コード例 #1
0
    def __init__(self, name, frame_size=512):
        self.file_pointer = WaveReader(name)
        self.name = name
        self.channels = self.file_pointer.channels
        self.data = numpy.zeros((self.channels, frame_size), numpy.float32, order='F')
        self.nframes = 1

        self.frame_size = frame_size
        self.samplerate = self.file_pointer.samplerate
コード例 #2
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class Replayer:
    def __init__(self, wav_file, start_pos=None):
        """
        Args:
            wav_file (str): File path
            start_pos (int): Starting chunk index
        """
        self.file_path = wav_file
        self.file = WaveReader(wav_file, config.SAMPLE_RATE,
                               config.NUM_CHANNELS, Format.WAV | Format.PCM_16)
        if start_pos is None:
            self.playback_pos = random.randint(0, self.file.frames)
        else:
            self.playback_pos = start_pos

        # Lower the amplitude so that older files (indicated in file name)
            # are softer
        # Time the file was created in seconds
        file_creation_time = int(os.path.basename(self.file_path)[:-6])
        age = int(time.time()) - file_creation_time
        adjusted_age = age / 200
        if adjusted_age == 0:
            adjusted_age = 0.0001
        amp = (1 / adjusted_age) * 0.9
        if amp > 0.3:
            amp = 0.3

        self.amplitude = Amplitude(amp)

    def get_chunk(self):
        """
        Returns: None
        """
        # TODO: Document me!
        self.file.seek(self.playback_pos)
        # TODO: Is this zeros bit necessary?
        chunk = numpy.zeros((config.NUM_CHANNELS, config.CHUNK_SIZE),
                            numpy.int16,
                            order='F')
        # chunk = next(self.file.read_iter(shared.CHUNK_SIZE))
        # chunk = self.data[self.playback_pos] * self.amplitude.value
        # Multiply by amplitude here?
        chunk = chunk * self.amplitude.value
        # print(chunk.shape)
        self.playback_pos += config.CHUNK_SIZE
        if self.playback_pos > self.file.frames:
            self.playback_pos = 0
        return chunk
コード例 #3
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 def addSound(self, sound_name):
     sound_file = WaveReader(sound_name)
     self.playing_files.append(sound_file)
     self.max_channels = max(self.max_channels, sound_file.channels)
     self.playing_files_buffers_for_read[sound_file] = numpy.zeros((sound_file.channels,512), numpy.float32, order='F')
     self.store_buffers.append([None])
     self.data_list = numpy.zeros((len(self.playing_files), self.max_channels,512), numpy.float32, order='F')
コード例 #4
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    def __init__(self, wav_file, start_pos=None):
        """
        Args:
            wav_file (str): File path
            start_pos (int): Starting chunk index
        """
        self.file_path = wav_file
        self.file = WaveReader(wav_file, config.SAMPLE_RATE,
                               config.NUM_CHANNELS, Format.WAV | Format.PCM_16)
        if start_pos is None:
            self.playback_pos = random.randint(0, self.file.frames)
        else:
            self.playback_pos = start_pos

        # Lower the amplitude so that older files (indicated in file name)
            # are softer
        # Time the file was created in seconds
        file_creation_time = int(os.path.basename(self.file_path)[:-6])
        age = int(time.time()) - file_creation_time
        adjusted_age = age / 200
        if adjusted_age == 0:
            adjusted_age = 0.0001
        amp = (1 / adjusted_age) * 0.9
        if amp > 0.3:
            amp = 0.3

        self.amplitude = Amplitude(amp)
コード例 #5
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 def __init__(self, sound_name):
     self.sound_name = sound_name
     sound_file = WaveReader(sound_name)
     self.playing_files = []
     self.playing_files.append(sound_file)
     print("playing_files = {}".format(self.playing_files))
     self.player_lib = pyaudio.PyAudio()
     self._volume = 1
     self.stream = None
コード例 #6
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def get_peak_volume(filepath):
    max_volume = 0.0
    try:
        with WaveReader(filepath) as r:
            for data in r.read_iter(size=streamChunk):
                left_channel = data[0]
                volume = np.linalg.norm(left_channel)
                if volume > max_volume:
                    max_volume = volume
    except Exception as e:
        logging.error("could not get peak volume, assuming 0. Exception: ", e)

    return max_volume
コード例 #7
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class SoundFile:
    def __init__(self, name, frame_size=512):
        self.file_pointer = WaveReader(name)
        self.name = name
        self.channels = self.file_pointer.channels
        self.data = numpy.zeros((self.channels, frame_size), numpy.float32, order='F')
        self.nframes = 1

        self.frame_size = frame_size
        self.samplerate = self.file_pointer.samplerate

