fix&feat(gui): 修复ASIO输出爆音, 增加不同种API音频设备支持

See https://github.com/RVC-Project/Retrieval-based-Voice-Conversion-WebUI/pull/2591

gui.py

@@ -76,8 +76,11 @@ if __name__ == "__main__":
     import json
     import multiprocessing
     import re
+    import threading
     import time
     from multiprocessing import Queue, cpu_count
+    from infer.lib.audio import AudioIoProcess
+    from multiprocessing.shared_memory import SharedMemory
 
     import librosa
     from infer.modules.gui import TorchGate
@@ -144,6 +147,15 @@ if __name__ == "__main__":
             self.input_devices_indices = None
             self.output_devices_indices = None
             self.stream = None
+            self.in_mem = None
+            self.out_mem = None
+            self.in_buf = None
+            self.out_buf = None
+            self.in_ptr = None
+            self.out_ptr = None
+            self.play_ptr = None
+            self.in_evt = None
+            self.stop_evt = None
             self.update_devices()
             self.launcher()
 
@@ -564,7 +576,7 @@ if __name__ == "__main__":
                 event, values = self.window.read()
                 if event == sg.WINDOW_CLOSED:
                     self.stop_stream()
-                    exit()
+                    # exit()
                 if event == "reload_devices" or event == "sg_hostapi":
                     self.gui_config.sg_hostapi = values["sg_hostapi"]
                     self.update_devices(hostapi_name=values["sg_hostapi"])
@@ -639,7 +651,7 @@ if __name__ == "__main__":
                             json.dump(settings, j)
                         if self.stream is not None:
                             self.delay_time = (
-                                self.stream.latency[-1]
+                                self.stream.get_latency()
                                 + values["block_time"]
                                 + values["crossfade_length"]
                                 + 0.01
@@ -667,7 +679,7 @@ if __name__ == "__main__":
                         self.rvc.set_index_rate(values["index_rate"])
                 elif event == "rms_mix_rate":
                     self.gui_config.rms_mix_rate = values["rms_mix_rate"]
-                elif event in ["pm", "dio", "harvest", "crepe", "rmvpe", "fcpe"]:
+                elif event in ["pm", "harvest", "crepe", "rmvpe", "fcpe"]:
                     self.gui_config.f0method = event
                 elif event == "I_noise_reduce":
                     self.gui_config.I_noise_reduce = values["I_noise_reduce"]
@@ -867,36 +879,80 @@ if __name__ == "__main__":
                     "WASAPI" in self.gui_config.sg_hostapi
                     and self.gui_config.sg_wasapi_exclusive
                 ):
-                    extra_settings = sd.WasapiSettings(exclusive=True)
+                    wasapi_exclusive = True
                 else:
-                    extra_settings = None
-                self.stream = sd.Stream(
-                    callback=self.audio_callback,
-                    blocksize=self.block_frame,
-                    samplerate=self.gui_config.samplerate,
-                    channels=self.gui_config.channels,
-                    dtype="float32",
-                    extra_settings=extra_settings,
+                    wasapi_exclusive = False
+                self.stream = AudioIoProcess(
+                    input_device=sd.default.device[0],
+                    output_device=sd.default.device[1],
+                    input_audio_block_size = self.block_frame,
+                    sample_rate = self.gui_config.samplerate,
+                    channel_num=self.gui_config.channels,
+                    is_input_wasapi_exclusive=wasapi_exclusive,
+                    is_output_wasapi_exclusive=wasapi_exclusive,
+                    is_device_combined = True
+                    # TODO: Add control UI to allow devices with different type API & different WASAPI settings
                 )
+                self.in_mem = SharedMemory(name=self.stream.get_in_mem_name())
+                self.out_mem = SharedMemory(name=self.stream.get_out_mem_name())
+                self.in_buf = np.ndarray(
+                    self.stream.get_np_shape(),
+                    dtype=self.stream.get_np_dtype(),
+                    buffer=self.in_mem.buf,
+                    order='C'
+                )
+                self.out_buf = np.ndarray(
+                    self.stream.get_np_shape(),
+                    dtype=self.stream.get_np_dtype(),
+                    buffer=self.out_mem.buf,
+                    order='C'
+                )
+                self.in_ptr, \
+                self.out_ptr, \
+                self.play_ptr, \
+                self.in_evt, \
+                self.stop_evt = self.stream.get_ptrs_and_events()
+
                 self.stream.start()
 
