipeline = gst.parse_launch(self.pipe)
if self.output_channels:
caps_channels = int(self.output_channels)
else:
caps_channels = "[ 1, 2 ]"
if self.output_samplerate:
caps_samplerate = int(self.output_samplerate)
else:
caps_samplerate = "{ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 96000 }"
sink_caps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
width=(int)32,
rate=(int)%s""" % (caps_channels, caps_samplerate))
self.src = self.pipeline.get_by_name('src')
self.conv = self.pipeline.get_by_name('audioconvert')
self.conv.get_pad("sink").connect("notify::caps", self._notify_caps_cb)
self.sink = self.pipeline.get_by_name('sink')
self.sink.set_property("caps", sink_caps)
self.sink.set_property('max-buffers', GST_APPSINK_MAX_BUFFERS)
self.sink.set_property("drop", False)
self.sink.set_property('emit-signals', True)
self.sink.connect("new-buffer", self._on_new_buffer_cb)
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect('message', self._on_message_cb)
self.queue = Queue.Queue(QUEUE_SIZE)
self.mainloop = gobject.MainLoop()
self.mainloopthread = MainloopThread(self.mainloop)
self.mainloopthread.start()
#self.mainloopthread = get_loop_thread()
##self.mainloop = self.mainloopthread.mainloop
# start pipeline
self.pipeline.set_state(gst.STATE_PLAYING)
self.discovered_cond.acquire()
while not self.discovered:
# print 'waiting'
self.discovered_cond.wait()
self.discovered_cond.release()
if not hasattr(self, 'input_samplerate'):
if hasattr(self, 'error_msg'):
raise IOError(self.error_msg)
else:
raise IOError('no known audio stream found')
def setup(self, channels=None, samplerate=None, blocksize=None):
self.eod = False
self.last_buffer = None
# a lock to wait wait for gstreamer thread to be ready
self.discovered_cond = threading.Condition(threading.Lock())
self.discovered = False
# the output data format we want
if blocksize:
self.output_blocksize = blocksize
if samplerate:
self.output_samplerate = int(samplerate)
if channels:
self.output_channels = int(channels)
# Create the pipe with standard Gstreamer uridecodbin
self.pipe = '''%s num-buffers=%d name=src
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
''' % (self.input_src, self.num_buffers)
self.pipeline = gst.parse_launch(self.pipe)
if self.output_channels:
caps_channels = int(self.output_channels)
else:
caps_channels = "[ 1, 2 ]"
if self.output_samplerate:
caps_samplerate = int(self.output_samplerate)
else:
caps_samplerate = "{ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 96000 }"
sink_caps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
width=(int)32,
rate=(int)%s""" % (caps_channels, caps_samplerate))
self.src = self.pipeline.get_by_name('src')
self.conv = self.pipeline.get_by_name('audioconvert')
self.conv.get_pad("sink").connect("notify::caps", self._notify_caps_cb)
self.sink = self.pipeline.get_by_name('sink')
self.sink.set_property("caps", sink_caps)
self.sink.set_property('max-buffers', GST_APPSINK_MAX_BUFFERS)
self.sink.set_property("drop", False)
self.sink.set_property('emit-signals', True)
self.sink.connect("new-buffer", self._on_new_buffer_cb)
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect('message', self._on_message_cb)
self.queue = Queue.Queue(QUEUE_SIZE)
self.mainloop = gobject.MainLoop()
self.mainloopthread = MainloopThread(self.mainloop)
self.mainloopthread.start()
#self.mainloopthread = get_loop_thread()
##self.mainloop = self.mainloopthread.mainloop
# start pipeline
self.pipeline.set_state(gst.STATE_PLAYING)
self.discovered_cond.acquire()
while not self.discovered:
# print 'waiting'
self.discovered_cond.wait()
self.discovered_cond.release()
if not hasattr(self, 'input_samplerate'):
if hasattr(self, 'error_msg'):
raise IOError(self.error_msg)
else:
raise IOError('no known audio stream found')
class FileDecoder(Decoder):
""" File Decoder based on Gstreamer
Parameters
----------
uri : str
uri of the media
start : float, optional
start time of the segment in seconds
duration : float, optional
duration of the segment in seconds
stack : boolean, optional
