mirror of
https://github.com/fumiama/Retrieval-based-Voice-Conversion-WebUI.git
synced 2026-06-05 01:10:22 +08:00
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
This commit is contained in:
haofanurusai
源文雨
124
gui.py
124
gui.py
@@ -76,8 +76,11 @@ if __name__ == "__main__":
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import json
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import multiprocessing
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import re
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import threading
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import time
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from multiprocessing import Queue, cpu_count
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from infer.lib.audio import AudioIoProcess
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from multiprocessing.shared_memory import SharedMemory
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import librosa
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from infer.modules.gui import TorchGate
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@@ -144,6 +147,15 @@ if __name__ == "__main__":
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self.input_devices_indices = None
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self.output_devices_indices = None
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self.stream = None
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self.in_mem = None
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self.out_mem = None
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self.in_buf = None
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self.out_buf = None
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self.in_ptr = None
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self.out_ptr = None
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self.play_ptr = None
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self.in_evt = None
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self.stop_evt = None
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self.update_devices()
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self.launcher()
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@@ -564,7 +576,7 @@ if __name__ == "__main__":
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event, values = self.window.read()
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if event == sg.WINDOW_CLOSED:
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self.stop_stream()
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exit()
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# exit()
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if event == "reload_devices" or event == "sg_hostapi":
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self.gui_config.sg_hostapi = values["sg_hostapi"]
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self.update_devices(hostapi_name=values["sg_hostapi"])
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@@ -639,7 +651,7 @@ if __name__ == "__main__":
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json.dump(settings, j)
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if self.stream is not None:
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self.delay_time = (
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self.stream.latency[-1]
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self.stream.get_latency()
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+ values["block_time"]
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+ values["crossfade_length"]
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+ 0.01
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@@ -667,7 +679,7 @@ if __name__ == "__main__":
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self.rvc.set_index_rate(values["index_rate"])
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elif event == "rms_mix_rate":
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self.gui_config.rms_mix_rate = values["rms_mix_rate"]
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elif event in ["pm", "dio", "harvest", "crepe", "rmvpe", "fcpe"]:
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elif event in ["pm", "harvest", "crepe", "rmvpe", "fcpe"]:
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self.gui_config.f0method = event
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elif event == "I_noise_reduce":
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self.gui_config.I_noise_reduce = values["I_noise_reduce"]
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@@ -867,36 +879,80 @@ if __name__ == "__main__":
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"WASAPI" in self.gui_config.sg_hostapi
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and self.gui_config.sg_wasapi_exclusive
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):
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extra_settings = sd.WasapiSettings(exclusive=True)
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wasapi_exclusive = True
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else:
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extra_settings = None
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self.stream = sd.Stream(
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callback=self.audio_callback,
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blocksize=self.block_frame,
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samplerate=self.gui_config.samplerate,
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channels=self.gui_config.channels,
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dtype="float32",
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extra_settings=extra_settings,
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wasapi_exclusive = False
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self.stream = AudioIoProcess(
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input_device=sd.default.device[0],
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output_device=sd.default.device[1],
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input_audio_block_size = self.block_frame,
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sample_rate = self.gui_config.samplerate,
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channel_num=self.gui_config.channels,
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is_input_wasapi_exclusive=wasapi_exclusive,
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is_output_wasapi_exclusive=wasapi_exclusive,
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is_device_combined = True
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# TODO: Add control UI to allow devices with different type API & different WASAPI settings
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)
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self.in_mem = SharedMemory(name=self.stream.get_in_mem_name())
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self.out_mem = SharedMemory(name=self.stream.get_out_mem_name())
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self.in_buf = np.ndarray(
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self.stream.get_np_shape(),
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dtype=self.stream.get_np_dtype(),
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buffer=self.in_mem.buf,
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order='C'
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)
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self.out_buf = np.ndarray(
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self.stream.get_np_shape(),
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dtype=self.stream.get_np_dtype(),
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buffer=self.out_mem.buf,
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order='C'
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)
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self.in_ptr, \
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self.out_ptr, \
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self.play_ptr, \
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self.in_evt, \
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self.stop_evt = self.stream.get_ptrs_and_events()
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self.stream.start()
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def audio_loop():
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while flag_vc:
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self.audio_infer(self.block_frame << 1)
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threading.Thread(
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target=audio_loop,
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daemon=True
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).start()
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def stop_stream(self):
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global flag_vc
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if flag_vc:
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flag_vc = False
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if self.stream is not None:
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self.stream.abort()
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self.stream.close()
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print("Exiting")
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self.stop_evt.set()
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self.in_mem.close()
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self.out_mem.close()
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self.stream.join()
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self.stream = None
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def audio_callback(
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self, indata: np.ndarray, outdata: np.ndarray, frames, times, status
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def audio_infer(
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self, buf_size:int # 2 * self.block_frame
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):
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"""
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音频处理
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"""
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global flag_vc
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self.in_evt.wait()
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rptr = self.in_ptr.value
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self.in_evt.clear()
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start_time = time.perf_counter()
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rend = rptr + self.block_frame
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indata = np.copy(self.in_buf[rptr:rend])
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indata = librosa.to_mono(indata.T)
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if self.gui_config.threhold > -60:
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indata = np.append(self.rms_buffer, indata)
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@@ -1039,13 +1095,47 @@ if __name__ == "__main__":
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self.sola_buffer[:] = infer_wav[
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self.block_frame : self.block_frame + self.sola_buffer_frame
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]
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outdata[:] = (
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outdata = (
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infer_wav[: self.block_frame]
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.repeat(self.gui_config.channels, 1)
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.t()
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.cpu()
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.numpy()
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)
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# 装填输出缓冲
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start = self.out_ptr.value
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play_pos = self.play_ptr.value
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# 计算播放进度差（写指针距离播放指针的帧数）
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delta = (start - play_pos + buf_size) % buf_size
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if delta < self.block_frame:
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# 装填赶不上播放，导致播放进度追上来了，
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# 此时已产生无法挽回的破音，
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# 只好直接卡着播放指针写入，保证接下来的尽快放出来
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print("[W] Output underrun")
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write_pos = play_pos
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else:
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# 否则按块对齐
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write_pos = (start + self.block_frame) % buf_size
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# 写入共享缓冲区
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end = (write_pos + self.block_frame) % buf_size
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if end > write_pos:
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self.out_buf[write_pos:end] = outdata
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else:
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first = buf_size - write_pos
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self.out_buf[write_pos:] = outdata[:first]
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self.out_buf[:end] = outdata[first:]
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# 更新写指针
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self.out_ptr.value = write_pos
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if self.in_evt.is_set():
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print("[W] Input overrun")
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self.in_evt.clear()
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total_time = time.perf_counter() - start_time
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if flag_vc:
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self.window["infer_time"].update(int(total_time * 1000))
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