Add tts python example and change onnx to paddle (#420)

* add tts example

* update example

* update use fd engine

* add tts python example

* add readme

* fix comment

* change paddle model

* fix readme style

Co-authored-by: Jason <jiangjiajun@baidu.com>

examples/audio/pp-tts/python/README.md

@@ -0,0 +1,27 @@
+([简体中文](./README_cn.md)|English)
+
+# PP-TTS Streaming Text-to-Speech Python Example
+
+## Introduction
+This demo is an implementation of starting the streaming speech synthesis.
+
+## Usage
+
+### 1. Installation
+```bash
+apt-get install libsndfile1 wget zip
+For Centos, yum install libsndfile-devel wget zip
+python3 -m pip install --upgrade pip
+pip3 install -U fastdeploy-python -f https://www.paddlepaddle.org.cn/whl/fastdeploy.html
+pip3 install -U paddlespeech paddlepaddle soundfile matplotlib
+```
+
+### 2. Run the example
+```bash
+python3 stream_play_tts.py
+```
+
+### 3. Result
+The complete voice synthesis audio is saved as `demo_stream.wav`.
+
+User can install `pyaudio` on their own terminals to play the results of speech synthesis in real time. The relevant code is in `stream_play_tts.py` and you can debug and run it yourself.

examples/audio/pp-tts/python/README_cn.md

@@ -0,0 +1,26 @@
+(简体中文|[English](./README.md))
+
+# PP-TTS流式语音合成Python示例
+
+## 介绍
+本文介绍了使用FastDeploy运行流式语音合成的示例.
+
+## 使用
+### 1. 安装
+```bash
+apt-get install libsndfile1 wget zip
+对于Centos系统,使用yum install libsndfile-devel wget zip
+python3 -m pip install --upgrade pip
+pip3 install -U fastdeploy-python -f https://www.paddlepaddle.org.cn/whl/fastdeploy.html
+pip3 install -U paddlespeech paddlepaddle soundfile matplotlib
+```
+
+### 2. 运行示例
+```bash
+python3 stream_play_tts.py
+```
+
+### 3. 运行效果
+完整的语音合成音频被保存为`demo_stream.wav`文件.
+
+用户可以在自己的终端上安装pyaudio, 对语音合成的结果进行实时播放, 相关代码在stream_play_tts.py处于注释状态, 用户可自行调试运行.

