improve: tune server device mode

This commit is contained in:
w-okada 2023-08-03 03:41:23 +09:00
parent 4abbe0839e
commit 839755e82f

View File

@ -20,6 +20,13 @@ AudioDeviceKind: TypeAlias = Literal["input", "output"]
logger = VoiceChangaerLogger.get_instance().getLogger()
# See https://github.com/w-okada/voice-changer/issues/620
LocalServerDeviceMode: TypeAlias = Literal[
"NoMonitorSeparate",
"WithMonitorStandard",
"WithMonitorAllSeparate",
]
@dataclass
class ServerDeviceSettings:
@ -95,6 +102,14 @@ class ServerDevice:
self.monQueue = Queue()
self.performance = []
# setting change確認用
self.currentServerInputDeviceId = -1
self.currentServerOutputDeviceId = -1
self.currentServerMonitorDeviceId = -1
self.currentModelSamplingRate = -1
self.currentInputChunkNum = -1
self.currentAudioSampleRate = -1
def getServerInputAudioDevice(self, index: int):
audioinput, _audiooutput = list_audio_device()
serverAudioDevice = [x for x in audioinput if x.index == index]
@ -111,36 +126,51 @@ class ServerDevice:
else:
return None
def audio_callback(self, indata: np.ndarray, outdata: np.ndarray, frames, times, status):
###########################################
# Callback Section
###########################################
def _processData(self, indata: np.ndarray):
indata = indata * self.settings.serverInputAudioGain
unpackedData = librosa.to_mono(indata.T) * 32768.0
unpackedData = unpackedData.astype(np.int16)
out_wav, times = self.serverDeviceCallbacks.on_request(unpackedData)
return out_wav, times
def _processDataWithTime(self, indata: np.ndarray):
with Timer("all_inference_time") as t:
out_wav, times = self._processData(indata)
all_inference_time = t.secs
self.performance = [all_inference_time] + times
self.serverDeviceCallbacks.emitTo(self.performance)
self.performance = [round(x * 1000) for x in self.performance]
return out_wav
def audio_callback_outQueue(self, indata: np.ndarray, outdata: np.ndarray, frames, times, status):
try:
indata = indata * self.settings.serverInputAudioGain
with Timer("all_inference_time") as t:
unpackedData = librosa.to_mono(indata.T) * 32768.0
unpackedData = unpackedData.astype(np.int16)
out_wav, times = self.serverDeviceCallbacks.on_request(unpackedData)
outputChannels = outdata.shape[1]
outdata[:] = np.repeat(out_wav, outputChannels).reshape(-1, outputChannels) / 32768.0
outdata[:] = outdata * self.settings.serverOutputAudioGain
all_inference_time = t.secs
self.performance = [all_inference_time] + times
self.serverDeviceCallbacks.emitTo(self.performance)
self.performance = [round(x * 1000) for x in self.performance]
out_wav = self._processDataWithTime(indata)
self.outQueue.put(out_wav)
outputChannels = outdata.shape[1] # Monitorへのアウトプット
outdata[:] = np.repeat(out_wav, outputChannels).reshape(-1, outputChannels) / 32768.0
outdata[:] = outdata * self.settings.serverOutputAudioGain
except Exception as e:
print("[Voice Changer] ex:", e)
def audioInput_callback(self, indata: np.ndarray, frames, times, status):
def audioInput_callback_outQueue(self, indata: np.ndarray, frames, times, status):
try:
indata = indata * self.settings.serverInputAudioGain
with Timer("all_inference_time") as t:
unpackedData = librosa.to_mono(indata.T) * 32768.0
unpackedData = unpackedData.astype(np.int16)
out_wav, times = self.serverDeviceCallbacks.on_request(unpackedData)
self.outQueue.put(out_wav)
self.monQueue.put(out_wav)
all_inference_time = t.secs
self.performance = [all_inference_time] + times
self.serverDeviceCallbacks.emitTo(self.performance)
self.performance = [round(x * 1000) for x in self.performance]
out_wav = self._processDataWithTime(indata)
self.outQueue.put(out_wav)
except Exception as e:
