gstreamer/subprojects/gst-python/examples/plugins/python/audioplot.py

243 lines
8.7 KiB
Python

'''
Element that transforms audio samples to video frames representing
the waveform.
Requires matplotlib, numpy and numpy_ringbuffer
Example pipeline:
gst-launch-1.0 audiotestsrc ! audioplot window-duration=0.01 ! videoconvert ! autovideosink
'''
import gi
gi.require_version('Gst', '1.0')
gi.require_version('GstBase', '1.0')
gi.require_version('GstAudio', '1.0')
gi.require_version('GstVideo', '1.0')
from gi.repository import Gst, GLib, GObject, GstBase, GstAudio, GstVideo
try:
import numpy as np
import matplotlib.patheffects as pe
from numpy_ringbuffer import RingBuffer
from matplotlib import pyplot as plt
from matplotlib.backends.backend_agg import FigureCanvasAgg
except ImportError:
Gst.error('audioplot requires numpy, numpy_ringbuffer and matplotlib')
raise
Gst.init(None)
AUDIO_FORMATS = [f.strip() for f in
GstAudio.AUDIO_FORMATS_ALL.strip('{ }').split(',')]
ICAPS = Gst.Caps(Gst.Structure('audio/x-raw',
format=Gst.ValueList(AUDIO_FORMATS),
layout='interleaved',
rate = Gst.IntRange(range(1, GLib.MAXINT)),
channels = Gst.IntRange(range(1, GLib.MAXINT))))
OCAPS = Gst.Caps(Gst.Structure('video/x-raw',
format='ARGB',
width=Gst.IntRange(range(1, GLib.MAXINT)),
height=Gst.IntRange(range(1, GLib.MAXINT)),
framerate=Gst.FractionRange(Gst.Fraction(1, 1),
Gst.Fraction(GLib.MAXINT, 1))))
DEFAULT_WINDOW_DURATION = 1.0
DEFAULT_WIDTH = 640
DEFAULT_HEIGHT = 480
DEFAULT_FRAMERATE_NUM = 25
DEFAULT_FRAMERATE_DENOM = 1
class AudioPlotFilter(GstBase.BaseTransform):
__gstmetadata__ = ('AudioPlotFilter','Filter', \
'Plot audio waveforms', 'Mathieu Duponchelle')
__gsttemplates__ = (Gst.PadTemplate.new("src",
Gst.PadDirection.SRC,
Gst.PadPresence.ALWAYS,
OCAPS),
Gst.PadTemplate.new("sink",
Gst.PadDirection.SINK,
Gst.PadPresence.ALWAYS,
ICAPS))
__gproperties__ = {
"window-duration": (float,
"Window Duration",
"Duration of the sliding window, in seconds",
0.01,
100.0,
DEFAULT_WINDOW_DURATION,
GObject.ParamFlags.READWRITE
)
}
def __init__(self):
GstBase.BaseTransform.__init__(self)
self.window_duration = DEFAULT_WINDOW_DURATION
def do_get_property(self, prop):
if prop.name == 'window-duration':
return self.window_duration
else:
raise AttributeError('unknown property %s' % prop.name)
def do_set_property(self, prop, value):
if prop.name == 'window-duration':
self.window_duration = value
else:
raise AttributeError('unknown property %s' % prop.name)
def do_transform(self, inbuf, outbuf):
if not self.h:
self.h, = self.ax.plot(np.array(self.ringbuffer),
lw=0.5,
color='k',
path_effects=[pe.Stroke(linewidth=1.0,
foreground='g'),
pe.Normal()])
else:
self.h.set_ydata(np.array(self.ringbuffer))
self.fig.canvas.restore_region(self.background)
self.ax.draw_artist(self.h)
self.fig.canvas.blit(self.ax.bbox)
s = self.agg.tostring_argb()
outbuf.fill(0, s)
outbuf.pts = self.next_time
outbuf.duration = self.frame_duration
self.next_time += self.frame_duration
return Gst.FlowReturn.OK
def __append(self, data):
arr = np.array(data)
end = self.thinning_factor * int(len(arr) / self.thinning_factor)
arr = np.mean(arr[:end].reshape(-1, self.thinning_factor), 1)
