make audio encoding multi-gpus compatible

run_stable_speech_training.py

@@ -32,6 +32,8 @@ from typing import Dict, List, Optional, Union
 import datasets
 import numpy as np
 import torch
+from torch.utils.data import DataLoader
+
 from datasets import DatasetDict, load_dataset, Dataset, IterableDataset, interleave_datasets, concatenate_datasets
 
 import transformers
@@ -43,13 +45,15 @@ from transformers import (
     HfArgumentParser,
     Seq2SeqTrainer,
     Seq2SeqTrainingArguments,
-    set_seed,
 )
 from transformers.trainer_utils import get_last_checkpoint, is_main_process
 from transformers.utils import check_min_version, send_example_telemetry
 from transformers.utils.versions import require_version
 from transformers.integrations import is_wandb_available
-from accelerate import PartialState
+
+from accelerate import Accelerator
+from accelerate.utils import set_seed
+
 
 from stable_speech import StableSpeechForConditionalGeneration, StableSpeechConfig
 
@@ -214,10 +218,6 @@ class DataSeq2SeqTrainingArguments:
         default="audio",
         metadata={"help": "The name of the dataset column containing the target audio data. Defaults to 'audio'"},
     )
-    conditional_audio_column_name: str = field( # TODO
-        default=None,
-        metadata={"help": "The name of the dataset column containing the conditional audio data. Defaults to 'audio'"},
-    )
     description_column_name: str = field( #TODO
         default=None,
         metadata={"help": "The name of the dataset column containing the text data. Defaults to 'None'."},
@@ -312,6 +312,29 @@ class DataSeq2SeqTrainingArguments:
         }
     )
     
+@dataclass
+class DataCollatorEncodecWithPadding:
+    """
+    Data collator that will dynamically pad the inputs received to the longest sequence in the batch.
+    """
+
+    feature_extractor: AutoFeatureExtractor
+    feature_extractor_input_name: Optional[str] = "input_values"
+
+    def __call__(self, features: List[Dict[str, Union[List[int], torch.Tensor]]]) -> Dict[str, torch.Tensor]:
+        # split inputs and labels since they have to be of different lengths and need
+        # different padding methods
+        audios = [torch.tensor(feature["labels"]).squeeze() for feature in features]
+        len_audio = [len(audio) for audio in audios]
+        max_audio = max(len_audio)
+        
+        input_features = {self.feature_extractor_input_name: audios}
+        batch = self.feature_extractor.pad(input_features, return_tensors="pt", padding="longest", return_attention_mask=True)
+        batch[self.feature_extractor_input_name] = batch[self.feature_extractor_input_name].unsqueeze(1) # add mono-channel 
+        batch["padding_mask"] = batch.pop("attention_mask")   
+        batch["len_audio"] = torch.tensor(len_audio).unsqueeze(1)
+        return batch
+
     
 @dataclass
 class DataCollatorStableSpeechWithPadding:
@@ -437,6 +460,7 @@ def convert_dataset_str_to_list(
 
 
 def load_multiple_datasets(
+    accelerator: Accelerator,
     dataset_names: Union[List, str],
     dataset_config_names: Union[List, str],
     metadata_dataset_names: Optional[str]=None,
@@ -463,6 +487,7 @@ def load_multiple_datasets(
     all_datasets = []
     # iterate over the datasets we want to interleave
     for dataset_dict in tqdm(dataset_names_dict, desc="Combining datasets..."):
+        with accelerator.main_process_first():
             dataset = load_dataset(
                 dataset_dict["name"],
                 dataset_dict["config"],
@@ -522,6 +547,7 @@ def load_multiple_datasets(
             seed=seed,
         )
     else:
+        with accelerator.main_process_first():
             interleaved_dataset = concatenate_datasets(all_datasets)
 
     return interleaved_dataset
@@ -545,6 +571,8 @@ def main():
     # information sent is the one passed as arguments along with your Python/PyTorch versions.
     send_example_telemetry("run_stable_speech", model_args, data_args)
     