print("[Voice Changer][ServerDevice][audioInput_callback] ex:", e)
# import traceback
# traceback.print_exc()
def audioInput_callback_outQueue_monQueue(self, indata: np.ndarray, frames, times, status):
try:
out_wav = self._processDataWithTime(indata)
self.outQueue.put(out_wav)
self.monQueue.put(out_wav)
except Exception as e:
print("[Voice Changer][ServerDevice][audioInput_callback] ex:", e)
# import traceback
@ -173,8 +203,138 @@ class ServerDevice:
# import traceback
# traceback.print_exc()
###########################################
# Main Loop Section
###########################################
def checkSettingChanged(self):
if self.settings.serverAudioStated != 1:
print(f"serverAudioStarted Changed: {self.settings.serverAudioStated}")
return True
elif self.currentServerInputDeviceId != self.settings.serverInputDeviceId:
print(f"serverInputDeviceId Changed: {self.currentServerInputDeviceId} -> {self.settings.serverInputDeviceId}")
return True
elif self.currentServerOutputDeviceId != self.settings.serverOutputDeviceId:
print(f"serverOutputDeviceId Changed: {self.currentServerOutputDeviceId} -> {self.settings.serverOutputDeviceId}")
return True
elif self.currentServerMonitorDeviceId != self.settings.serverMonitorDeviceId:
print(f"serverMonitorDeviceId Changed: {self.currentServerMonitorDeviceId} -> {self.settings.serverMonitorDeviceId}")
return True
elif self.currentModelSamplingRate != self.serverDeviceCallbacks.get_processing_sampling_rate():
print(f"currentModelSamplingRate Changed: {self.currentModelSamplingRate} -> {self.serverDeviceCallbacks.get_processing_sampling_rate()}")
return True
elif self.currentInputChunkNum != self.settings.serverReadChunkSize:
print(f"currentInputChunkNum Changed: {self.currentInputChunkNum} -> {self.settings.serverReadChunkSize}")
return True
elif self.currentAudioSampleRate != self.settings.serverAudioSampleRate:
print(f"currentAudioSampleRate Changed: {self.currentAudioSampleRate} -> {self.settings.serverAudioSampleRate}")
return True
else:
return False
def runNoMonitorSeparate(self, block_frame: int, inputMaxChannel: int, outputMaxChannel: int, inputExtraSetting, outputExtraSetting):
with sd.InputStream(
callback=self.audioInput_callback_outQueue,
dtype="float32",
device=self.settings.serverInputDeviceId,
blocksize=block_frame,
samplerate=self.settings.serverInputAudioSampleRate,
channels=inputMaxChannel,
extra_settings=inputExtraSetting
):
with sd.OutputStream(
callback=self.audioOutput_callback,
dtype="float32",
device=self.settings.serverOutputDeviceId,
blocksize=block_frame,
samplerate=self.settings.serverOutputAudioSampleRate,
channels=outputMaxChannel,
extra_settings=outputExtraSetting
):
while True:
changed = self.checkSettingChanged()
if changed:
break
time.sleep(2)
print(f"[Voice Changer] server audio performance {self.performance}")
print(f" status: started:{self.settings.serverAudioStated}, model_sr:{self.currentModelSamplingRate}, chunk:{self.currentInputChunkNum}")
print(f" input : id:{self.settings.serverInputDeviceId}, sr:{self.settings.serverInputAudioSampleRate}, ch:{inputMaxChannel}")
print(f" output : id:{self.settings.serverOutputDeviceId}, sr:{self.settings.serverOutputAudioSampleRate}, ch:{outputMaxChannel}")
# print(f" monitor: id:{self.settings.serverMonitorDeviceId}, sr:{self.settings.serverMonitorAudioSampleRate}, ch:{self.serverMonitorAudioDevice.maxOutputChannels}")
def runWithMonitorStandard(self, block_frame: int, inputMaxChannel: int, outputMaxChannel: int, monitorMaxChannel: int, inputExtraSetting, outputExtraSetting, monitorExtraSetting):
with sd.Stream(
callback=self.audio_callback_outQueue,
dtype="float32",