keep decoded data in the stack
sha1 : boolean, optional
compute the sha1 hash of the data
Examples
--------
>>> import timeside
>>> from timeside.core import get_processor
>>> from timeside.core.tools.test_samples import samples
>>> audio_source = samples['sweep.wav']
>>> FileDecoder = get_processor('file_decoder') # Get the decoder class
>>> # Use decoder with default parameters
>>> decoder = FileDecoder(uri=audio_source)
>>> analyzer = get_processor('level')() # Pick a arbitrary analyzer
>>> pipe =(decoder | analyzer)
>>> pipe.run() # Run the pipe for the given audio source
"""
implements(IDecoder)
output_blocksize = 8 * 1024
pipeline = None
mainloopthread = None
# IProcessor methods
@staticmethod
@interfacedoc
def id():
return "file_decoder"
def __init__(self, uri, start=0, duration=None, stack=False, sha1=None):
super(FileDecoder, self).__init__(start=start, duration=duration)
self.from_stack = False
self.stack = stack
self.uri = get_uri(uri).encode('utf8')
if not sha1:
self._sha1 = get_sha1(uri)
else:
self._sha1 = sha1.encode('utf8')
self.uri_total_duration = get_media_uri_info(self.uri)['duration']
self.mimetype = None
def setup(self, channels=None, samplerate=None, blocksize=None):
self.eod = False
self.last_buffer = None
if self.from_stack:
self._frames_iterator = iter(self.process_pipe.frames_stack)
return
if self.stack:
self.process_pipe.frames_stack = []
if self.uri_duration is None:
# Set the duration from the length of the file
self.uri_duration = self.uri_total_duration - self.uri_start
if self.is_segment:
# Check start and duration value
if self.uri_start > self.uri_total_duration:
raise ValueError(('Segment start time value exceed media ' +
'duration'))
if self.uri_start + self.uri_duration > self.uri_total_duration:
raise ValueError("""Segment duration value is too large \
given the media duration""")
# a lock to wait wait for gstreamer thread to be ready
self.discovered_cond = threading.Condition(threading.Lock())
self.discovered = False
# the output data format we want
if blocksize:
self.output_blocksize = blocksize
if samplerate:
self.output_samplerate = int(samplerate)
if channels:
self.output_channels = int(channels)
if self.is_segment:
# Create the pipe with Gnonlin gnlurisource
self.pipe = ''' gnlurisource name=src uri={uri}
start=0
duration={uri_duration}
media-start={uri_start}
media-duration={uri_duration}
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
'''.format(uri=self.uri,
uri_start=np.uint64(
round(self.uri_start * gst.SECOND)),
uri_duration=np.int64(round(self.uri_duration * gst.SECOND)))
# convert uri_start and uri_duration to
# nanoseconds
else:
# Create the pipe with standard Gstreamer uridecodebin
self.pipe = ''' uridecodebin name=src uri={uri}
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
'''.format(uri=self.uri)
self.pipeline = gst.parse_launch(self.pipe)
if self.output_channels:
caps_channels = int(self.output_channels)
else:
caps_channels = "[ 1, 2 ]"
if self.output_samplerate:
caps_samplerate = int(self.output_samplerate)
else:
caps_samplerate = "{ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 96000 }"
sink_caps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
width=(int)32,
rate=(int)%s""" % (caps_channels, caps_samplerate))
self.src = self.pipeline.get_by_name('src')
if not self.is_segment:
self.src.connect("autoplug-continue", self._autoplug_cb)
else:
uridecodebin = self.src.get_by_name('internal-uridecodebin')
uridecodebin.connect("autoplug-continue", self._autoplug_cb)
self.conv = self.pipeline.get_by_name('audioconvert')
self.conv.get_pad("sink").connect("notify::caps", self._notify_caps_cb)
self.sink = self.pipeline.get_by_name('sink')
self.sink.set_property("caps", sink_caps)
self.sink.set_property('max-buffers', GST_APPSINK_MAX_BUFFERS)
self.sink.set_property("drop", False)
self.sink.set_property('emit-signals', True)
self.sink.connect("new-buffer", self._on_new_buffer_cb)
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect('message', self._on_message_cb)
