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Update app.py
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app.py
CHANGED
@@ -133,81 +133,174 @@ class InferenceModel(object):
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self.restore_from_checkpoint(checkpoint_path)
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@property
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title = "MT3"
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description = "MT3:多任务多音轨音乐转录的 Gradio 演示。要使用它,只需上传音频文件,或点击示例以查看效果。更多信息请参阅下面的链接。"
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@@ -224,4 +317,4 @@ gr.Interface(
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description=description,
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article=article,
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examples=examples
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).launch(server_port=7861)
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self.restore_from_checkpoint(checkpoint_path)
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@property
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def input_shapes(self):
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return {
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'encoder_input_tokens': (self.batch_size, self.inputs_length),
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'decoder_input_tokens': (self.batch_size, self.outputs_length)
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}
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def _parse_gin(self, gin_files):
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"""解析用于训练模型的 gin 文件。"""
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print(f"[{current_time()}] 日志:解析 gin 文件")
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gin_bindings = [
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'from __gin__ import dynamic_registration',
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'from mt3 import vocabularies',
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'VOCAB_CONFIG=@vocabularies.VocabularyConfig()',
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'vocabularies.VocabularyConfig.num_velocity_bins=%NUM_VELOCITY_BINS'
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]
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with gin.unlock_config():
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gin.parse_config_files_and_bindings(
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gin_files, gin_bindings, finalize_config=False)
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def _load_model(self):
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"""在解析训练 gin 配置后加载 T5X `Model`。"""
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print(f"[{current_time()}] 日志:加载 T5X 模型")
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model_config = gin.get_configurable(network.T5Config)()
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module = network.Transformer(config=model_config)
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return models.ContinuousInputsEncoderDecoderModel(
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module=module,
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input_vocabulary=self.output_features['inputs'].vocabulary,
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output_vocabulary=self.output_features['targets'].vocabulary,
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optimizer_def=t5x.adafactor.Adafactor(decay_rate=0.8, step_offset=0),
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input_depth=spectrograms.input_depth(self.spectrogram_config))
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def restore_from_checkpoint(self, checkpoint_path):
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"""从检查点中恢复训练状态,重置 self._predict_fn()。"""
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print(f"[{current_time()}] 日志:从检查点恢复训练状态")
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train_state_initializer = t5x.utils.TrainStateInitializer(
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optimizer_def=self.model.optimizer_def,
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init_fn=self.model.get_initial_variables,
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input_shapes=self.input_shapes,
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partitioner=self.partitioner)
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restore_checkpoint_cfg = t5x.utils.RestoreCheckpointConfig(
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path=checkpoint_path, mode='specific', dtype='float32')
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train_state_axes = train_state_initializer.train_state_axes
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self._predict_fn = self._get_predict_fn(train_state_axes)
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self._train_state = train_state_initializer.from_checkpoint_or_scratch(
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[restore_checkpoint_cfg], init_rng=jax.random.PRNGKey(0))
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@functools.lru_cache()
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def _get_predict_fn(self, train_state_axes):
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"""生成一个分区的预测函数用于解码。"""
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print(f"[{current_time()}] 日志:生成用于解码的预测函数")
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def partial_predict_fn(params, batch, decode_rng):
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return self.model.predict_batch_with_aux(
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params, batch, decoder_params={'decode_rng': None})
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return self.partitioner.partition(
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partial_predict_fn,
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in_axis_resources=(
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train_state_axes.params,
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t5x.partitioning.PartitionSpec('data',), None),
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out_axis_resources=t5x.partitioning.PartitionSpec('data',)
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)
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def predict_tokens(self, batch, seed=0):
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"""从预处理的数据集批次中预测 tokens。"""
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print(f"[{current_time()}] 运行:从预处理数据集中预测音符序列")
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prediction, _ = self._predict_fn(
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self._train_state.params, batch, jax.random.PRNGKey(seed))
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return self.vocabulary.decode_tf(prediction).numpy()
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def __call__(self, audio):
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"""从音频样本推断出音符序列。
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参数:
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audio:16kHz 的单个音频样本的 1 维 numpy 数组。
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返回:
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转录音频的音符序列。
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"""
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print(f"[{current_time()}] 运行:从音频样本中推断音符序列")
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ds = self.audio_to_dataset(audio)
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ds = self.preprocess(ds)
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model_ds = self.model.FEATURE_CONVERTER_CLS(pack=False)(
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ds, task_feature_lengths=self.sequence_length)
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model_ds = model_ds.batch(self.batch_size)
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inferences = (tokens for batch in model_ds.as_numpy_iterator()
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for tokens in self.predict_tokens(batch))
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predictions = []
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for example, tokens in zip(ds.as_numpy_iterator(), inferences):
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predictions.append(self.postprocess(tokens, example))
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result = metrics_utils.event_predictions_to_ns(
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predictions, codec=self.codec, encoding_spec=self.encoding_spec)
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return result['est_ns']
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def audio_to_dataset(self, audio):
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"""从输入音频创建一个包含频谱图的 TF Dataset。"""
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print(f"[{current_time()}] 运行:从音频创建包含频谱图的 TF Dataset")
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frames, frame_times = self._audio_to_frames(audio)
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return tf.data.Dataset.from_tensors({
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'inputs': frames,
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'input_times': frame_times,
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})
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def _audio_to_frames(self, audio):
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"""从音频计算频谱图帧。"""
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print(f"[{current_time()}] 运行:从音频计算频谱图帧")
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frame_size = self.spectrogram_config.hop_width
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padding = [0, frame_size - len(audio) % frame_size]
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audio = np.pad(audio, padding, mode='constant')
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frames = spectrograms.split_audio(audio, self.spectrogram_config)
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num_frames = len(audio) // frame_size
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times = np.arange(num_frames) / self.spectrogram_config.frames_per_second
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return frames, times
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def preprocess(self, ds):
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pp_chain = [
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functools.partial(
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t5.data.preprocessors.split_tokens_to_inputs_length,
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sequence_length=self.sequence_length,
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output_features=self.output_features,
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feature_key='inputs',
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additional_feature_keys=['input_times']),
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# 在训练期间进行缓存。
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preprocessors.add_dummy_targets,
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functools.partial(
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preprocessors.compute_spectrograms,
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spectrogram_config=self.spectrogram_config)
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]
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for pp in pp_chain:
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ds = pp(ds)
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return ds
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def postprocess(self, tokens, example):
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tokens = self._trim_eos(tokens)
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start_time = example['input_times'][0]
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# 向下取整到最接近的符号化时间步。
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start_time -= start_time % (1 / self.codec.steps_per_second)
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return {
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'est_tokens': tokens,
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'start_time': start_time,
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# 内部 MT3 代码期望原始输入,这里不使用。
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'raw_inputs': []
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}
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@staticmethod
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def _trim_eos(tokens):
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tokens = np.array(tokens, np.int32)
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if vocabularies.DECODED_EOS_ID in tokens:
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tokens = tokens[:np.argmax(tokens == vocabularies.DECODED_EOS_ID)]
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return tokens
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inference_model = InferenceModel('/home/user/app/checkpoints/mt3/', 'mt3')
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def inference(audio):
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filename = os.path.basename(audio) # 获取输入文件的文件名
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print(f"[{current_time()}] 运行:输入文件: {filename}")
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with open(audio, 'rb') as fd:
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contents = fd.read()
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audio = upload_audio(contents,sample_rate=16000)
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est_ns = inference_model(audio)
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note_seq.sequence_proto_to_midi_file(est_ns, './transcribed.mid')
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return './transcribed.mid'
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title = "MT3"
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description = "MT3:多任务多音轨音乐转录的 Gradio 演示。要使用它,只需上传音频文件,或点击示例以查看效果。更多信息请参阅下面的链接。"
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description=description,
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article=article,
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examples=examples
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).launch(server_port=7861)
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