device=(self.settings.serverInputDeviceId, self.settings.serverMonitorDeviceId),
blocksize=block_frame,
samplerate=self.settings.serverInputAudioSampleRate,
channels=(inputMaxChannel, monitorMaxChannel),
extra_settings=[inputExtraSetting, monitorExtraSetting]
):
with sd.OutputStream(
callback=self.audioOutput_callback,
dtype="float32",
device=self.settings.serverOutputDeviceId,
blocksize=block_frame,
samplerate=self.settings.serverOutputAudioSampleRate,
channels=outputMaxChannel,
extra_settings=outputExtraSetting
):
while True:
changed = self.checkSettingChanged()
if changed:
break
time.sleep(2)
print(f"[Voice Changer] server audio performance {self.performance}")
print(f" status: started:{self.settings.serverAudioStated}, model_sr:{self.currentModelSamplingRate}, chunk:{self.currentInputChunkNum}")
print(f" input : id:{self.settings.serverInputDeviceId}, sr:{self.settings.serverInputAudioSampleRate}, ch:{inputMaxChannel}")
print(f" output : id:{self.settings.serverOutputDeviceId}, sr:{self.settings.serverOutputAudioSampleRate}, ch:{outputMaxChannel}")
print(f" monitor: id:{self.settings.serverMonitorDeviceId}, sr:{self.settings.serverMonitorAudioSampleRate}, ch:{monitorMaxChannel}")
def runWithMonitorAllSeparate(self, block_frame: int, inputMaxChannel: int, outputMaxChannel: int, monitorMaxChannel: int, inputExtraSetting, outputExtraSetting, monitorExtraSetting):
with sd.InputStream(
callback=self.audioInput_callback_outQueue_monQueue,
dtype="float32",
device=self.settings.serverInputDeviceId,
blocksize=block_frame,
samplerate=self.settings.serverInputAudioSampleRate,
channels=inputMaxChannel,
extra_settings=inputExtraSetting
):
with sd.OutputStream(
callback=self.audioOutput_callback,
dtype="float32",
device=self.settings.serverOutputDeviceId,
blocksize=block_frame,
samplerate=self.settings.serverOutputAudioSampleRate,
channels=outputMaxChannel,
extra_settings=outputExtraSetting
):
with sd.OutputStream(
callback=self.audioMonitor_callback,
dtype="float32",
device=self.settings.serverMonitorDeviceId,
blocksize=block_frame,
samplerate=self.settings.serverMonitorAudioSampleRate,
channels=monitorMaxChannel,
extra_settings=monitorExtraSetting
):
while True:
changed = self.checkSettingChanged()
if changed:
break
time.sleep(2)
print(f"[Voice Changer] server audio performance {self.performance}")
print(f" status: started:{self.settings.serverAudioStated}, model_sr:{self.currentModelSamplingRate}, chunk:{self.currentInputChunkNum}")
print(f" input : id:{self.settings.serverInputDeviceId}, sr:{self.settings.serverInputAudioSampleRate}, ch:{inputMaxChannel}")
print(f" output : id:{self.settings.serverOutputDeviceId}, sr:{self.settings.serverOutputAudioSampleRate}, ch:{outputMaxChannel}")
print(f" monitor: id:{self.settings.serverMonitorDeviceId}, sr:{self.settings.serverMonitorAudioSampleRate}, ch:{monitorMaxChannel}")
###########################################
# Start Section
###########################################
def start(self):
currentModelSamplingRate = -1
self.currentModelSamplingRate = -1
while True:
if self.settings.serverAudioStated == 0 or self.settings.serverInputDeviceId == -1:
time.sleep(2)
@ -183,9 +343,9 @@ class ServerDevice:
sd._initialize()
# Curret Device ID
currentServerInputDeviceId = self.settings.serverInputDeviceId
currentServerOutputDeviceId = self.settings.serverOutputDeviceId
currentServerMonitorDeviceId = self.settings.serverMonitorDeviceId
self.currentServerInputDeviceId = self.settings.serverInputDeviceId
self.currentServerOutputDeviceId = self.settings.serverOutputDeviceId
self.currentServerMonitorDeviceId = self.settings.serverMonitorDeviceId