self.queue = Queue.Queue(QUEUE_SIZE)
self.mainloop = gobject.MainLoop()
self.mainloopthread = MainloopThread(self.mainloop)
self.mainloopthread.start()
#self.mainloopthread = get_loop_thread()
##self.mainloop = self.mainloopthread.mainloop
# start pipeline
self.pipeline.set_state(gst.STATE_PLAYING)
self.discovered_cond.acquire()
while not self.discovered:
# print 'waiting'
self.discovered_cond.wait()
self.discovered_cond.release()
if not hasattr(self, 'input_samplerate'):
if hasattr(self, 'error_msg'):
raise IOError(self.error_msg)
else:
raise IOError('no known audio stream found')
def _autoplug_cb(self, src, arg0, arg1):
# use the autoplug-continue callback from uridecodebin
# to get the mimetype string
if not self.mimetype:
self.mimetype = arg1.to_string().split(',')[0]
return True
def _notify_caps_cb(self, pad, args):
self.discovered_cond.acquire()
caps = pad.get_negotiated_caps()
if not caps:
pad.info("no negotiated caps available")
self.discovered = True
self.discovered_cond.notify()
self.discovered_cond.release()
return
# the caps are fixed
# We now get the total length of that stream
q = gst.query_new_duration(gst.FORMAT_TIME)
pad.info("sending duration query")
if pad.get_peer().query(q):
format, length = q.parse_duration()
if format == gst.FORMAT_TIME:
pad.info("got duration (time) : %s" % (gst.TIME_ARGS(length),))
else:
pad.info("got duration : %d [format:%d]" % (length, format))
else:
length = -1
gst.warning("duration query failed")
# We store the caps and length in the proper location
if "audio" in caps.to_string():
self.input_samplerate = caps[0]["rate"]
if not self.output_samplerate:
self.output_samplerate = self.input_samplerate
self.input_channels = caps[0]["channels"]
if not self.output_channels:
self.output_channels = self.input_channels
self.input_duration = length / gst.SECOND
self.input_totalframes = int(
self.input_duration * self.input_samplerate)
if "x-raw-float" in caps.to_string():
self.input_width = caps[0]["width"]
else:
self.input_width = caps[0]["depth"]
self.discovered = True
self.discovered_cond.notify()
self.discovered_cond.release()
def _on_message_cb(self, bus, message):
t = message.type
if t == gst.MESSAGE_EOS:
self.queue.put(gst.MESSAGE_EOS)
self.pipeline.set_state(gst.STATE_NULL)
self.mainloop.quit()
elif t == gst.MESSAGE_ERROR:
self.pipeline.set_state(gst.STATE_NULL)
err, debug = message.parse_error()
self.discovered_cond.acquire()
self.discovered = True
self.mainloop.quit()
self.error_msg = "Error: %s" % err, debug
self.discovered_cond.notify()
self.discovered_cond.release()
elif t == gst.MESSAGE_TAG:
# TODO
# msg.parse_tags()
pass
def _on_new_buffer_cb(self, sink):
buf = sink.emit('pull-buffer')
new_array = gst_buffer_to_numpy_array(buf, self.output_channels)
# print 'processing new buffer', new_array.shape
if self.last_buffer is None:
self.last_buffer = new_array
else:
self.last_buffer = np.concatenate(
(self.last_buffer, new_array), axis=0)
while self.last_buffer.shape[0] >= self.output_blocksize:
new_block = self.last_buffer[:self.output_blocksize]
self.last_buffer = self.last_buffer[self.output_blocksize:]
# print 'queueing', new_block.shape, 'remaining',
# self.last_buffer.shape
self.queue.put([new_block, False])
@interfacedoc
@stack
def process(self):
buf = self.queue.get()
if buf == gst.MESSAGE_EOS:
return self.last_buffer, True
frames, eod = buf
return frames, eod
@interfacedoc
def totalframes(self):
if self.input_samplerate == self.output_samplerate:
return self.input_totalframes
else:
ratio = self.output_samplerate / self.input_samplerate
return int(self.input_totalframes * ratio)
@interfacedoc
def release(self):
if self.stack:
self.stack = False
self.from_stack = True
# IDecoder methods
@interfacedoc
def format(self):
return self.mime_type()
@interfacedoc
def mime_type(self):
if self.mimetype == 'application/x-id3':
self.mimetype = 'audio/mpeg'
return self.mimetype
@interfacedoc
def encoding(self):
# TODO check
return self.mime_type().split('/')[-1]
@interfacedoc
def resolution(self):