    def read(self):
        self.nframes = self.file_pointer.read(self.data)
        return self.data[:self.channels, :self.nframes]
コード例 #8
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def test_record_200ms():
    filename = tempfile.gettempdir() + "/unittest_record.wav"
    recording_duration = 0.2
    removeFileIfItExists(filename)

    sound_input.record(filename, 0.2)

    assert os.path.isfile(
        filename), "Expected recording to be present at {}.".format(filename)

    with WaveReader(filename) as f:
        file_duration = f.frames / f.samplerate

    assert file_duration == pytest.approx(
        recording_duration
    ), "Expected file recording to be of length {}s. It is {}s.".format(
        recording_duration, file_duration)
コード例 #9
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ファイル: octoaudio.py プロジェクト: dcooperdalrymple/octopy
    def run(self):
        while self._destroy == False:
            if (self.active == True) and (self.filepath):
                with WaveReader(self.filepath) as wav:
                    print "Title:", wav.metadata.title
                    print "Artist:", wav.metadata.artist
                    print "Channels:", wav.channels
                    print "Format: 0x%x" % wav.format
                    print "Sample Rate:", wav.samplerate

                    # Set device attributes
                    self.device.setchannels(wav.channels)
                    self.device.setrate(wav.samplerate)

                    data = wav.buffer(self.periodsize)
                    nframes = wav.read(data)
                    while (nframes) and (self.active):
                        self.device.write(data[:, :nframes])
                        nframes = wav.read(data)
                    wav.close()
                self.active = False
            time.sleep(0.1)
コード例 #10
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import threading
import wave
from wavefile import WaveReader
from pyaudio import PyAudio,paInt16
import time

recordFile = 'audio_record/temp0.wav'

with WaveReader(recordFile) as r:
    for data in r.read_iter(size=512):
        left_channel = data[0]
        volume = np.linalg.norm(left_channel)
        print volume
コード例 #11
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ファイル: Sound.py プロジェクト: mendrew/questroom-lovecraft
 def __init__(self, sound_name):
     self.sound_name = sound_name
     self.sound_file = WaveReader(sound_name)
     self.channels = self.sound_file.channels
     self.player_lib = pyaudio.PyAudio()
     self._volume = 1
コード例 #12
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class Sound:
    CHUNK = 512

    def __init__(self, sound_name):
        self.sound_name = sound_name
        self.sound_file = WaveReader(sound_name)
        self.channels = self.sound_file.channels
        self.player_lib = pyaudio.PyAudio()
        self._volume = 1

        # self.wf = wave.open(sound_name, 'rb')
        # self.p = pyaudio.PyAudio()
        # self.device_index, self.max_channels = self.get_valid_device_info(self.p)

    def callback(self, in_data, frame_count, time_info, status):
        # data = self.wf.readframes(frame_count)
        # numpy_array = numpy.fromstring(data, 'int16') * self._volume
        # new_data = numpy_array.astype('int16').tostring()
        new_data = self.sound_file.read_iter(size=512)
        # new_data = self.sound_file.read(frame_count)
        return (new_data, pyaudio.paContinue)

    def open_stream(self):
        # stream = self.p.open(format=self.p.get_format_from_width(self.wf.getsampwidth()),
        #     channels=min(self.wf.getnchannels(), self.max_channels),
        #     output_device_index=1,
        #     rate=self.wf.getframerate(),
        #     output=True)
        # return stream
        #
        stream = self.player_lib.open(format=pyaudio.paFloat32,
                                      channels=self.channels,
                                      rate=self.sound_file.samplerate,
                                      frames_per_buffer=512,
                                      output=True)
        return stream

    @run_in_thread
    def play(self, repeat_count=1):
        stream = self.open_stream()
        stream.start_stream()
        while repeat_count != 0:
            repeat_count -= 1
            # data = self.wf.readframes(self.CHUNK)
            for frame in self.sound_file.read_iter(size=512):
                stream.write(frame, frame.shape[1])

            # self.wf.rewind()
            self.sound_file.seek(0)
            # data = self.wf.readframes(self.CHUNK)

        stream.stop_stream()
        stream.close()
        self.sound_file.close()
        self.player_lib.terminate()

    @property
    def volume(self):
        return self._volume

    @volume.setter
    def volume(self, value):
        self._volume = value
        self._volume_fraction = Fraction(self._volume).limit_denominator()
コード例 #13
0
                                             "\t\tDOLBY SMPTE 5.1 Channel order (L R C LFE Ls Rs)\n"
                                             "\t\t\t\tto\n"
                                             "\t\tProTools & Film Channel order (L C R Ls Rs LFE)\n"),
                                 formatter_class=RawTextHelpFormatter)
parser.add_argument("-i", help="File/Folder for processing")
args = parser.parse_args()
# Initialize variables
item = args.i
wav_ext = '.wav'

