#!/usr/bin/env python3
# Copyright (c) 2017-present, Facebook, Inc.
# All rights reserved.
#
# This source code is licensed under the license found in the LICENSE file in
# the root directory of this source tree. An additional grant of patent rights
# can be found in the PATENTS file in the same directory.
"""
Following `a simple but efficient real-time voice activity detection algorithm
<https://www.eurasip.org/Proceedings/Eusipco/Eusipco2009/contents/papers/1569192958.pdf>`__.

There are three criteria to decide if a frame contains speech: energy, most
dominant frequency, and spectral flatness. If any two of those are higher than
a minimum plus a threshold, then the frame contains speech.  In the offline
case, the list of frames is postprocessed to remove too short silence and
speech sequences. In the online case here, inertia is added before switching
from speech to silence or vice versa.
"""

import queue
from collections import deque

import librosa
import numpy as np
import pyaudio
import torch
import torchaudio


def compute_spectral_flatness(frame, epsilon=0.01):
    # epsilon protects against log(0)
    geometric_mean = torch.exp((frame + epsilon).log().mean(-1)) - epsilon
    arithmetic_mean = frame.mean(-1)
    return -10 * torch.log10(epsilon + geometric_mean / arithmetic_mean)


class VoiceActivityDetection:
    def __init__(
        self,
        num_init_frames=30,
        ignore_silent_count=4,
        ignore_speech_count=1,
        energy_prim_thresh=60,
        frequency_prim_thresh=10,
        spectral_flatness_prim_thresh=3,
        verbose=False,
    ):

        self.num_init_frames = num_init_frames
        self.ignore_silent_count = ignore_silent_count
        self.ignore_speech_count = ignore_speech_count

        self.energy_prim_thresh = energy_prim_thresh
        self.frequency_prim_thresh = frequency_prim_thresh
        self.spectral_flatness_prim_thresh = spectral_flatness_prim_thresh

        self.verbose = verbose

        self.speech_mark = True
        self.silence_mark = False

        self.silent_count = 0
        self.speech_count = 0
        self.n = 0

        if self.verbose:
            self.energy_list = []
            self.frequency_list = []
            self.spectral_flatness_list = []

    def iter(self, frame):

        frame_fft = torch.rfft(frame, 1)
        amplitudes = torchaudio.functional.complex_norm(frame_fft)

        # Compute frame energy
        energy = frame.pow(2).sum(-1)

        # Most dominant frequency component
        frequency = amplitudes.argmax()

        # Spectral flatness measure
        spectral_flatness = compute_spectral_flatness(amplitudes)

        if self.verbose:
            self.energy_list.append(energy)
            self.frequency_list.append(frequency)
            self.spectral_flatness_list.append(spectral_flatness)

        if self.n == 0:
            self.min_energy = energy
            self.min_frequency = frequency
            self.min_spectral_flatness = spectral_flatness
        elif self.n < self.num_init_frames:
            self.min_energy = min(energy, self.min_energy)
            self.min_frequency = min(frequency, self.min_frequency)
            self.min_spectral_flatness = min(spectral_flatness, self.min_spectral_flatness)

        self.n += 1

        # Add 1. to avoid log(0)
        thresh_energy = self.energy_prim_thresh * torch.log(1.0 + self.min_energy)
        thresh_frequency = self.frequency_prim_thresh
        thresh_spectral_flatness = self.spectral_flatness_prim_thresh

        # Check all three conditions

        counter = 0
        if energy - self.min_energy >= thresh_energy:
            counter += 1
        if frequency - self.min_frequency >= thresh_frequency:
            counter += 1
        if spectral_flatness - self.min_spectral_flatness >= thresh_spectral_flatness:
            counter += 1