# TODO check: width or depth?
return self.input_width
@interfacedoc
def metadata(self):
# TODO check
return self.tags
def stop(self):
self.src.send_event(gst.event_new_eos())
def setup(self, channels=None, samplerate=None, blocksize=None):
self.eod = False
self.last_buffer = None
if self.from_stack:
self._frames_iterator = iter(self.process_pipe.frames_stack)
return
if self.stack:
self.process_pipe.frames_stack = []
if self.uri_duration is None:
# Set the duration from the length of the file
self.uri_duration = self.uri_total_duration - self.uri_start
if self.is_segment:
# Check start and duration value
if self.uri_start > self.uri_total_duration:
raise ValueError(('Segment start time value exceed media ' +
'duration'))
if self.uri_start + self.uri_duration > self.uri_total_duration:
raise ValueError("""Segment duration value is too large \
given the media duration""")
# a lock to wait wait for gstreamer thread to be ready
self.discovered_cond = threading.Condition(threading.Lock())
self.discovered = False
# the output data format we want
if blocksize:
self.output_blocksize = blocksize
if samplerate:
self.output_samplerate = int(samplerate)
if channels:
self.output_channels = int(channels)
if self.is_segment:
# Create the pipe with Gnonlin gnlurisource
self.pipe = ''' gnlurisource name=src uri={uri}
start=0
duration={uri_duration}
media-start={uri_start}
media-duration={uri_duration}
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
'''.format(uri=self.uri,
uri_start=np.uint64(
round(self.uri_start * gst.SECOND)),
uri_duration=np.int64(round(self.uri_duration * gst.SECOND)))
# convert uri_start and uri_duration to
# nanoseconds
else:
# Create the pipe with standard Gstreamer uridecodebin
self.pipe = ''' uridecodebin name=src uri={uri}
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
'''.format(uri=self.uri)
self.pipeline = gst.parse_launch(self.pipe)
if self.output_channels:
caps_channels = int(self.output_channels)
else:
caps_channels = "[ 1, 2 ]"
if self.output_samplerate:
caps_samplerate = int(self.output_samplerate)
else:
caps_samplerate = "{ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 96000 }"
sink_caps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
width=(int)32,
rate=(int)%s""" % (caps_channels, caps_samplerate))
self.src = self.pipeline.get_by_name('src')
if not self.is_segment:
self.src.connect("autoplug-continue", self._autoplug_cb)
else:
uridecodebin = self.src.get_by_name('internal-uridecodebin')
uridecodebin.connect("autoplug-continue", self._autoplug_cb)
self.conv = self.pipeline.get_by_name('audioconvert')
self.conv.get_pad("sink").connect("notify::caps", self._notify_caps_cb)
self.sink = self.pipeline.get_by_name('sink')
self.sink.set_property("caps", sink_caps)
self.sink.set_property('max-buffers', GST_APPSINK_MAX_BUFFERS)
self.sink.set_property("drop", False)
self.sink.set_property('emit-signals', True)
self.sink.connect("new-buffer", self._on_new_buffer_cb)
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect('message', self._on_message_cb)
self.queue = Queue.Queue(QUEUE_SIZE)
self.mainloop = gobject.MainLoop()
self.mainloopthread = MainloopThread(self.mainloop)
self.mainloopthread.start()
#self.mainloopthread = get_loop_thread()
##self.mainloop = self.mainloopthread.mainloop
# start pipeline
self.pipeline.set_state(gst.STATE_PLAYING)
self.discovered_cond.acquire()
while not self.discovered:
# print 'waiting'
self.discovered_cond.wait()
self.discovered_cond.release()
if not hasattr(self, 'input_samplerate'):
if hasattr(self, 'error_msg'):
raise IOError(self.error_msg)
else:
raise IOError('no known audio stream found')
class FileDecoder(Decoder):
""" File Decoder based on Gstreamer
Parameters
----------
uri : str
uri of the media
start : float, optional
start time of the segment in seconds
duration : float, optional
duration of the segment in seconds
stack : boolean, optional
keep decoded data in the stack
sha1 : boolean, optional
compute the sha1 hash of the data