+    accelerator = Accelerator()
+
     # Detecting last checkpoint.
     last_checkpoint = None
     if os.path.isdir(training_args.output_dir) and training_args.do_train and not training_args.overwrite_output_dir:
@@ -566,7 +594,7 @@ def main():
         datefmt="%m/%d/%Y %H:%M:%S",
         handlers=[logging.StreamHandler(sys.stdout)],
     )
-    logger.setLevel(logging.INFO if is_main_process(training_args.local_rank) else logging.WARN)
+    logger.setLevel(logging.INFO if accelerator.is_main_process else logging.WARN)
 
     # Log on each process the small summary:
     logger.warning(
@@ -574,8 +602,9 @@ def main():
         f"distributed training: {training_args.parallel_mode.value == 'distributed'}, 16-bits training: {training_args.fp16}"
     )
     # Set the verbosity to info of the Transformers logger (on main process only):
-    if is_main_process(training_args.local_rank):
+    if accelerator.is_main_process:
         transformers.utils.logging.set_verbosity_info()
+
     logger.info("Training/evaluation parameters %s", training_args)
 
     # Set seed before initializing model.
@@ -585,71 +614,51 @@ def main():
     raw_datasets = DatasetDict()
     num_workers = data_args.preprocessing_num_workers
     
-    columns_to_keep = [data_args.target_audio_column_name, data_args.prompt_column_name]
+    columns_to_keep = {
+        "target_audio_column_name": data_args.target_audio_column_name,
+        "prompt_column_name": data_args.prompt_column_name
+    }
     if data_args.description_column_name is not None:
-        columns_to_keep.append(data_args.description_column_name)
-    if data_args.conditional_audio_column_name is not None:
-        columns_to_keep.append(data_args.conditional_audio_column_name)
+        columns_to_keep["description_column_nam"] = data_args.description_column_name
         
     if training_args.do_train:
         raw_datasets["train"] = load_multiple_datasets(
+            accelerator,
             data_args.train_dataset_name,
             data_args.train_dataset_config_name,
-            data_args.train_metadata_dataset_name,
+            metadata_dataset_names=data_args.train_metadata_dataset_name,
             splits=data_args.train_split_name,
             dataset_samples=data_args.train_dataset_samples,
             seed=training_args.seed,
             cache_dir=model_args.cache_dir,
             num_proc=data_args.preprocessing_num_workers,
             id_column_name=data_args.id_column_name,
-            columns_to_keep=columns_to_keep,
+            columns_to_keep=columns_to_keep.values(),
             # streaming=data_args.streaming, TODO(SG): optionally enable streaming mode
         )
         
-        if data_args.target_audio_column_name not in raw_datasets["train"].column_names:
+        for key in columns_to_keep:
+            if columns_to_keep[key] not in raw_datasets["train"].column_names:
                 raise ValueError(
-                f"--target_audio_column_name '{data_args.target_audio_column_name}' not found in dataset '{data_args.train_dataset_name}'."
-                " Make sure to set `--target_audio_column_name` to the correct audio column - one of"
+                    f"--{key} '{columns_to_keep[key]}' not found in dataset '{data_args.train_dataset_name}'."
+                    f" Make sure to set `--{key}` to the correct audio column - one of"
                     f" {', '.join(raw_datasets['train'].column_names)}."
                 )        
 
-        if data_args.description_column_name is not None and data_args.description_column_name not in raw_datasets["train"].column_names:
-            raise ValueError(
-                f"--description_column_name {data_args.description_column_name} not found in dataset '{data_args.train_dataset_name}'. "
-                "Make sure to set `--description_column_name` to the correct text column - one of "
-                f"{', '.join(raw_datasets['train'].column_names)}."
-            )
-            
-        if data_args.prompt_column_name not in raw_datasets["train"].column_names:
-            raise ValueError(
-                f"--description_column_name {data_args.prompt_column_name} not found in dataset '{data_args.train_dataset_name}'. "
-                "Make sure to set `--prompt_column_name` to the correct text column - one of "
-                f"{', '.join(raw_datasets['train'].column_names)}."
-            )
-            
-        if data_args.conditional_audio_column_name is not None and data_args.conditional_audio_column_name not in raw_datasets["train"].column_names:
-            raise ValueError(
-                f"--conditional_audio_column_name {data_args.conditional_audio_column_name} not found in dataset '{data_args.train_dataset_name}'. "
-                "Make sure to set `--conditional_audio_column_name` to the correct text column - one of "
-                f"{', '.join(raw_datasets['train'].column_names)}."
-            )
-        
-
         if data_args.max_train_samples is not None:
             raw_datasets["train"] = raw_datasets["train"].select(range(data_args.max_train_samples))
 