# Device 特定
serverInputAudioDevice = self.getServerInputAudioDevice(self.settings.serverInputDeviceId)
@ -220,17 +380,17 @@ class ServerDevice:
# サンプリングレート
# 同一サンプリングレートに統一(変換時にサンプルが不足する場合があるため。パディング方法が明らかになれば、それぞれ設定できるかも)
currentAudioSampleRate = self.settings.serverAudioSampleRate
self.currentAudioSampleRate = self.settings.serverAudioSampleRate
try:
currentModelSamplingRate = self.serverDeviceCallbacks.get_processing_sampling_rate()
self.currentModelSamplingRate = self.serverDeviceCallbacks.get_processing_sampling_rate()
except Exception as e:
print("[Voice Changer] ex: get_processing_sampling_rate", e)
time.sleep(2)
continue
self.settings.serverInputAudioSampleRate = currentAudioSampleRate
self.settings.serverOutputAudioSampleRate = currentAudioSampleRate
self.settings.serverMonitorAudioSampleRate = currentAudioSampleRate
self.settings.serverInputAudioSampleRate = self.currentAudioSampleRate
self.settings.serverOutputAudioSampleRate = self.currentAudioSampleRate
self.settings.serverMonitorAudioSampleRate = self.currentAudioSampleRate
# Sample Rate Check
inputAudioSampleRateAvailable = checkSamplingRate(self.settings.serverInputDeviceId, self.settings.serverInputAudioSampleRate, "input")
@ -238,7 +398,7 @@ class ServerDevice:
monitorAudioSampleRateAvailable = checkSamplingRate(self.settings.serverMonitorDeviceId, self.settings.serverMonitorAudioSampleRate, "output") if serverMonitorAudioDevice else True
print("Sample Rate:")
print(f" [Model]: {currentModelSamplingRate}")
print(f" [Model]: {self.currentModelSamplingRate}")
print(f" [Input]: {self.settings.serverInputAudioSampleRate} -> {inputAudioSampleRateAvailable}")
print(f" [Output]: {self.settings.serverOutputAudioSampleRate} -> {outputAudioSampleRateAvailable}")
if serverMonitorAudioDevice is not None:
@ -274,153 +434,51 @@ class ServerDevice:
self.serverDeviceCallbacks.setOutputSamplingRate(self.settings.serverOutputAudioSampleRate)
# Blockサイズを計算
currentInputChunkNum = self.settings.serverReadChunkSize
self.currentInputChunkNum = self.settings.serverReadChunkSize
# block_frame = currentInputChunkNum * 128
block_frame = int(currentInputChunkNum * 128 * (self.settings.serverInputAudioSampleRate / 48000))
block_frame = int(self.currentInputChunkNum * 128 * (self.settings.serverInputAudioSampleRate / 48000))
sd.default.blocksize = block_frame
# main loop
try:
with sd.InputStream(
callback=self.audioInput_callback,
dtype="float32",
device=self.settings.serverInputDeviceId,
blocksize=block_frame,
samplerate=self.settings.serverInputAudioSampleRate,
channels=serverInputAudioDevice.maxInputChannels,
extra_settings=inputExtraSetting
):
with sd.OutputStream(
callback=self.audioOutput_callback,
dtype="float32",
device=self.settings.serverOutputDeviceId,
blocksize=block_frame,
samplerate=self.settings.serverOutputAudioSampleRate,
channels=serverOutputAudioDevice.maxOutputChannels,
extra_settings=outputExtraSetting
):
if self.settings.serverMonitorDeviceId != -1:
with sd.OutputStream(
callback=self.audioMonitor_callback,
dtype="float32",
device=self.settings.serverMonitorDeviceId,
blocksize=block_frame,
samplerate=self.settings.serverMonitorAudioSampleRate,
channels=serverMonitorAudioDevice.maxOutputChannels,
extra_settings=monitorExtraSetting
):
while (
self.settings.serverAudioStated == 1 and
currentServerInputDeviceId == self.settings.serverInputDeviceId and
currentServerOutputDeviceId == self.settings.serverOutputDeviceId and
currentServerMonitorDeviceId == self.settings.serverMonitorDeviceId and
currentModelSamplingRate == self.serverDeviceCallbacks.get_processing_sampling_rate() and
currentInputChunkNum == self.settings.serverReadChunkSize and
currentAudioSampleRate == self.settings.serverAudioSampleRate