Examples
--------
>>> import timeside
>>> from timeside.core import get_processor
>>> from timeside.core.tools.test_samples import samples
>>> audio_source = samples['sweep.wav']
>>> FileDecoder = get_processor('file_decoder') # Get the decoder class
>>> # Use decoder with default parameters
>>> decoder = FileDecoder(uri=audio_source)
>>> analyzer = get_processor('level')() # Pick a arbitrary analyzer
>>> pipe =(decoder | analyzer)
>>> pipe.run() # Run the pipe for the given audio source
"""
implements(IDecoder)
output_blocksize = 8 * 1024
pipeline = None
mainloopthread = None
# IProcessor methods
@staticmethod
@interfacedoc
def id():
return "file_decoder"
def __init__(self, uri, start=0, duration=None, stack=False, sha1=None):
super(FileDecoder, self).__init__(start=start, duration=duration)
self.from_stack = False
self.stack = stack
self.uri = get_uri(uri).encode('utf8')
if not sha1:
self._sha1 = get_sha1(uri)
else:
self._sha1 = sha1.encode('utf8')
self.uri_total_duration = get_media_uri_info(self.uri)['duration']
self.mimetype = None
def setup(self, channels=None, samplerate=None, blocksize=None):
self.eod = False
self.last_buffer = None
if self.from_stack:
self._frames_iterator = iter(self.process_pipe.frames_stack)
return
if self.stack:
self.process_pipe.frames_stack = []
if self.uri_duration is None:
# Set the duration from the length of the file
self.uri_duration = self.uri_total_duration - self.uri_start
if self.is_segment:
# Check start and duration value
if self.uri_start > self.uri_total_duration:
raise ValueError(
('Segment start time value exceed media ' + 'duration'))
if self.uri_start + self.uri_duration > self.uri_total_duration:
raise ValueError("""Segment duration value is too large \
given the media duration""")
# a lock to wait wait for gstreamer thread to be ready
self.discovered_cond = threading.Condition(threading.Lock())
self.discovered = False
# the output data format we want
if blocksize:
self.output_blocksize = blocksize
if samplerate:
self.output_samplerate = int(samplerate)
if channels:
self.output_channels = int(channels)
if self.is_segment:
# Create the pipe with Gnonlin gnlurisource
self.pipe = ''' gnlurisource name=src uri={uri}
start=0
duration={uri_duration}
media-start={uri_start}
media-duration={uri_duration}
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
'''.format(
uri=self.uri,
uri_start=np.uint64(round(self.uri_start * gst.SECOND)),
uri_duration=np.int64(round(self.uri_duration * gst.SECOND)))
# convert uri_start and uri_duration to
# nanoseconds
else:
# Create the pipe with standard Gstreamer uridecodebin
self.pipe = ''' uridecodebin name=src uri={uri}
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
'''.format(uri=self.uri)
self.pipeline = gst.parse_launch(self.pipe)
if self.output_channels:
caps_channels = int(self.output_channels)
else:
caps_channels = "[ 1, 2 ]"
if self.output_samplerate:
caps_samplerate = int(self.output_samplerate)
else:
caps_samplerate = "{ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 96000 }"
sink_caps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
width=(int)32,
rate=(int)%s""" % (caps_channels, caps_samplerate))
self.src = self.pipeline.get_by_name('src')
if not self.is_segment:
self.src.connect("autoplug-continue", self._autoplug_cb)
else:
uridecodebin = self.src.get_by_name('internal-uridecodebin')
uridecodebin.connect("autoplug-continue", self._autoplug_cb)
self.conv = self.pipeline.get_by_name('audioconvert')
self.conv.get_pad("sink").connect("notify::caps", self._notify_caps_cb)
self.sink = self.pipeline.get_by_name('sink')
self.sink.set_property("caps", sink_caps)
self.sink.set_property('max-buffers', GST_APPSINK_MAX_BUFFERS)
self.sink.set_property("drop", False)
self.sink.set_property('emit-signals', True)
self.sink.connect("new-buffer", self._on_new_buffer_cb)
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect('message', self._on_message_cb)
self.queue = Queue.Queue(QUEUE_SIZE)