     if training_args.do_eval:
         raw_datasets["eval"] = load_multiple_datasets(
+            accelerator,
             data_args.eval_dataset_name if data_args.eval_dataset_name else data_args.train_dataset_name,
-            data_args.eval_dataset_config_name
-            if data_args.eval_dataset_config_name
-            else data_args.train_dataset_config_name,
-            data_args.eval_metadata_dataset_name,
+            data_args.eval_dataset_config_name if data_args.eval_dataset_config_name else data_args.train_dataset_config_name,
+            metadata_dataset_names=data_args.eval_metadata_dataset_name,
             splits=data_args.eval_split_name,
             cache_dir=model_args.cache_dir,
             num_proc=data_args.preprocessing_num_workers,
             id_column_name=data_args.id_column_name,
-            columns_to_keep=columns_to_keep,
+            columns_to_keep=columns_to_keep.values(),
             # streaming=data_args.streaming, TODO(SG): optionally enable streaming mode
         )
 
@@ -657,9 +666,8 @@ def main():
             raw_datasets["eval"] = raw_datasets["eval"].select(range(data_args.max_eval_samples))
 
 
-    # TODO: is is the right way to do ?
-    # 3. Next, let's load the config as we might need it to create
-    # load config
+    # 2. Next, let's load the config as we might need it to create
+    # load config TODO: add the option to create the config from scratch
     config = StableSpeechConfig.from_pretrained(
         model_args.model_name_or_path,
         cache_dir=model_args.cache_dir,
@@ -673,8 +681,7 @@ def main():
         "decoder_start_token_id": model_args.decoder_start_token_id if model_args.decoder_start_token_id is not None else model.config.decoder_start_token_id,
     })
 
-
-    # 4. Now we can instantiate the feature extractor, tokenizers and model
+    # 3. Now we can instantiate the feature extractor, tokenizers and model
     # Note for distributed training, the .from_pretrained methods guarantee that only
     # one local process can concurrently download model & vocab.
     
@@ -692,6 +699,7 @@ def main():
         cache_dir=model_args.cache_dir,
         token=data_args.token,
         trust_remote_code=data_args.trust_remote_code,
+        use_fast=model_args.use_fast_tokenizer,
     )
         
     # load description tokenizer
@@ -700,8 +708,15 @@ def main():
         cache_dir=model_args.cache_dir,
         token=data_args.token,
         trust_remote_code=data_args.trust_remote_code,
+        use_fast=model_args.use_fast_tokenizer,
     )
     
+    if model_args.use_fast_tokenizer:
+        logger.warning("Disabling fast tokenizer warning: https://github.com/huggingface/transformers/blob/main/src/transformers/tokenization_utils_base.py#L3231-L3235")
+        prompt_tokenizer.deprecation_warnings["Asking-to-pad-a-fast-tokenizer"] = True
+        description_tokenizer.deprecation_warnings["Asking-to-pad-a-fast-tokenizer"] = True
+
+    
     # create model + TODO: not from_pretrained probably
     model = StableSpeechForConditionalGeneration.from_pretrained(
         model_args.model_name_or_path,
@@ -711,46 +726,31 @@ def main():
         trust_remote_code=data_args.trust_remote_code,
     )
     
-    # take audio_encoder_feature_extractor
-    audio_encoder_feature_extractor = AutoFeatureExtractor.from_pretrained(
-        model.config.audio_encoder._name_or_path,                                                   
-        )
     
-    # 5. Now we preprocess the datasets including loading the audio, resampling and normalization
+    # 4. Now we preprocess the datasets including loading the audio, resampling and normalization
     # Thankfully, `datasets` takes care of automatically loading and resampling the audio,
     # so that we just need to set the correct target sampling rate and normalize the input
     # via the `feature_extractor`
     
-    
-    # resample target audio
-    raw_datasets = raw_datasets.cast_column(
-        data_args.target_audio_column_name, datasets.features.Audio(sampling_rate=audio_encoder_feature_extractor.sampling_rate)
-    )
-
-    if data_args.conditional_audio_column_name is not None:
-        raw_datasets = raw_datasets.cast_column(
-            data_args.conditional_audio_column_name, datasets.features.Audio(sampling_rate=feature_extractor.sampling_rate)
-        )
-
     # derive max & min input length for sample rate & max duration
-    max_target_length = data_args.max_duration_in_seconds * feature_extractor.sampling_rate
-    min_target_length = data_args.min_duration_in_seconds * feature_extractor.sampling_rate
+    sampling_rate = feature_extractor.sampling_rate
+    max_target_length = data_args.max_duration_in_seconds * sampling_rate
+    min_target_length = data_args.min_duration_in_seconds * sampling_rate
     target_audio_column_name = data_args.target_audio_column_name
-    conditional_audio_column_name = data_args.conditional_audio_column_name
     description_column_name = data_args.description_column_name
     prompt_column_name = data_args.prompt_column_name
     feature_extractor_input_name = feature_extractor.model_input_names[0]
     
+    # resample target audio
+    raw_datasets = raw_datasets.cast_column(
+        target_audio_column_name, datasets.features.Audio(sampling_rate=sampling_rate)
+    )
+    
 
     # Preprocessing the datasets.
-    # We need to read the audio files as arrays and tokenize the targets.
+    # We need to read the audio files as arrays and tokenize the texts.
     def pass_through_processors(batch):
         # load audio
-        if conditional_audio_column_name is not None:
-            sample = batch[target_audio_column_name]
-            inputs = feature_extractor(sample["array"], sampling_rate=sample["sampling_rate"])
-            batch[feature_extractor_input_name] = getattr(inputs, feature_extractor_input_name)[0]
-            
         if description_column_name is not None:
             text = batch[description_column_name]
             batch["input_ids"] = description_tokenizer(text.strip())["input_ids"]
@@ -761,14 +761,14 @@ def main():
 
         # load audio
         target_sample = batch[target_audio_column_name]
-        labels = audio_encoder_feature_extractor(target_sample["array"], sampling_rate=target_sample["sampling_rate"])
+        labels = feature_extractor(target_sample["array"], sampling_rate=target_sample["sampling_rate"])
         batch["labels"] = labels["input_values"]
         
         # take length of raw audio waveform
         batch["target_length"] = len(target_sample["array"].squeeze())
         return batch
 
-    with training_args.main_process_first(desc="dataset map preprocessing"):
+    with accelerator.main_process_first():
         vectorized_datasets = raw_datasets.map(
             pass_through_processors,
             remove_columns=next(iter(raw_datasets.values())).column_names,
@@ -786,33 +786,80 @@ def main():
             input_columns=["target_length"],
         )
 
+
+    # 5. Now we encode the audio labels with encodec.
+    # We use Accelerate to perform distributed inference
+
+    logger.info("*** Encode target audio with encodec ***")
+    
+    # no need to prepare audio_decoder because used for inference without mixed precision
+    # see: https://huggingface.co/docs/accelerate/main/en/package_reference/accelerator#accelerate.Accelerator.prepare
+    # TODO: load another model
     audio_decoder = model.audio_encoder
 
+    encoder_data_collator = DataCollatorEncodecWithPadding(feature_extractor, feature_extractor_input_name)
+
     def apply_audio_decoder(batch):
-        labels = audio_decoder.encode(torch.tensor(batch["labels"]).to(audio_decoder.device))["audio_codes"]
+        len_audio = batch.pop("len_audio")
+        audio_decoder.to(batch["input_values"].device).eval()
+        labels = audio_decoder.encode(**batch)["audio_codes"]
+        output = {}
+        output["len_audio"] = len_audio
+        # (1, bsz, codebooks, seq_len) -> (bsz, seq_len, codebooks)
+        output["labels"] = labels.squeeze(0).transpose(1,2)
+        output["ratio"] = torch.ones_like(len_audio) * labels.shape[-1] / len_audio.max() 
+        return output
+
+    for split in vectorized_datasets:
+        data_loader = DataLoader(
+            vectorized_datasets[split],
+            batch_size=training_args.per_device_eval_batch_size,
+            collate_fn=encoder_data_collator,
+            num_workers=training_args.dataloader_num_workers,
+            pin_memory=True,
+        )
+        data_loader = accelerator.prepare(data_loader)        
+        
+        all_generated_labels = []
+        all_ratios = []
+        all_lens = []
+        for batch in tqdm(data_loader, disable=not accelerator.is_local_main_process):
+            generate_labels = apply_audio_decoder(batch)
+            generate_labels = accelerator.pad_across_processes(generate_labels, dim=1, pad_index=0)
+            generate_labels = accelerator.gather_for_metrics(generate_labels)
+            
+            all_generated_labels.extend(generate_labels["labels"].cpu())
+            all_ratios.extend(generate_labels["ratio"].cpu())
+            all_lens.extend(generate_labels["len_audio"].cpu())
+            
+        def postprocess_dataset(sample, idx):
+            # (1, seq_len, codebooks, bsz)
+            labels = all_generated_labels[idx].transpose(0,1).unsqueeze(0)
             labels, delay_pattern_mask = model.decoder.build_delay_pattern_mask(labels, 
                                                     model.generation_config.decoder_start_token_id, 
-                                                model.generation_config.max_length + 1)
+                                                    model.generation_config.max_length + model.decoder.config.num_codebooks)
             
             labels = model.decoder.apply_delay_pattern_mask(labels, delay_pattern_mask)
+            len_ = int(all_ratios[idx] * all_lens[idx])
 
             # the first timestamp is associated to a row full of BOS, let's get rid of it
-        batch["labels"] = labels[:, 1:]   
-        return batch
+            sample["labels"] = labels[:, 1:len_]
+            return sample
 
-    with training_args.main_process_first(desc="audio target preprocessing"):
-        # for now on CPU
-        # TODO: enrich for GPU
-        vectorized_datasets = vectorized_datasets.map(
-            apply_audio_decoder,
+        # TODO: done multiple times, how to deal with it.
+        with accelerator.main_process_first():
+            vectorized_datasets[split] = vectorized_datasets[split].map(
+                postprocess_dataset,
                 num_proc=num_workers,
-            desc="preprocess datasets",
+                desc="Postprocessing labeling",
+                with_indices=True,
             )
 
-    # TODO: will it work on GPU ? unmerged for now https://github.com/huggingface/accelerate/pull/2433
-    # for split in vectorized_datasets:
-    #     with distributed_state.split_between_processes(vectorized_datasets[split]["labels"]) as input_labels:
-    #         result = audio_decoder(input_labels)
+            
+    accelerator.free_memory()
+    del generate_labels
+
+
     
     if data_args.add_audio_samples_to_wandb and "wandb" in training_args.report_to:
         if is_wandb_available():
@@ -827,6 +874,7 @@ def main():
     # In a second step ``args.preprocessing_only`` can then be set to `False` to load the
     # cached dataset
     if data_args.preprocessing_only:
+        # TODO: save to disk in this step instead of something else ??
         logger.info(f"Data preprocessing finished. Files cached at {vectorized_datasets.cache_files}")
         return
     
@@ -865,9 +913,9 @@ def main():
 
     # Now save everything to be able to create a single processor later
     # make sure all processes wait until data is saved
-    with training_args.main_process_first():
+    with accelerator.main_process_first():
         # only the main process saves them
-        if is_main_process(training_args.local_rank):
+        if accelerator.is_main_process:
             # save feature extractor, tokenizer and config
             if model_args.prompt_tokenizer_name is None and model_args.description_tokenizer_name or (model_args.prompt_tokenizer_name==model_args.description_tokenizer_name):
                 prompt_tokenizer.save_pretrained(training_args.output_dir)
@@ -936,7 +984,7 @@ def main():
                 audios = predictions["audio"]
 
                 # log the table to wandb
-                self._wandb.log({"sample_songs": [self._wandb.Audio(audio, caption=text, sample_rate=audio_encoder_feature_extractor.sampling_rate) for (audio, text) in zip(audios, texts)]})
+                self._wandb.log({"sample_songs": [self._wandb.Audio(audio, caption=text, sample_rate=sampling_rate) for (audio, text) in zip(audios, texts)]})