+                def audio_loop():
+                    while flag_vc:
+                        self.audio_infer(self.block_frame << 1)
+
+                threading.Thread(
+                    target=audio_loop,
+                    daemon=True
+                ).start()
+
         def stop_stream(self):
             global flag_vc
             if flag_vc:
                 flag_vc = False
                 if self.stream is not None:
-                    self.stream.abort()
-                    self.stream.close()
+                    print("Exiting")
+                    self.stop_evt.set()
+                    self.in_mem.close()
+                    self.out_mem.close()
+                    self.stream.join()
                     self.stream = None
 
-        def audio_callback(
-            self, indata: np.ndarray, outdata: np.ndarray, frames, times, status
+        def audio_infer(
+            self, buf_size:int # 2 * self.block_frame
         ):
             """
             音频处理
             """
             global flag_vc
+
+            self.in_evt.wait()
+            rptr = self.in_ptr.value
+            self.in_evt.clear()
+            
             start_time = time.perf_counter()
+
+            rend = rptr + self.block_frame
+            indata = np.copy(self.in_buf[rptr:rend])
+
             indata = librosa.to_mono(indata.T)
             if self.gui_config.threhold > -60:
                 indata = np.append(self.rms_buffer, indata)
@@ -1039,13 +1095,47 @@ if __name__ == "__main__":
             self.sola_buffer[:] = infer_wav[
                 self.block_frame : self.block_frame + self.sola_buffer_frame
             ]
-            outdata[:] = (
+            outdata = (
                 infer_wav[: self.block_frame]
                 .repeat(self.gui_config.channels, 1)
                 .t()
                 .cpu()
                 .numpy()
             )
+
+            # 装填输出缓冲
+            start = self.out_ptr.value
+            play_pos = self.play_ptr.value
+
+            # 计算播放进度差（写指针距离播放指针的帧数）
+            delta = (start - play_pos + buf_size) % buf_size
+
+            if delta < self.block_frame:
+                # 装填赶不上播放，导致播放进度追上来了，
+                # 此时已产生无法挽回的破音，
+                # 只好直接卡着播放指针写入，保证接下来的尽快放出来
+                print("[W] Output underrun")
+                write_pos = play_pos
+            else:
+                # 否则按块对齐
+                write_pos = (start + self.block_frame) % buf_size
+
+            # 写入共享缓冲区
+            end = (write_pos + self.block_frame) % buf_size
+            if end > write_pos:
+                self.out_buf[write_pos:end] = outdata
+            else:
+                first = buf_size - write_pos
+                self.out_buf[write_pos:] = outdata[:first]
+                self.out_buf[:end] = outdata[first:]
+
+            # 更新写指针
+            self.out_ptr.value = write_pos
+
+            if self.in_evt.is_set():
+                print("[W] Input overrun")
+                self.in_evt.clear()
+
             total_time = time.perf_counter() - start_time
             if flag_vc:
                 self.window["infer_time"].update(int(total_time * 1000))

infer/lib/audio.py

@@ -4,6 +4,9 @@ from typing import Dict, Tuple, Optional, Union, List
 import os
 import math
 import wave
+import signal
+from multiprocessing import Process, Value, Event
+from multiprocessing.shared_memory import SharedMemory
 
