add tts stage

egs/speech_llm/SPEECH2SPEECH/qwen_omni/data_module.py

@@ -24,7 +24,7 @@ from pathlib import Path
 from typing import Any, Dict, Optional
 
 import torch
-from datasets import interleave_datasets, load_dataset
+from datasets import interleave_datasets, load_dataset, Audio, Features, Value, Sequence
 from lhotse import (
     CutSet,
     WhisperFbank,
@@ -49,7 +49,9 @@ from lhotse.dataset.input_strategies import (  # noqa F401 For AudioSamples
 from lhotse.utils import fix_random_seed
 from torch.utils.data import DataLoader
 from utils import get_local_rank, str2bool
-
+import io
+import wave
+import random
 
 class _SeedWorkers:
     def __init__(self, seed: int):
@@ -751,4 +753,61 @@ class AsrDataModule:
                 423_000,
                 28539,
             ],
-        )
+        )
+
+    @lru_cache()
+    def train_cuts_emilia_en(self) -> CutSet:
+        logging.info("About to get train cuts")
+        data_path = "/lustre/fsw/general_sa/yuekaiz/s2s" + "/emilia_en"
+        # if self.args.huggingface_dataset_path_or_name is not None:
+        #     data_path = self.args.huggingface_dataset_path_or_name + "/emilia_en"
+        # else:
+        #     data_path = "yuekai/emilia_en"
+
+        emilia_en_data = load_dataset(
+            data_path, split="train", streaming=True
+        )
+
+        def update_wav_path(example):
+            sampling_rate = 16000  # From current_features
+            duration = 1  # seconds, arbitrary duration for random audio
+            num_channels = 1  # mono
+            sample_width = 2  # 2 bytes = 16-bit audio
+
+            num_frames = int(duration * sampling_rate)
+            
+            # Generate random bytes for the PCM data part
+            # This will be random noise, but structurally valid for a WAV file
+            pcm_data = bytes([random.randint(0, 255) for _ in range(num_frames * num_channels * sample_width)])
+
+            # Create a WAV file in memory
+            audio_buffer = io.BytesIO()
+            with wave.open(audio_buffer, 'wb') as wf:
+                wf.setnchannels(num_channels)
+                wf.setsampwidth(sample_width)
+                wf.setframerate(sampling_rate)
+                wf.writeframes(pcm_data) # writeframes expects bytes
+            
+            example["wav"] = audio_buffer.getvalue()
+            return example
+
+        emilia_en_data = emilia_en_data.map(update_wav_path)
+        current_features = Features({
+            'id': Value('string'),
+            'text': Value('string'),
+            'duration': Value('float'),
+            'language': Value('string'),
+            'dnsmos': Value('float'),
+            'speech_token': Sequence(Value('int32')),
+            'wav': Audio(sampling_rate=16000)
+
+        })
+        emilia_en_data = emilia_en_data.rename_column("code", "speech_token")
+        emilia_en_data = emilia_en_data.cast(current_features)
+
+        emilia_en_train_cuts = CutSet.from_huggingface_dataset(
+            emilia_en_data, # Adjusted from instruct_s2s_train
+            audio_key="wav",
+            text_key="text",
+        )
+        return emilia_en_train_cuts 

egs/speech_llm/SPEECH2SPEECH/qwen_omni/train.py

@@ -1032,6 +1032,9 @@ def run(rank, world_size, args):
     elif params.dataset == "gigaspeech":
         train_cuts = data_module.train_cuts_gigaspeech()
         valid_cuts = data_module.valid_cuts_ultravox()
+    elif params.dataset == "emilia_en":
+        train_cuts = data_module.train_cuts_emilia_en()
+        valid_cuts = data_module.valid_cuts_emilia_en()
     else:
         raise ValueError(f"Unknown dataset: {params.dataset}")