self.mainloop = gobject.MainLoop()
self.mainloopthread = MainloopThread(self.mainloop)
self.mainloopthread.start()
#self.mainloopthread = get_loop_thread()
##self.mainloop = self.mainloopthread.mainloop
# start pipeline
self.pipeline.set_state(gst.STATE_PLAYING)
self.discovered_cond.acquire()
while not self.discovered:
# print 'waiting'
self.discovered_cond.wait()
self.discovered_cond.release()
if not hasattr(self, 'input_samplerate'):
if hasattr(self, 'error_msg'):
raise IOError(self.error_msg)
else:
raise IOError('no known audio stream found')
def _autoplug_cb(self, src, arg0, arg1):
# use the autoplug-continue callback from uridecodebin
# to get the mimetype string
if not self.mimetype:
self.mimetype = arg1.to_string().split(',')[0]
return True
def _notify_caps_cb(self, pad, args):
self.discovered_cond.acquire()
caps = pad.get_negotiated_caps()
if not caps:
pad.info("no negotiated caps available")
self.discovered = True
self.discovered_cond.notify()
self.discovered_cond.release()
return
# the caps are fixed
# We now get the total length of that stream
q = gst.query_new_duration(gst.FORMAT_TIME)
pad.info("sending duration query")
if pad.get_peer().query(q):
format, length = q.parse_duration()
if format == gst.FORMAT_TIME:
pad.info("got duration (time) : %s" %
(gst.TIME_ARGS(length), ))
else:
pad.info("got duration : %d [format:%d]" % (length, format))
else:
length = -1
gst.warning("duration query failed")
# We store the caps and length in the proper location
if "audio" in caps.to_string():
self.input_samplerate = caps[0]["rate"]
if not self.output_samplerate:
self.output_samplerate = self.input_samplerate
self.input_channels = caps[0]["channels"]
if not self.output_channels:
self.output_channels = self.input_channels
self.input_duration = length / gst.SECOND
self.input_totalframes = int(self.input_duration *
self.input_samplerate)
if "x-raw-float" in caps.to_string():
self.input_width = caps[0]["width"]
else:
self.input_width = caps[0]["depth"]
self.discovered = True
self.discovered_cond.notify()
self.discovered_cond.release()
def _on_message_cb(self, bus, message):
t = message.type
if t == gst.MESSAGE_EOS:
self.queue.put(gst.MESSAGE_EOS)
self.pipeline.set_state(gst.STATE_NULL)
self.mainloop.quit()
elif t == gst.MESSAGE_ERROR:
self.pipeline.set_state(gst.STATE_NULL)
err, debug = message.parse_error()
self.discovered_cond.acquire()
self.discovered = True
self.mainloop.quit()
self.error_msg = "Error: %s" % err, debug
self.discovered_cond.notify()
self.discovered_cond.release()
elif t == gst.MESSAGE_TAG:
# TODO
# msg.parse_tags()
pass
def _on_new_buffer_cb(self, sink):
buf = sink.emit('pull-buffer')
new_array = gst_buffer_to_numpy_array(buf, self.output_channels)
# print 'processing new buffer', new_array.shape
if self.last_buffer is None:
self.last_buffer = new_array
else:
self.last_buffer = np.concatenate((self.last_buffer, new_array),
axis=0)
while self.last_buffer.shape[0] >= self.output_blocksize:
new_block = self.last_buffer[:self.output_blocksize]
self.last_buffer = self.last_buffer[self.output_blocksize:]
# print 'queueing', new_block.shape, 'remaining',
# self.last_buffer.shape
self.queue.put([new_block, False])
@interfacedoc
@stack
def process(self):
buf = self.queue.get()
if buf == gst.MESSAGE_EOS:
return self.last_buffer, True
frames, eod = buf
return frames, eod
@interfacedoc
def totalframes(self):
if self.input_samplerate == self.output_samplerate:
return self.input_totalframes
else:
ratio = self.output_samplerate / self.input_samplerate
return int(self.input_totalframes * ratio)
@interfacedoc
def release(self):
if self.stack:
self.stack = False
self.from_stack = True
# IDecoder methods
@interfacedoc
def format(self):
return self.mime_type()
@interfacedoc
def mime_type(self):
if self.mimetype == 'application/x-id3':
self.mimetype = 'audio/mpeg'
return self.mimetype
@interfacedoc
def encoding(self):
# TODO check
return self.mime_type().split('/')[-1]
@interfacedoc
def resolution(self):