):
time.sleep(2)
print(f"[Voice Changer] server audio performance {self.performance}")
print(f" status: started:{self.settings.serverAudioStated}, model_sr:{currentModelSamplingRate}, chunk:{currentInputChunkNum}")
print(f" input : id:{self.settings.serverInputDeviceId}, sr:{self.settings.serverInputAudioSampleRate}, ch:{serverInputAudioDevice.maxInputChannels}")
print(f" output : id:{self.settings.serverOutputDeviceId}, sr:{self.settings.serverOutputAudioSampleRate}, ch:{serverOutputAudioDevice.maxOutputChannels}")
print(f" monitor: id:{self.settings.serverMonitorDeviceId}, sr:{self.settings.serverMonitorAudioSampleRate}, ch:{serverMonitorAudioDevice.maxOutputChannels}")
# See https://github.com/w-okada/voice-changer/issues/620
def judgeServerDeviceMode() -> LocalServerDeviceMode:
if self.settings.serverMonitorDeviceId == -1:
return "NoMonitorSeparate"
else:
if serverInputAudioDevice.hostAPI == serverOutputAudioDevice.hostAPI and serverInputAudioDevice.hostAPI == serverMonitorAudioDevice.hostAPI: # すべて同じ
return "WithMonitorStandard"
elif serverInputAudioDevice.hostAPI != serverOutputAudioDevice.hostAPI and serverInputAudioDevice.hostAPI != serverMonitorAudioDevice.hostAPI and serverOutputAudioDevice.hostAPI != serverMonitorAudioDevice.hostAPI: # すべて違う
return "WithMonitorAllSeparate"
elif serverInputAudioDevice.hostAPI == serverOutputAudioDevice.hostAPI: # in/outだけが同じ
return "WithMonitorAllSeparate"
elif serverInputAudioDevice.hostAPI == serverMonitorAudioDevice.hostAPI: # in/monだけが同じ
return "WithMonitorStandard"
elif serverOutputAudioDevice.hostAPI == serverMonitorAudioDevice.hostAPI: # out/monだけが同じ
return "WithMonitorAllSeparate"
else:
while (
self.settings.serverAudioStated == 1 and
currentServerInputDeviceId == self.settings.serverInputDeviceId and
currentServerOutputDeviceId == self.settings.serverOutputDeviceId and
currentServerMonitorDeviceId == self.settings.serverMonitorDeviceId and
currentModelSamplingRate == self.serverDeviceCallbacks.get_processing_sampling_rate() and
currentInputChunkNum == self.settings.serverReadChunkSize and
currentAudioSampleRate == self.settings.serverAudioSampleRate
):
time.sleep(2)
print(f"[Voice Changer] server audio performance {self.performance}")
print(f" status: started:{self.settings.serverAudioStated}, model_sr:{currentModelSamplingRate}, chunk:{currentInputChunkNum}]")
print(f" input : id:{self.settings.serverInputDeviceId}, sr:{self.settings.serverInputAudioSampleRate}, ch:{serverInputAudioDevice.maxInputChannels}")
print(f" output : id:{self.settings.serverOutputDeviceId}, sr:{self.settings.serverOutputAudioSampleRate}, ch:{serverOutputAudioDevice.maxOutputChannels}")
raise RuntimeError(f"Cannot JudgeServerMode, in:{serverInputAudioDevice.hostAPI}, mon:{serverMonitorAudioDevice.hostAPI}, out:{serverOutputAudioDevice.hostAPI}")
serverDeviceMode = judgeServerDeviceMode()
if serverDeviceMode == "NoMonitorSeparate":
self.runNoMonitorSeparate(block_frame, serverInputAudioDevice.maxInputChannels, serverOutputAudioDevice.maxOutputChannels, inputExtraSetting, outputExtraSetting)
elif serverDeviceMode == "WithMonitorStandard":
self.runWithMonitorStandard(block_frame, serverInputAudioDevice.maxInputChannels, serverOutputAudioDevice.maxOutputChannels, serverMonitorAudioDevice.maxOutputChannels, inputExtraSetting, outputExtraSetting, monitorExtraSetting)
elif serverDeviceMode == "WithMonitorAllSeparate":
self.runWithMonitorAllSeparate(block_frame, serverInputAudioDevice.maxInputChannels, serverOutputAudioDevice.maxOutputChannels, serverMonitorAudioDevice.maxOutputChannels, inputExtraSetting, outputExtraSetting, monitorExtraSetting)
