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authorMathieu Duponchelle <mathieu@centricular.com>2018-07-29 20:06:09 +0200
committerMathieu Duponchelle <mathieu@centricular.com>2018-07-29 20:06:09 +0200
commit4ce52c0211d120d4aea6176412dbbc4ff225a805 (patch)
tree316f05ad2d37dcbcef5614d33a062d1f9b489b15
parent71bb950965bb3420cf89da9e29fe0d142d386ffe (diff)
Examples: add audiotestsrc plugin example
-rw-r--r--examples/plugins/python/py_audiotestsrc.py190
-rw-r--r--examples/requirements.txt4
2 files changed, 193 insertions, 1 deletions
diff --git a/examples/plugins/python/py_audiotestsrc.py b/examples/plugins/python/py_audiotestsrc.py
new file mode 100644
index 0000000..6d5ff21
--- /dev/null
+++ b/examples/plugins/python/py_audiotestsrc.py
@@ -0,0 +1,190 @@
+'''
+Element that generates a sine audio wave with the specified frequency
+
+Requires numpy
+
+Example pipeline:
+
+gst-launch-1.0 py_audiotestsrc ! autoaudiosink
+'''
+
+import gi
+
+gi.require_version('Gst', '1.0')
+gi.require_version('GstBase', '1.0')
+gi.require_version('GstAudio', '1.0')
+
+from gi.repository import Gst, GLib, GObject, GstBase, GstAudio
+
+try:
+ import numpy as np
+except ImportError:
+ Gst.error('py_audiotestsrc requires numpy')
+ raise
+
+OCAPS = Gst.Caps.from_string (
+ 'audio/x-raw, format=F32LE, layout=interleaved, rate=44100, channels=2')
+
+SAMPLESPERBUFFER = 1024
+
+DEFAULT_FREQ = 440
+DEFAULT_VOLUME = 0.8
+DEFAULT_MUTE = False
+DEFAULT_IS_LIVE = False
+
+class AudioTestSrc(GstBase.BaseSrc):
+ __gstmetadata__ = ('CustomSrc','Src', \
+ 'Custom test src element', 'Mathieu Duponchelle')
+
+ __gproperties__ = {
+ "freq": (int,
+ "Frequency",
+ "Frequency of test signal",
+ 1,
+ GLib.MAXINT,
+ DEFAULT_FREQ,
+ GObject.ParamFlags.READWRITE
+ ),
+ "volume": (float,
+ "Volume",
+ "Volume of test signal",
+ 0.0,
+ 1.0,
+ DEFAULT_VOLUME,
+ GObject.ParamFlags.READWRITE
+ ),
+ "mute": (bool,
+ "Mute",
+ "Mute the test signal",
+ DEFAULT_MUTE,
+ GObject.ParamFlags.READWRITE
+ ),
+ "is-live": (bool,
+ "Is live",
+ "Whether to act as a live source",
+ DEFAULT_IS_LIVE,
+ GObject.ParamFlags.READWRITE
+ ),
+ }
+
+ __gsttemplates__ = Gst.PadTemplate.new("src",
+ Gst.PadDirection.SRC,
+ Gst.PadPresence.ALWAYS,
+ OCAPS)
+
+ def __init__(self):
+ GstBase.BaseSrc.__init__(self)
+ self.info = GstAudio.AudioInfo()
+
+ self.freq = DEFAULT_FREQ
+ self.volume = DEFAULT_VOLUME
+ self.mute = DEFAULT_MUTE
+
+ self.set_live(DEFAULT_IS_LIVE)
+ self.set_format(Gst.Format.TIME)
+
+ def do_set_caps(self, caps):
+ self.info.from_caps(caps)
+ self.set_blocksize(self.info.bpf * SAMPLESPERBUFFER)
+ return True
+
+ def do_get_property(self, prop):
+ if prop.name == 'freq':
+ return self.freq
+ elif prop.name == 'volume':
+ return self.volume
+ elif prop.name == 'mute':
+ return self.mute
+ elif prop.name == 'is-live':
+ return self.is_live
+ else:
+ raise AttributeError('unknown property %s' % prop.name)
+
+ def do_set_property(self, prop, value):
+ if prop.name == 'freq':
+ self.freq = value
+ elif prop.name == 'volume':
+ self.volume = value
+ elif prop.name == 'mute':
+ self.mute = value
+ elif prop.name == 'is-live':
+ self.set_live(value)
+ else:
+ raise AttributeError('unknown property %s' % prop.name)
+
+ def do_start (self):
+ self.next_sample = 0
+ self.next_byte = 0
+ self.next_time = 0
+ self.accumulator = 0
+ self.generate_samples_per_buffer = SAMPLESPERBUFFER
+
+ return True
+
+ def do_gst_base_src_query(self, query):
+ if query.type == Gst.QueryType.LATENCY:
+ latency = Gst.util_uint64_scale_int(self.generate_samples_per_buffer,
+ Gst.SECOND, self.info.rate)
+ is_live = self.is_live
+ query.set_latency(is_live, latency, Gst.CLOCK_TIME_NONE)
+ res = True
+ else:
+ res = GstBase.BaseSrc.do_query(self, query)
+ return res
+
+ def do_get_times(self, buf):
+ end = 0
+ start = 0
+ if self.is_live:
+ ts = buf.pts
+ if ts != Gst.CLOCK_TIME_NONE:
+ duration = buf.duration
+ if duration != Gst.CLOCK_TIME_NONE:
+ end = ts + duration
+ start = ts
+ else:
+ start = Gst.CLOCK_TIME_NONE
+ end = Gst.CLOCK_TIME_NONE
+
+ return start, end
+
+ def do_create(self, offset, length):
+ if length == -1:
+ samples = SAMPLESPERBUFFER
+ else:
+ samples = int(length / self.info.bpf)
+
+ self.generate_samples_per_buffer = samples
+
+ bytes_ = samples * self.info.bpf
+
+ next_sample = self.next_sample + samples
+ next_byte = self.next_byte + bytes_
+ next_time = Gst.util_uint64_scale_int(next_sample, Gst.SECOND, self.info.rate)
+
+ if not self.mute:
+ r = np.repeat(
+ np.arange(self.accumulator, self.accumulator + samples),
+ self.info.channels)
+ data = ((np.sin(2 * np.pi * r * self.freq / self.info.rate) * self.volume)
+ .astype(np.float32))
+ else:
+ data = [0] * bytes_
+
+ buf = Gst.Buffer.new_wrapped(bytes(data))
+
+ buf.offset = self.next_sample
+ buf.offset_end = next_sample
+ buf.pts = self.next_time
+ buf.duration = next_time - self.next_time
+
+ self.next_time = next_time
+ self.next_sample = next_sample
+ self.next_byte = next_byte
+ self.accumulator += samples
+ self.accumulator %= self.info.rate / self.freq
+
+ return (Gst.FlowReturn.OK, buf)
+
+
+__gstelementfactory__ = ("py_audiotestsrc", Gst.Rank.NONE, AudioTestSrc)
diff --git a/examples/requirements.txt b/examples/requirements.txt
index e3962db..b0358b3 100644
--- a/examples/requirements.txt
+++ b/examples/requirements.txt
@@ -3,5 +3,7 @@ Pillow >= 5.1.0
# audioplot plugin
matplotlib >= 2.1.1
-numpy >= 1.14.5
numpy_ringbuffer >= 0.2.1
+
+# audioplot and py_audiotestsrc plugins
+numpy >= 1.14.5