# TODO check: width or depth?
return self.input_width
@interfacedoc
def metadata(self):
# TODO check
return self.tags
def stop(self):
self.src.send_event(gst.event_new_eos())
def setup(self, channels=None, samplerate=None, blocksize=None):
self.eod = False
self.last_buffer = None
if self.from_stack:
self._frames_iterator = iter(self.process_pipe.frames_stack)
return
if self.stack:
self.process_pipe.frames_stack = []
if self.uri_duration is None:
# Set the duration from the length of the file
self.uri_duration = self.uri_total_duration - self.uri_start
if self.is_segment:
# Check start and duration value
if self.uri_start > self.uri_total_duration:
raise ValueError(
('Segment start time value exceed media ' + 'duration'))
if self.uri_start + self.uri_duration > self.uri_total_duration:
raise ValueError("""Segment duration value is too large \
given the media duration""")
# a lock to wait wait for gstreamer thread to be ready
self.discovered_cond = threading.Condition(threading.Lock())
self.discovered = False
# the output data format we want
if blocksize:
self.output_blocksize = blocksize
if samplerate:
self.output_samplerate = int(samplerate)
if channels:
self.output_channels = int(channels)
if self.is_segment:
# Create the pipe with Gnonlin gnlurisource
self.pipe = ''' gnlurisource name=src uri={uri}
start=0
duration={uri_duration}
media-start={uri_start}
media-duration={uri_duration}
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
'''.format(
uri=self.uri,
uri_start=np.uint64(round(self.uri_start * gst.SECOND)),
uri_duration=np.int64(round(self.uri_duration * gst.SECOND)))
# convert uri_start and uri_duration to
# nanoseconds
else:
# Create the pipe with standard Gstreamer uridecodebin
self.pipe = ''' uridecodebin name=src uri={uri}
! audioconvert name=audioconvert
! audioresample
! appsink name=sink sync=False async=True
'''.format(uri=self.uri)
self.pipeline = gst.parse_launch(self.pipe)
if self.output_channels:
caps_channels = int(self.output_channels)
else:
caps_channels = "[ 1, 2 ]"
if self.output_samplerate:
caps_samplerate = int(self.output_samplerate)
else:
caps_samplerate = "{ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 96000 }"
sink_caps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
width=(int)32,
rate=(int)%s""" % (caps_channels, caps_samplerate))
self.src = self.pipeline.get_by_name('src')
if not self.is_segment:
self.src.connect("autoplug-continue", self._autoplug_cb)
else:
uridecodebin = self.src.get_by_name('internal-uridecodebin')
uridecodebin.connect("autoplug-continue", self._autoplug_cb)
self.conv = self.pipeline.get_by_name('audioconvert')
self.conv.get_pad("sink").connect("notify::caps", self._notify_caps_cb)
self.sink = self.pipeline.get_by_name('sink')
self.sink.set_property("caps", sink_caps)
self.sink.set_property('max-buffers', GST_APPSINK_MAX_BUFFERS)
self.sink.set_property("drop", False)
self.sink.set_property('emit-signals', True)
self.sink.connect("new-buffer", self._on_new_buffer_cb)
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect('message', self._on_message_cb)
self.queue = Queue.Queue(QUEUE_SIZE)
self.mainloop = gobject.MainLoop()
self.mainloopthread = MainloopThread(self.mainloop)
self.mainloopthread.start()
#self.mainloopthread = get_loop_thread()
##self.mainloop = self.mainloopthread.mainloop
# start pipeline
self.pipeline.set_state(gst.STATE_PLAYING)
self.discovered_cond.acquire()
while not self.discovered:
# print 'waiting'
self.discovered_cond.wait()
self.discovered_cond.release()
if not hasattr(self, 'input_samplerate'):
if hasattr(self, 'error_msg'):
raise IOError(self.error_msg)
else:
raise IOError('no known audio stream found')