else:
raise RuntimeError(f"Unknown ServerDeviceMode: {serverDeviceMode}")
except Exception as e:
print("[Voice Changer] processing, ex:", e)
import traceback
traceback.print_exc()
time.sleep(2)
def start2(self):
# currentInputDeviceId = -1
# currentOutputDeviceId = -1
# currentInputChunkNum = -1
currentModelSamplingRate = -1
while True:
if self.settings.serverAudioStated == 0 or self.settings.serverInputDeviceId == -1:
time.sleep(2)
else:
sd._terminate()
sd._initialize()
sd.default.device[0] = self.settings.serverInputDeviceId
sd.default.device[1] = self.settings.serverOutputDeviceId
serverInputAudioDevice = self.getServerInputAudioDevice(sd.default.device[0])
serverOutputAudioDevice = self.getServerOutputAudioDevice(sd.default.device[1])
print("Devices:", serverInputAudioDevice, serverOutputAudioDevice)
if serverInputAudioDevice is None or serverOutputAudioDevice is None:
time.sleep(2)
print("serverInputAudioDevice or serverOutputAudioDevice is None")
continue
sd.default.channels[0] = serverInputAudioDevice.maxInputChannels
sd.default.channels[1] = serverOutputAudioDevice.maxOutputChannels
currentInputChunkNum = self.settings.serverReadChunkSize
block_frame = currentInputChunkNum * 128
# sample rate precheck(alsa cannot use 40000?)
try:
currentModelSamplingRate = self.serverDeviceCallbacks.get_processing_sampling_rate()
except Exception as e:
print("[Voice Changer] ex: get_processing_sampling_rate", e)
continue
try:
with sd.Stream(
callback=self.audio_callback,
blocksize=block_frame,
# samplerate=currentModelSamplingRate,
dtype="float32",
# dtype="int16",
# channels=[currentInputChannelNum, currentOutputChannelNum],
):
pass
self.settings.serverInputAudioSampleRate = currentModelSamplingRate
self.serverDeviceCallbacks.setInputSamplingRate(currentModelSamplingRate)
self.serverDeviceCallbacks.setOutputSamplingRate(currentModelSamplingRate)
print(f"[Voice Changer] sample rate {self.settings.serverInputAudioSampleRate}")
except Exception as e:
print("[Voice Changer] ex: fallback to device default samplerate", e)
print("[Voice Changer] device default samplerate", serverInputAudioDevice.default_samplerate)
self.settings.serverInputAudioSampleRate = round(serverInputAudioDevice.default_samplerate)
self.serverDeviceCallbacks.setInputSamplingRate(round(serverInputAudioDevice.default_samplerate))
self.serverDeviceCallbacks.setOutputSamplingRate(round(serverInputAudioDevice.default_samplerate))
sd.default.samplerate = self.settings.serverInputAudioSampleRate
sd.default.blocksize = block_frame
# main loop
try:
with sd.Stream(
callback=self.audio_callback,
# blocksize=block_frame,
# samplerate=vc.settings.serverInputAudioSampleRate,
dtype="float32",
# dtype="int16",
# channels=[currentInputChannelNum, currentOutputChannelNum],
):
while self.settings.serverAudioStated == 1 and sd.default.device[0] == self.settings.serverInputDeviceId and sd.default.device[1] == self.settings.serverOutputDeviceId and currentModelSamplingRate == self.serverDeviceCallbacks.get_processing_sampling_rate() and currentInputChunkNum == self.settings.serverReadChunkSize:
time.sleep(2)
print("[Voice Changer] server audio", self.performance)
print(f"[Voice Changer] started:{self.settings.serverAudioStated}, input:{sd.default.device[0]}, output:{sd.default.device[1]}, mic_sr:{self.settings.serverInputAudioSampleRate}, model_sr:{currentModelSamplingRate}, chunk:{currentInputChunkNum}, ch:[{sd.default.channels}]")
except Exception as e:
print("[Voice Changer] ex:", e)
time.sleep(2)
###########################################
# Info Section
###########################################
def get_info(self):
data = asdict(self.settings)
try: