examples : add embd_to_audio to tts-outetts.py

This commit contains a suggestion for adding the missing embd_to_audion
function from tts.cpp to tts-outetts.py. This functions uses numpy which
I was not sure if that is acceptable (only PyTorch was mentioned in
referened PR).

Also the README has been updated with instructions to run the example
with llama-server and the python script.

Refs: ggml-org#10784 (comment)

Author: danbev

examples/tts/README.md

@@ -78,3 +78,38 @@ play the audio:
 $ aplay output.wav
 ```
 
+### Running the example with llama-server
+Running this example with `llama-server` is also possible and requires two
+server instances to be started. One will server the LLM model and the other
+will serve the voice decoder model.
+
+The LLM model server can be started with the following command:
+```console
+$ ./build/bin/llama-server -m ./models/outetts-0.2-0.5B-q8_0.gguf --port 8020
+```
+And the voice decoder model server can be started using:
+```console
+./build/bin/llama-server -m ./models/wavtokenizer-large-75-f16.gguf --port 8021 --embeddings --pooling none
+```
+Then we can run the following python script to generate the audio.
+
+First create a virtual environment for python and install the required
+dependencies (this in only required to be done once):
+```console
+$ python3 -m venv venv
+$ source venv/bin/activate
+(venv) pip install requests numpy
+```
+
+And then run the script using:
+```conole
+(venv) python ./examples/tts/tts-outetts.py http://localhost:8020 http://localhost:8021 "Hello world"
+spectrogram generated: n_codes: 90, n_embd: 1282
+converting to audio ...
+audio generated: 28800 samples
+audio written to file "output.wav"
+```
+And to play the audio we can again use aplay or any other media player:
+```console
+$ aplay output.wav
+```

examples/tts/tts-outetts.py

@@ -3,6 +3,122 @@
 #import struct
 import requests
 import re
+import struct
+import numpy as np
+from concurrent.futures import ThreadPoolExecutor
+
+
+def fill_hann_window(size, periodic=True):
+    if periodic:
+        return np.hanning(size + 1)[:-1]
+    return np.hanning(size)
+
+
+def irfft(n_fft, complex_input):
+    # Ensure proper shape for IFFT
+    return np.fft.irfft(complex_input, n=n_fft)
+
+
+def fold(buffer, n_out, n_win, n_hop, n_pad):
+    result = np.zeros(n_out)
+    n_frames = len(buffer) // n_win
+
+    for i in range(n_frames):
+        start = i * n_hop
+        end = start + n_win
+        result[start:end] += buffer[i * n_win:(i + 1) * n_win]
+
+    return result[n_pad:-n_pad] if n_pad > 0 else result
+
+
+def process_frame(args):
+    l, n_fft, ST, hann = args
+    frame = irfft(n_fft, ST[l])
+    frame = frame * hann
+    hann2 = hann * hann
+    return frame, hann2
+
+
+def embd_to_audio(embd, n_codes, n_embd, n_thread=4):
+    embd = np.asarray(embd, dtype=np.float32).reshape(n_codes, n_embd)
+
+    n_fft = 1280
+    n_hop = 320
+    n_win = 1280
+    n_pad = (n_win - n_hop) // 2
+    n_out = (n_codes - 1) * n_hop + n_win
+
+    hann = fill_hann_window(n_fft, True)
+
+    E = np.zeros((n_embd, n_codes), dtype=np.float32)
+    for l in range(n_codes):
+        for k in range(n_embd):
+            E[k, l] = embd[l, k]
+
+    half_embd = n_embd // 2
+    S = np.zeros((n_codes, half_embd + 1), dtype=np.complex64)
+
+    for k in range(half_embd):
+        for l in range(n_codes):
+            mag = E[k, l]
+            phi = E[k + half_embd, l]
+
+            mag = np.clip(np.exp(mag), 0, 1e2)
+            S[l, k] = mag * np.exp(1j * phi)
+
+    res = np.zeros(n_codes * n_fft)
+    hann2_buffer = np.zeros(n_codes * n_fft)
+
+    with ThreadPoolExecutor(max_workers=n_thread) as executor:
+        args = [(l, n_fft, S, hann) for l in range(n_codes)]
+        results = list(executor.map(process_frame, args))
+
+        for l, (frame, hann2) in enumerate(results):
+            res[l*n_fft:(l+1)*n_fft] = frame
+            hann2_buffer[l*n_fft:(l+1)*n_fft] = hann2
+
+    audio = fold(res, n_out, n_win, n_hop, n_pad)
+    env = fold(hann2_buffer, n_out, n_win, n_hop, n_pad)
+
+    mask = env > 1e-10
+    audio[mask] /= env[mask]
+
+    return audio
+
+
+def save_wav(filename, audio_data, sample_rate):
+    num_channels = 1
+    bits_per_sample = 16
+    bytes_per_sample = bits_per_sample // 8
+    data_size = len(audio_data) * bytes_per_sample
+    byte_rate = sample_rate * num_channels * bytes_per_sample
+    block_align = num_channels * bytes_per_sample
+    chunk_size = 36 + data_size  # 36 = size of header minus first 8 bytes
+
+    header = struct.pack(
+        '<4sI4s4sIHHIIHH4sI',
+        b'RIFF',
+        chunk_size,
+        b'WAVE',
+        b'fmt ',
+        16,                # fmt chunk size
+        1,                 # audio format (PCM)
+        num_channels,
+        sample_rate,
+        byte_rate,
+        block_align,
+        bits_per_sample,
+        b'data',
+        data_size
+    )
+
+    audio_data = np.clip(audio_data * 32767, -32768, 32767)
+    pcm_data = audio_data.astype(np.int16)
+
+    with open(filename, 'wb') as f:
+        f.write(header)
+        f.write(pcm_data.tobytes())
+
 
 def process_text(text: str):
     text = re.sub(r'\d+(\.\d+)?', lambda x: x.group(), text.lower()) # TODO this needs to be fixed
@@ -170,6 +286,15 @@ def process_text(text: str):
 print('spectrogram generated: n_codes: %d, n_embd: %d' % (n_codes, n_embd))
 
 # post-process the spectrogram to convert to audio
-# TODO: see the tts.cpp:embd_to_audio() and implement it in Python
 print('converting to audio ...')
-print('TODO: see the tts.cpp:embd_to_audio() and implement it in Python')
+audio = embd_to_audio(embd, n_codes, n_embd)
+print('audio generated: %d samples' % len(audio))
+
+filename = "output.wav"
+sample_rate = 24000 # sampling rate
+
+# zero out first 0.25 seconds
+audio[:24000 // 4] = 0.0
+
+save_wav(filename, audio, sample_rate)
+print('audio written to file "%s"' % filename)