# figure out if arg is folder or file
if os.path.isdir(item):
    for file in os.listdir(item):
        if file.endswith(wav_ext):
            # check to see if file is 5.1 or 7.1 for processing
            read_wav = WaveReader(os.path.join(item, file))
            if read_wav.channels == 6:
                in_file = os.path.join(item, file)
                out_file = os.path.join(item, file[:-4])
                reorderFolder = ffmpy3.FFmpeg(
                    inputs={in_file: None},
                    # set Film order
                    outputs={out_file + '_film.wav': "-rf64 auto -filter "
                                                     "'channelmap=FL-FL|FR-FC|FC-FR|LFE-SR|SL-LFE|SR-SL'"}
                    )
                reorderFolder.run()
            elif read_wav.channels == 8:
                in_file = os.path.join(item, file)
                out_file = os.path.join(item, file[:-4])
                reorderFolder = ffmpy3.FFmpeg(
                    inputs={in_file: None},
コード例 #14
0
                target_gain = (1+threshold-self.envelope)
            else:
                target_gain = 1.0
            self.gain = ( self.gain*self.attack_coeff +
                          target_gain*(1-self.attack_coeff) )

            # limit the delayed signal
            signal[i] = self.delay_line[self.delay_index] * self.gain
            
            
            
if len(sys.argv) < 2:
    print("Plays a wave file.\n\nUsage: %s filename.wav" % sys.argv[0])
    sys.exit(-1)

wf = WaveReader(sys.argv[1])
#wf = wave.open(sys.argv[1], 'rb')

# instantiate PyAudio (1)
p = pyaudio.PyAudio()

limiter = Limiter(attack_coeff, release_coeff, delay, dtype)

def callback(in_data, frame_count, time_info, flag):
    if flag:
        print("Playback Error: %i" % flag)
    data = np.zeros((1,block_length), np.float32, order='F')

    nframes = wf.read(data)
    played_frames = callback.counter
    callback.counter += nframes
コード例 #15
0
import numpy
import pyfftw
import os
from base import *
from wavefile import WaveReader,Format
from time import sleep

SAMPLING_RATE = 44100
CHANNELS = 1
FIFO = "/tmp/mpd.fifo"

os.nice(-20)

sleep(1) # wait for mpd to come up

w = WaveReader(FIFO, SAMPLING_RATE, CHANNELS, Format.PCM_16 | Format.RAW | Format.ENDIAN_LITTLE)
m = 8
colors = bytearray(b'\x00'*900)
i = 0
for a in reversed(range(33, 154, 20)):
    for b in reversed(range(33, 154, 20)):
        if a != b:
            for c in reversed(range(33, 154, 20)):
                if i < 900 and b != c:
                    colors[i] = a; i+=1
                    colors[i] = b; i+=1
                    colors[i] = c; i+=1    

pyfftw.interfaces.cache.enable()
a = pyfftw.empty_aligned(1024)
for data in w.read_iter(size=1024):
コード例 #16
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class Sound:
    CHUNK = 512

    def __init__(self, sound_name):
        self.sound_name = sound_name
        self.sound_file = WaveReader(sound_name)
        self.channels = self.sound_file.channels
        self.player_lib = pyaudio.PyAudio()
        self._volume = 1

        # self.wf = wave.open(sound_name, 'rb')
        # self.p = pyaudio.PyAudio()
        # self.device_index, self.max_channels = self.get_valid_device_info(self.p)

    def callback(self, in_data, frame_count, time_info, status):
        # data = self.wf.readframes(frame_count)
        # numpy_array = numpy.fromstring(data, 'int16') * self._volume
        # new_data = numpy_array.astype('int16').tostring()
        data = self.sound_file.buffer(512)
        nframes = self.sound_file.read(data)
        new_data = data[:, :nframes]

        # new_data = self.sound_file.read(frame_count)

        # new_data = self.sound_file.read_iter(size=frame_count)
        # new_data = self.sound_file.read(frame_count)
        return (new_data, pyaudio.paContinue)

    def open_stream(self):
        # stream = self.p.open(format=self.p.get_format_from_width(self.wf.getsampwidth()),
        #     channels=min(self.wf.getnchannels(), self.max_channels),
        #     output_device_index=1,
        #     rate=self.wf.getframerate(),
        #     output=True)
        # return stream
        #
        stream = self.player_lib.open(format=pyaudio.paFloat32,
                                      channels=self.channels,
                                      rate=self.sound_file.samplerate,
                                      frames_per_buffer=512,
                                      output=True,
                                      stream_callback=self.callback)
        return stream

