"""
Online ASR with Emformer RNN-T
==============================

**Author**: `Jeff Hwang <jeffhwang@meta.com>`__, `Moto Hira <moto@meta.com>`__

This tutorial shows how to use Emformer RNN-T and streaming API
to perform online speech recognition.

"""

######################################################################
#
# .. note::
#
#    This tutorial requires FFmpeg libraries (>=4.1, <4.4) and SentencePiece.
#
#    There are multiple ways to install FFmpeg libraries.
#    If you are using Anaconda Python distribution,
#    ``conda install 'ffmpeg<4.4'`` will install
#    the required FFmpeg libraries.
#
#    You can install SentencePiece by running ``pip install sentencepiece``.

######################################################################
# 1. Overview
# -----------
#
# Performing online speech recognition is composed of the following steps
#
# 1. Build the inference pipeline
#    Emformer RNN-T is composed of three components: feature extractor,
#    decoder and token processor.
# 2. Format the waveform into chunks of expected sizes.
# 3. Pass data through the pipeline.

######################################################################
# 2. Preparation
# --------------
#

import torch
import torchaudio

print(torch.__version__)
print(torchaudio.__version__)

######################################################################
#
import IPython
import matplotlib.pyplot as plt

try:
    from torchaudio.io import StreamReader
except ModuleNotFoundError:
    try:
        import google.colab

        print(
            """
            To enable running this notebook in Google Colab, install the requisite
            third party libraries by running the following code block:

            !add-apt-repository -y ppa:savoury1/ffmpeg4
            !apt-get -qq install -y ffmpeg
            """
        )
    except ModuleNotFoundError:
        pass
    raise


######################################################################
# 3. Construct the pipeline
# -------------------------
#
# Pre-trained model weights and related pipeline components are
# bundled as :py:class:`torchaudio.pipelines.RNNTBundle`.
#
# We use :py:data:`torchaudio.pipelines.EMFORMER_RNNT_BASE_LIBRISPEECH`,
# which is a Emformer RNN-T model trained on LibriSpeech dataset.
#

bundle = torchaudio.pipelines.EMFORMER_RNNT_BASE_LIBRISPEECH

feature_extractor = bundle.get_streaming_feature_extractor()
decoder = bundle.get_decoder()
token_processor = bundle.get_token_processor()

######################################################################
# Streaming inference works on input data with overlap.
# Emformer RNN-T model treats the newest portion of the input data
# as the "right context" — a preview of future context.
# In each inference call, the model expects the main segment
# to start from this right context from the previous inference call.
# The following figure illustrates this.
#
# .. image:: https://download.pytorch.org/torchaudio/tutorial-assets/emformer_rnnt_context.png
#
# The size of main segment and right context, along with
# the expected sample rate can be retrieved from bundle.
#

sample_rate = bundle.sample_rate
segment_length = bundle.segment_length * bundle.hop_length
context_length = bundle.right_context_length * bundle.hop_length

print(f"Sample rate: {sample_rate}")
print(f"Main segment: {segment_length} frames ({segment_length / sample_rate} seconds)")
print(f"Right context: {context_length} frames ({context_length / sample_rate} seconds)")

######################################################################
# 4. Configure the audio stream
# -----------------------------
#
# Next, we configure the input audio stream using :py:class:`torchaudio.io.StreamReader`.
#
# For the detail of this API, please refer to the
# `StreamReader Basic Usage <./streamreader_basic_tutorial.html>`__.
#

######################################################################
# The following audio file was originally published by LibriVox project,
# and it is in the public domain.
#
# https://librivox.org/great-pirate-stories-by-joseph-lewis-french/
#
# It was re-uploaded for the sake of the tutorial.
#
src = "https://download.pytorch.org/torchaudio/tutorial-assets/greatpiratestories_00_various.mp3"

streamer = StreamReader(src)
streamer.add_basic_audio_stream(frames_per_chunk=segment_length, sample_rate=bundle.sample_rate)

print(streamer.get_src_stream_info(0))
print(streamer.get_out_stream_info(0))

######################################################################
# As previously explained, Emformer RNN-T model expects input data with
# overlaps; however, `Streamer` iterates the source media without overlap,
# so we make a helper structure that caches a part of input data from
# `Streamer` as right context and then appends it to the next input data from
# `Streamer`.
#
# The following figure illustrates this.
#
# .. image:: https://download.pytorch.org/torchaudio/tutorial-assets/emformer_rnnt_streamer_context.png
#


class ContextCacher:
    """Cache the end of input data and prepend the next input data with it.

    Args:
        segment_length (int): The size of main segment.
            If the incoming segment is shorter, then the segment is padded.
        context_length (int): The size of the context, cached and appended.
    """

    def __init__(self, segment_length: int, context_length: int):
        self.segment_length = segment_length
        self.context_length = context_length
        self.context = torch.zeros([context_length])

    def __call__(self, chunk: torch.Tensor):
        if chunk.size(0) < self.segment_length:
            chunk = torch.nn.functional.pad(chunk, (0, self.segment_length - chunk.size(0)))
        chunk_with_context = torch.cat((self.context, chunk))
        self.context = chunk[-self.context_length :]
        return chunk_with_context


######################################################################
# 5. Run stream inference
# -----------------------
#
# Finally, we run the recognition.
#
# First, we initialize the stream iterator, context cacher, and
# state and hypothesis that are used by decoder to carry over the
# decoding state between inference calls.
#

cacher = ContextCacher(segment_length, context_length)

state, hypothesis = None, None

######################################################################
# Next we, run the inference.
#
# For the sake of better display, we create a helper function which
# processes the source stream up to the given times and call it
# repeatedly.
#

stream_iterator = streamer.stream()


def _plot(feats, num_iter, unit=25):
    unit_dur = segment_length / sample_rate * unit
    num_plots = num_iter // unit + (1 if num_iter % unit else 0)
    fig, axes = plt.subplots(num_plots, 1)
    t0 = 0
    for i, ax in enumerate(axes):
        feats_ = feats[i*unit:(i+1)*unit]
        t1 = t0 + segment_length / sample_rate * len(feats_)
        feats_ = torch.cat([f[2:-2] for f in feats_])  # remove boundary effect and overlap
        ax.imshow(feats_.T, extent=[t0, t1, 0, 1], aspect="auto", origin="lower")
        ax.tick_params(which='both', left=False, labelleft=False)
        ax.set_xlim(t0, t0 + unit_dur)
        t0 = t1
    fig.suptitle("MelSpectrogram Feature")
    plt.tight_layout()


@torch.inference_mode()
def run_inference(num_iter=100):
    global state, hypothesis
    chunks = []
    feats = []
    for i, (chunk,) in enumerate(stream_iterator, start=1):
        segment = cacher(chunk[:, 0])
        features, length = feature_extractor(segment)
        hypos, state = decoder.infer(features, length, 10, state=state, hypothesis=hypothesis)
        hypothesis = hypos
        transcript = token_processor(hypos[0][0], lstrip=False)
        print(transcript, end="\r", flush=True)

        chunks.append(chunk)
        feats.append(features)
        if i == num_iter:
            break

    # Plot the features
    _plot(feats, num_iter)
    return IPython.display.Audio(torch.cat(chunks).T.numpy(), rate=bundle.sample_rate)


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#

run_inference()

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run_inference()

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run_inference()

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run_inference()

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#
# Tag: :obj:`torchaudio.io`