 import numpy as np
 from numba import jit
@@ -195,3 +198,167 @@ def get_audio_properties(input_path: str) -> Tuple[int, int]:
     rate = audio_stream.base_rate
     container.close()
     return channels, rate
+
+class AudioIoProcess(Process):
+    def __init__(self,
+                 input_device,
+                 output_device,
+                 input_audio_block_size: int,
+                 sample_rate: int,
+                 channel_num: int = 2,
+                 is_device_combined: bool = True,
+                 is_input_wasapi_exclusive: bool = False,
+                 is_output_wasapi_exclusive: bool = False
+                 ):
+        super().__init__()
+        self.in_dev = input_device
+        self.out_dev = output_device
+        self.block_size: int = input_audio_block_size
+        self.buf_size: int = self.block_size << 1  # 双缓冲
+        self.sample_rate: int = sample_rate
+        self.channels: int = channel_num
+        self.is_device_combined: bool = is_device_combined
+        self.is_input_wasapi_exclusive: bool = is_input_wasapi_exclusive
+        self.is_output_wasapi_exclusive: bool = is_output_wasapi_exclusive
+
+        self.__rec_ptr = 0
+        self.in_ptr = Value('i', 0)  # 当收满一个block时由本进程设置
+        self.out_ptr = Value('i', 0)  # 由主进程设置，指示下一次预期写入位置
+        self.play_ptr = Value('i', 0)  # 由本进程设置，指示当前音频已经播放到哪里
+        self.in_evt = Event()  # 当收满一个block时由本进程设置
+        self.stop_evt = Event()  # 当主进程停止音频活动时由主进程设置
+
+        self.latency = Value('d', 114514.1919810)
+
+        self.buf_shape: tuple = (self.buf_size, self.channels)
+        self.buf_dtype: np.dtype = np.float32
+        self.buf_nbytes: int = int(
+            np.prod(self.buf_shape) * np.dtype(self.buf_dtype).itemsize)
+
+        self.in_mem = SharedMemory(create=True, size=self.buf_nbytes)
+        self.out_mem = SharedMemory(create=True, size=self.buf_nbytes)
+        self.in_mem_name: str = self.in_mem.name
+        self.out_mem_name: str = self.out_mem.name
+
+        self.in_buf = None
+        self.out_buf = None
+
+    def get_in_mem_name(self) -> str:
+        return self.in_mem_name
+
+    def get_out_mem_name(self) -> str:
+        return self.out_mem_name
+
+    def get_np_shape(self) -> tuple:
+        return self.buf_shape
+
+    def get_np_dtype(self) -> np.dtype:
+        return self.buf_dtype
+
+    def get_ptrs_and_events(self):
+        return self.in_ptr, \
+            self.out_ptr,\
+            self.play_ptr,\
+            self.in_evt, \
+            self.stop_evt\
+
+    def get_latency(self) -> float:
+        return self.latency.value
+
+    def run(self):
+        import sounddevice as sd
+
+        signal.signal(signal.SIGINT, signal.SIG_IGN)
+
+        in_mem = SharedMemory(name=self.in_mem_name)
+        self.in_buf = np.ndarray(
+            self.buf_shape, dtype=self.buf_dtype, buffer=in_mem.buf, order='C')
+        self.in_buf.fill(0.0)
+
+        out_mem = SharedMemory(name=self.out_mem_name)
+        self.out_buf = np.ndarray(
+            self.buf_shape, dtype=self.buf_dtype, buffer=out_mem.buf, order='C')
+        self.out_buf.fill(0.0)
+
+        exclusive_settings = sd.WasapiSettings(exclusive=True)
+
+        sd.default.device = (self.in_dev, self.out_dev)
+
+        def output_callback(outdata, frames, time_info, status):
+            play_ptr = self.play_ptr.value
+            end_ptr = play_ptr + frames
+
+            if end_ptr <= self.buf_size:
+                outdata[:] = self.out_buf[play_ptr:end_ptr]
+            else:
+                first = self.buf_size - play_ptr
+                second = end_ptr - self.buf_size
+                outdata[:first] = self.out_buf[play_ptr:]