    @run_in_thread
    def play(self, repeat_count=1):
        stream = self.open_stream()
        stream.start_stream()
        print("File: {} playing\n".format(self.sound_name))
        # while stream.is_active():
        #     time.sleep(0.1)
        #     sys.stdout.write("."); sys.stdout.flush()
        # stream.stop_stream()
        # stream.close()
        # self.sound_file.close()
        # self.player_lib.terminate()

    @property
    def volume(self):
        return self._volume

    @volume.setter
    def volume(self, value):
        self._volume = value
        self._volume_fraction = Fraction(self._volume).limit_denominator()
コード例 #17
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 def addSound(self, sound_name):
     sound_file = WaveReader(sound_name)
     self.playing_files.append(sound_file)
コード例 #18
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 def __init__(self, sound_name):
     self.sound_name = sound_name
     self.sound_file = WaveReader(sound_name)
     self.channels = self.sound_file.channels
     self.player_lib = pyaudio.PyAudio()
     self._volume = 1
コード例 #19
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#!/usr/bin/python
import serial
import numpy
from wavefile import WaveReader, Format
from time import sleep

SAMPLING_RATE = 44100
CHANNELS = 2
FIFO = "/tmp/mpd_stereo.fifo"

NUM_LEDS = 300

ser = serial.Serial('/dev/ttyUSB0', 1000000)

w = WaveReader(FIFO, SAMPLING_RATE, CHANNELS,
               Format.PCM_16 | Format.RAW | Format.ENDIAN_LITTLE)
m = 16
colors = bytearray(b'\x00' * 900)
i = 0
for a in reversed(range(33, 254, 40)):
    for b in reversed(range(33, 254, 40)):
        if a != b:
            for c in reversed(range(33, 254, 40)):
                if i < 900 and b != c:
                    colors[i] = a
                    i += 1
                    colors[i] = b
                    i += 1
                    colors[i] = c
                    i += 1
コード例 #20
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#!/usr/bin/env python

### Processing example

import sys
from wavefile import WaveReader, WaveWriter

with WaveReader(sys.argv[1]) as r:
    with WaveWriter(
            'output.wav',
            channels=r.channels,
            samplerate=r.samplerate,
            ) as w:
        w.metadata.title = r.metadata.title + " II"
        w.metadata.artist = r.metadata.artist

        for data in r.read_iter(size=512):
            sys.stdout.write("."); sys.stdout.flush()
            w.write(.8*data)

# vim: noet ts=4 sw=4
コード例 #21
0
ファイル: Sound.py プロジェクト: mendrew/questroom-lovecraft
class Sound:
    CHUNK = 512

    def __init__(self, sound_name):
        self.sound_name = sound_name
        self.sound_file = WaveReader(sound_name)
        self.channels = self.sound_file.channels
        self.player_lib = pyaudio.PyAudio()
        self._volume = 1


        # self.wf = wave.open(sound_name, 'rb')
        # self.p = pyaudio.PyAudio()
        # self.device_index, self.max_channels = self.get_valid_device_info(self.p)


    def callback(self, in_data, frame_count, time_info, status):
        # data = self.wf.readframes(frame_count)
        # numpy_array = numpy.fromstring(data, 'int16') * self._volume
        # new_data = numpy_array.astype('int16').tostring()
        new_data = self.sound_file.read_iter(size=512)
        # new_data = self.sound_file.read(frame_count)
        return (new_data, pyaudio.paContinue)


    def open_stream(self):
        # stream = self.p.open(format=self.p.get_format_from_width(self.wf.getsampwidth()),
        #     channels=min(self.wf.getnchannels(), self.max_channels),
        #     output_device_index=1,
        #     rate=self.wf.getframerate(),
        #     output=True)
        # return stream
        #
        stream = self.player_lib.open(
            format = pyaudio.paFloat32,
            channels = self.channels,
            rate = self.sound_file.samplerate,
            frames_per_buffer = 512,
            output=True)
        return stream

    @run_in_thread
    def play(self, repeat_count = 1):
        stream = self.open_stream()
        stream.start_stream()
        while repeat_count != 0:
            repeat_count -= 1
            # data = self.wf.readframes(self.CHUNK)
            for frame in self.sound_file.read_iter(size=512) :
                stream.write(frame, frame.shape[1])

            # self.wf.rewind()
            self.sound_file.seek(0)
            # data = self.wf.readframes(self.CHUNK)

        stream.stop_stream()
        stream.close()
        self.sound_file.close()
        self.player_lib.terminate()

    @property
    def volume(self):
        return self._volume

    @volume.setter
    def volume(self, value):
        self._volume = value
        self._volume_fraction = Fraction(self._volume).limit_denominator()