self.ringbuffer.extend(arr)
def do_generate_output(self):
inbuf = self.queued_buf
_, info = inbuf.map(Gst.MapFlags.READ)
res, data = self.converter.convert(GstAudio.AudioConverterFlags.NONE,
info.data)
data = memoryview(data).cast('i')
nsamples = len(data) - self.buf_offset
if nsamples == 0:
self.buf_offset = 0
inbuf.unmap(info)
return Gst.FlowReturn.OK, None
if self.cur_offset + nsamples < self.next_offset:
self.__append(data[self.buf_offset:])
self.buf_offset = 0
self.cur_offset += nsamples
inbuf.unmap(info)
return Gst.FlowReturn.OK, None
consumed = self.next_offset - self.cur_offset
self.__append(data[self.buf_offset:self.buf_offset + consumed])
inbuf.unmap(info)
_, outbuf = GstBase.BaseTransform.do_prepare_output_buffer(self, inbuf)
ret = self.do_transform(inbuf, outbuf)
self.next_offset += self.samplesperbuffer
self.cur_offset += consumed
self.buf_offset += consumed
return ret, outbuf
def do_transform_caps(self, direction, caps, filter_):
if direction == Gst.PadDirection.SRC:
res = ICAPS
else:
res = OCAPS
if filter_:
res = res.intersect(filter_)
return res
def do_fixate_caps(self, direction, caps, othercaps):
if direction == Gst.PadDirection.SRC:
return othercaps.fixate()
else:
so = othercaps.get_structure(0).copy()
so.fixate_field_nearest_fraction("framerate",
DEFAULT_FRAMERATE_NUM,
DEFAULT_FRAMERATE_DENOM)
so.fixate_field_nearest_int("width", DEFAULT_WIDTH)
so.fixate_field_nearest_int("height", DEFAULT_HEIGHT)
ret = Gst.Caps.new_empty()
ret.append_structure(so)
return ret.fixate()
def do_set_caps(self, icaps, ocaps):
in_info = GstAudio.AudioInfo()
in_info.from_caps(icaps)
out_info = GstVideo.VideoInfo()
out_info.from_caps(ocaps)
self.convert_info = GstAudio.AudioInfo()
self.convert_info.set_format(GstAudio.AudioFormat.S32,
in_info.rate,
in_info.channels,
in_info.position)
self.converter = GstAudio.AudioConverter.new(GstAudio.AudioConverterFlags.NONE,
in_info,
self.convert_info,
None)
self.fig = plt.figure()
dpi = self.fig.get_dpi()
self.fig.patch.set_alpha(0.3)
self.fig.set_size_inches(out_info.width / float(dpi),
out_info.height / float(dpi))
self.ax = plt.Axes(self.fig, [0., 0., 1., 1.])
self.fig.add_axes(self.ax)
self.ax.set_axis_off()
self.ax.set_ylim((GLib.MININT, GLib.MAXINT))
self.agg = self.fig.canvas.switch_backends(FigureCanvasAgg)
self.h = None
samplesperwindow = int(in_info.rate * in_info.channels * self.window_duration)
self.thinning_factor = max(int(samplesperwindow / out_info.width - 1), 1)
cap = int(samplesperwindow / self.thinning_factor)
self.ax.set_xlim([0, cap])
self.ringbuffer = RingBuffer(capacity=cap)
self.ringbuffer.extend([0.0] * cap)
self.frame_duration = Gst.util_uint64_scale_int(Gst.SECOND,
out_info.fps_d,
out_info.fps_n)
self.next_time = self.frame_duration
self.agg.draw()
self.background = self.fig.canvas.copy_from_bbox(self.ax.bbox)
self.samplesperbuffer = Gst.util_uint64_scale_int(in_info.rate * in_info.channels,
out_info.fps_d,
out_info.fps_n)
self.next_offset = self.samplesperbuffer
self.cur_offset = 0
self.buf_offset = 0
return True
GObject.type_register(AudioPlotFilter)
__gstelementfactory__ = ("audioplot", Gst.Rank.NONE, AudioPlotFilter)