examples/audio/pp-tts/python/stream_play_tts.py

@@ -0,0 +1,214 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+import math
+import os
+import time
+
+import fastdeploy as fd
+import numpy as np
+import soundfile as sf
+
+from paddlespeech.server.utils.util import denorm
+from paddlespeech.server.utils.util import get_chunks
+from paddlespeech.t2s.frontend.zh_frontend import Frontend
+
+model_name_fastspeech2 = "fastspeech2_cnndecoder_csmsc_streaming_static_1.0.0"
+model_zip_fastspeech2 = model_name_fastspeech2 + ".zip"
+model_url_fastspeech2 = "https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/" + model_zip_fastspeech2
+model_name_mb_melgan = "mb_melgan_csmsc_static_0.1.1"
+model_zip_mb_melgan = model_name_mb_melgan + ".zip"
+model_url_mb_melgan = "https://paddlespeech.bj.bcebos.com/Parakeet/released_models/mb_melgan/" + model_zip_mb_melgan
+
+dir_name = os.path.dirname(os.path.realpath(__file__)) + "/"
+
+if not os.path.exists(model_name_fastspeech2):
+    if os.path.exists(model_zip_fastspeech2):
+        os.remove(model_zip_fastspeech2)
+    fd.download_and_decompress(model_url_fastspeech2, path=dir_name)
+    os.remove(model_zip_fastspeech2)
+if not os.path.exists(model_name_mb_melgan):
+    if os.path.exists(model_zip_mb_melgan):
+        os.remove(model_zip_mb_melgan)
+    fd.download_and_decompress(model_url_mb_melgan, path=dir_name)
+    os.remove(model_zip_mb_melgan)
+
+voc_block = 36
+voc_pad = 14
+am_block = 72
+am_pad = 12
+voc_upsample = 300
+
+# 模型路径
+
+phones_dict = dir_name + model_name_fastspeech2 + "/phone_id_map.txt"
+am_stat_path = dir_name + model_name_fastspeech2 + "/speech_stats.npy"
+
+am_encoder_model = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_encoder_infer.pdmodel"
+am_decoder_model = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_decoder.pdmodel"
+am_postnet_model = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_postnet.pdmodel"
+voc_melgan_model = dir_name + model_name_mb_melgan + "/mb_melgan_csmsc.pdmodel"
+
+am_encoder_para = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_encoder_infer.pdiparams"
+am_decoder_para = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_decoder.pdiparams"
+am_postnet_para = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_postnet.pdiparams"
+voc_melgan_para = dir_name + model_name_mb_melgan + "/mb_melgan_csmsc.pdiparams"
+
+frontend = Frontend(phone_vocab_path=phones_dict, tone_vocab_path=None)
+am_mu, am_std = np.load(am_stat_path)
+
+option_1 = fd.RuntimeOption()
+option_1.set_model_path(am_encoder_model, am_encoder_para)
+option_1.use_cpu()
+option_1.use_ort_backend()
+option_1.set_cpu_thread_num(12)
+am_encoder_runtime = fd.Runtime(option_1)
+
+option_2 = fd.RuntimeOption()
+option_2.set_model_path(am_decoder_model, am_decoder_para)
+option_2.use_cpu()
+option_2.use_ort_backend()
+option_2.set_cpu_thread_num(12)
+am_decoder_runtime = fd.Runtime(option_2)
+
+option_3 = fd.RuntimeOption()
+option_3.set_model_path(am_postnet_model, am_postnet_para)
+option_3.use_cpu()
+option_3.use_ort_backend()
+option_3.set_cpu_thread_num(12)
+am_postnet_runtime = fd.Runtime(option_3)
+
+option_4 = fd.RuntimeOption()
+option_4.set_model_path(voc_melgan_model, voc_melgan_para)
+option_4.use_cpu()
+option_4.use_ort_backend()
+option_4.set_cpu_thread_num(12)
+voc_melgan_runtime = fd.Runtime(option_4)
+
+
+def depadding(data, chunk_num, chunk_id, block, pad, upsample):
+    """ 
+    Streaming inference removes the result of pad inference
+    """
+    front_pad = min(chunk_id * block, pad)
+    # first chunk
+    if chunk_id == 0:
+        data = data[:block * upsample]
+    # last chunk
+    elif chunk_id == chunk_num - 1:
+        data = data[front_pad * upsample:]
+    # middle chunk
+    else:
+        data = data[front_pad * upsample:(front_pad + block) * upsample]
+
+    return data
+
+
+def inference_stream(text):
+    input_ids = frontend.get_input_ids(
+        text, merge_sentences=False, get_tone_ids=False)
+    phone_ids = input_ids["phone_ids"]
+    for i in range(len(phone_ids)):
+        part_phone_ids = phone_ids[i].numpy()
+        voc_chunk_id = 0
+        orig_hs = am_encoder_runtime.infer({
+            'text':
+            part_phone_ids.astype("int64")
+        })
+        orig_hs = orig_hs[0]
+
+        # streaming voc chunk info
+        mel_len = orig_hs.shape[1]
+        voc_chunk_num = math.ceil(mel_len / voc_block)
+        start = 0
+        end = min(voc_block + voc_pad, mel_len)
+
+        # streaming am
+        hss = get_chunks(orig_hs, am_block, am_pad, "am")
+        am_chunk_num = len(hss)
+        for i, hs in enumerate(hss):
+            am_decoder_output = am_decoder_runtime.infer({
+                'xs':
+                hs.astype("float32")
+            })
+
+            am_postnet_output = am_postnet_runtime.infer({
+                'xs':
+                np.transpose(am_decoder_output[0], (0, 2, 1))
+            })
+            am_output_data = am_decoder_output + np.transpose(
+                am_postnet_output[0], (0, 2, 1))
+            normalized_mel = am_output_data[0][0]
+
+            sub_mel = denorm(normalized_mel, am_mu, am_std)
+            sub_mel = depadding(sub_mel, am_chunk_num, i, am_block, am_pad, 1)
+
+            if i == 0:
+                mel_streaming = sub_mel
+            else:
+                mel_streaming = np.concatenate((mel_streaming, sub_mel), axis=0)
+
+            # streaming voc
+            # 当流式AM推理的mel帧数大于流式voc推理的chunk size，开始进行流式voc 推理
+            while (mel_streaming.shape[0] >= end and
+                   voc_chunk_id < voc_chunk_num):
+                voc_chunk = mel_streaming[start:end, :]
+
+                sub_wav = voc_melgan_runtime.infer({
+                    'logmel':
+                    voc_chunk.astype("float32")
+                })
+                sub_wav = depadding(sub_wav[0], voc_chunk_num, voc_chunk_id,
+                                    voc_block, voc_pad, voc_upsample)
+
+                yield sub_wav
+
+                voc_chunk_id += 1
+                start = max(0, voc_chunk_id * voc_block - voc_pad)
+                end = min((voc_chunk_id + 1) * voc_block + voc_pad, mel_len)
+
+
+if __name__ == '__main__':
+    text = "欢迎使用飞桨语音合成系统，测试一下合成效果。"
+    # warm up
+    # onnxruntime 第一次时间会长一些，建议先 warmup 一下
+    '''
+    # pyaudio 播放
+    p = pyaudio.PyAudio()
+    stream = p.open(
+        format=p.get_format_from_width(2),  # int16
+        channels=1,
+        rate=24000,
+        output=True)
+    '''
+    # 计时
+    wavs = []
+    t1 = time.time()
+    for sub_wav in inference_stream(text):
+        print("响应时间：", time.time() - t1)
+        t1 = time.time()
+        wavs.append(sub_wav.flatten())
+        # float32 to int16
+        #wav = float2pcm(sub_wav)
+        # to bytes  
+        #wav_bytes = wav.tobytes()
+        #stream.write(wav_bytes)
+
+    # 关闭 pyaudio 播放器
+    #stream.stop_stream()
+    #stream.close()
+    #p.terminate()
+
+    # 流式合成的结果导出
+    wav = np.concatenate(wavs)
+    sf.write("demo_stream.wav", data=wav, samplerate=24000)