From 4ce52c0211d120d4aea6176412dbbc4ff225a805 Mon Sep 17 00:00:00 2001 From: Mathieu Duponchelle Date: Sun, 29 Jul 2018 20:06:09 +0200 Subject: [PATCH] Examples: add audiotestsrc plugin example --- examples/plugins/python/py_audiotestsrc.py | 190 +++++++++++++++++++++ examples/requirements.txt | 4 +- 2 files changed, 193 insertions(+), 1 deletion(-) create mode 100644 examples/plugins/python/py_audiotestsrc.py diff --git a/examples/plugins/python/py_audiotestsrc.py b/examples/plugins/python/py_audiotestsrc.py new file mode 100644 index 0000000000..6d5ff21a8f --- /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 e3962db8f7..b0358b3faa 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