+                outdata[first:] = self.out_buf[:second]
+
+            self.play_ptr.value = end_ptr % self.buf_size
+
+        def input_callback(indata, frames, time_info, status):
+            # 收录输入数据
+            end_ptr = self.__rec_ptr + frames
+            if end_ptr <= self.buf_size:  # 整块拷贝
+                self.in_buf[self.__rec_ptr:end_ptr] = indata
+            else:  # 处理回绕
+                first = self.buf_size - self.__rec_ptr
+                second = end_ptr - self.buf_size
+                self.in_buf[self.__rec_ptr:] = indata[:first]
+                self.in_buf[:second] = indata[first:]
+            write_pos = self.__rec_ptr
+            self.__rec_ptr = end_ptr % self.buf_size
+
+            # 设置信号
+            if write_pos < self.block_size and self.__rec_ptr >= self.block_size:
+                self.in_ptr.value = 0
+                self.in_evt.set()  # 通知主线程来取甲缓冲
+            elif write_pos < self.buf_size and self.__rec_ptr < write_pos:
+                self.in_ptr.value = self.block_size
+                self.in_evt.set()  # 通知主线程来取乙缓冲
+
+        def combined_callback(indata, outdata, frames, time_info, status):
+            output_callback(outdata, frames, time_info, status)  # 优先出声
+            input_callback(indata, frames, time_info, status)
+
+        if self.is_device_combined:
+            with sd.Stream(
+                samplerate=self.sample_rate,
+                channels=self.channels,
+                dtype=self.buf_dtype,
+                latency='low',
+                extra_settings=exclusive_settings if
+                        self.is_input_wasapi_exclusive and
+                        self.is_output_wasapi_exclusive else None,
+                callback=combined_callback
+            ) as s:
+                self.latency.value = s.latency[-1]
+                self.stop_evt.wait()
+                self.out_buf.fill(0.0)
+        else:
+            with sd.InputStream(
+                samplerate=self.sample_rate,
+                channels=self.channels,
+                dtype=self.buf_dtype,
+                latency='low',
+                extra_settings=exclusive_settings if self.is_input_wasapi_exclusive else None,
+                callback=input_callback
+            ) as si, sd.OutputStream(
+                samplerate=self.sample_rate,
+                channels=self.channels,
+                dtype=self.buf_dtype,
+                latency='low',
+                extra_settings=exclusive_settings if self.is_output_wasapi_exclusive else None,
+                callback=output_callback
+            ) as so:
+                self.latency.value = si.latency[-1] + so.latency[-1]
+                self.stop_evt.wait()
+                self.out_buf.fill(0.0)
+
+        # 清理共享内存
+        in_mem.close()
+        out_mem.close()
+        in_mem.unlink()
+        out_mem.unlink()

requirements/gui-dml.txt

@@ -4,11 +4,6 @@
 #torch 1.11.0 with cuda 11.3
 #pip install torch==1.11.0+cu113 torchvision==0.12.0+cu113 torchaudio==0.11.0 --extra-index-url https://download.pytorch.org/whl/cu113
 einops
-fairseq
-flask
-flask_cors
-gin
-gin_config
 librosa
 local_attention
 matplotlib
@@ -20,7 +15,6 @@ scikit_learn
 scipy
 tensorboard
 tqdm
-wave
 FreeSimpleGUI
 sounddevice
 gradio

requirements/gui.txt

@@ -4,11 +4,6 @@
 #torch 1.11.0 with cuda 11.3
 #pip install torch==1.11.0+cu113 torchvision==0.12.0+cu113 torchaudio==0.11.0 --extra-index-url https://download.pytorch.org/whl/cu113
 einops
-fairseq
-flask
-flask_cors
-gin
-gin_config
 librosa
 local_attention
 matplotlib
@@ -20,7 +15,6 @@ scikit_learn
 scipy
 tensorboard
 tqdm
-wave
 FreeSimpleGUI
 sounddevice
 gradio