        # Detection
        if counter > 1:
            # Speech detected
            self.speech_count += 1
            # Inertia against switching
            if self.n >= self.num_init_frames and self.speech_count <= self.ignore_speech_count:
                # Too soon to change
                return self.silence_mark
            else:
                self.silent_count = 0
                return self.speech_mark
        else:
            # Silence detected
            self.min_energy = ((self.silent_count * self.min_energy) + energy) / (self.silent_count + 1)
            self.silent_count += 1
            # Inertia against switching
            if self.n >= self.num_init_frames and self.silent_count <= self.ignore_silent_count:
                # Too soon to change
                return self.speech_mark
            else:
                self.speech_count = 0
                return self.silence_mark


class MicrophoneStream:
    """Opens a recording stream as a generator yielding the audio chunks."""

    def __init__(self, device=None, rate=22050, chunk=2205):
        """
        The 22050 is the librosa default, which is what our models were
        trained on.  The ratio of [chunk / rate] is the amount of time between
        audio samples - for example, with these defaults,
        an audio fragment will be processed every tenth of a second.
        """
        self._rate = rate
        self._chunk = chunk
        self._device = device

        # Create a thread-safe buffer of audio data
        self._buff = queue.Queue()
        self.closed = True

    def __enter__(self):
        self._audio_interface = pyaudio.PyAudio()
        self._audio_stream = self._audio_interface.open(
            # format=pyaudio.paInt16,
            format=pyaudio.paFloat32,
            # The API currently only supports 1-channel (mono) audio
            # https://goo.gl/z757pE
            channels=1,
            rate=self._rate,
            input=True,
            frames_per_buffer=self._chunk,
            input_device_index=self._device,
            # Run the audio stream asynchronously to fill the buffer object.
            # This is necessary so that the input device's buffer doesn't
            # overflow while the calling thread makes network requests, etc.
            stream_callback=self._fill_buffer,
        )

        self.closed = False

        return self

    def __exit__(self, type, value, traceback):
        self._audio_stream.stop_stream()
        self._audio_stream.close()
        self.closed = True
        # Signal the generator to terminate so that the client's
        # streaming_recognize method will not block the process termination.
        self._buff.put(None)
        self._audio_interface.terminate()

    def _fill_buffer(self, in_data, frame_count, time_info, status_flags):
        """Continuously collect data from the audio stream, into the buffer."""
        self._buff.put(in_data)
        return None, pyaudio.paContinue

    def generator(self):
        while not self.closed:
            # Use a blocking get() to ensure there's at least one chunk of
            # data, and stop iteration if the chunk is None, indicating the
            # end of the audio stream.
            chunk = self._buff.get()
            if chunk is None:
                return
            data = [chunk]

            # Now consume whatever other data's still buffered.
            while True:
                try:
                    chunk = self._buff.get(block=False)
                    if chunk is None:
                        return
                    data.append(chunk)
                except queue.Empty:
                    break

            ans = np.fromstring(b"".join(data), dtype=np.float32)
            # yield uniform-sized chunks
            ans = np.split(ans, np.shape(ans)[0] / self._chunk)
            # Resample the audio to 22050, librosa default
            for chunk in ans:
                yield librosa.core.resample(chunk, self._rate, 22050)


def get_microphone_chunks(
    min_to_cumulate=5,  # 0.5 seconds
    max_to_cumulate=100,  # 10 seconds
    precumulate=5,
    max_to_visualize=100,
):

    vad = VoiceActivityDetection()

    cumulated = []
    precumulated = deque(maxlen=precumulate)

    with MicrophoneStream() as stream:
        audio_generator = stream.generator()
        chunk_length = stream._chunk
        waveform = torch.zeros(max_to_visualize * chunk_length)

        for chunk in audio_generator:
            # Is speech?

            chunk = torch.tensor(chunk)
            is_speech = vad.iter(chunk)

            # Cumulate speech

            if is_speech or cumulated:
                cumulated.append(chunk)
            else:
                precumulated.append(chunk)

            if (not is_speech and len(cumulated) >= min_to_cumulate) or (len(cumulated) > max_to_cumulate):
                waveform = torch.cat(list(precumulated) + cumulated, -1)
                yield (waveform * stream._rate, stream._rate)
                cumulated = []
                precumulated = deque(maxlen=precumulate)
