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remove cache router

root vor 2 Monaten
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Commit
79116ac32e

+ 164 - 30
runtime/triton_trtllm/client_grpc.py

@@ -59,12 +59,14 @@ import tritonclient.grpc.aio as grpcclient_aio  # Renamed original import
 import tritonclient.grpc as grpcclient_sync  # Added sync client import
 from tritonclient.utils import np_to_triton_dtype, InferenceServerException  # Added InferenceServerException
 
+from datetime import datetime
 
 # --- Added UserData and callback ---
 class UserData:
     def __init__(self):
         self._completed_requests = queue.Queue()
         self._first_chunk_time = None
+        self._second_chunk_time = None
         self._start_time = None
 
     def record_start_time(self):
@@ -75,14 +77,44 @@ class UserData:
             return self._first_chunk_time - self._start_time
         return None
 
+    def get_second_chunk_latency(self):
+        if self._first_chunk_time and self._second_chunk_time:
+            return self._second_chunk_time - self._first_chunk_time
+        return None
+
 
 def callback(user_data, result, error):
-    if user_data._first_chunk_time is None and not error:
-        user_data._first_chunk_time = time.time()  # Record time of first successful chunk
+    if not error:
+        if user_data._first_chunk_time is None:
+            user_data._first_chunk_time = time.time()  # Record time of first successful chunk
+        elif user_data._second_chunk_time is None:
+            user_data._second_chunk_time = time.time()
+
     if error:
         user_data._completed_requests.put(error)
     else:
         user_data._completed_requests.put(result)
+
+
+def stream_callback(user_data_map, result, error):
+    request_id = None
+    if error:
+        # Note: InferenceServerException doesn't have a public request_id() method in all versions.
+        # This part might need adjustment depending on the tritonclient library version.
+        # A more robust way would be to wrap the error with the request_id if possible.
+        # For now, we assume we can't get request_id from error and it will timeout on the client side.
+        print(f"An error occurred in the stream callback: {error}")
+    else:
+        request_id = result.get_response().id
+
+    if request_id:
+        user_data = user_data_map.get(request_id)
+        if user_data:
+            callback(user_data, result, error)
+        else:
+            print(f"Warning: Could not find user_data for request_id {request_id}")
+
+
 # --- End Added UserData and callback ---
 
 
@@ -142,6 +174,68 @@ def write_triton_stats(stats, summary_file):
                 )
 
 
+def subtract_stats(stats_after, stats_before):
+    """Subtracts two Triton inference statistics objects."""
+    # Deep copy to avoid modifying the original stats_after
+    stats_diff = json.loads(json.dumps(stats_after))
+
+    model_stats_before_map = {
+        s["name"]: {
+            "version": s["version"],
+            "last_inference": s.get("last_inference", 0),
+            "inference_count": s.get("inference_count", 0),
+            "execution_count": s.get("execution_count", 0),
+            "inference_stats": s.get("inference_stats", {}),
+            "batch_stats": s.get("batch_stats", []),
+        }
+        for s in stats_before["model_stats"]
+    }
+
+    for model_stat_after in stats_diff["model_stats"]:
+        model_name = model_stat_after["name"]
+        if model_name in model_stats_before_map:
+            model_stat_before = model_stats_before_map[model_name]
+
+            # Subtract counts
+            model_stat_after["inference_count"] = str(
+                int(model_stat_after.get("inference_count", 0)) - int(model_stat_before.get("inference_count", 0))
+            )
+            model_stat_after["execution_count"] = str(
+                int(model_stat_after.get("execution_count", 0)) - int(model_stat_before.get("execution_count", 0))
+            )
+
+            # Subtract aggregate stats (like queue, compute times)
+            if "inference_stats" in model_stat_after and "inference_stats" in model_stat_before:
+                for key in ["success", "fail", "queue", "compute_input", "compute_infer", "compute_output", "cache_hit", "cache_miss"]:
+                    if key in model_stat_after["inference_stats"] and key in model_stat_before["inference_stats"]:
+                        if "ns" in model_stat_after["inference_stats"][key]:
+                            ns_after = int(model_stat_after["inference_stats"][key]["ns"])
+                            ns_before = int(model_stat_before["inference_stats"][key]["ns"])
+                            model_stat_after["inference_stats"][key]["ns"] = str(ns_after - ns_before)
+                        if "count" in model_stat_after["inference_stats"][key]:
+                            count_after = int(model_stat_after["inference_stats"][key]["count"])
+                            count_before = int(model_stat_before["inference_stats"][key]["count"])
+                            model_stat_after["inference_stats"][key]["count"] = str(count_after - count_before)
+
+            # Subtract batch execution stats
+            if "batch_stats" in model_stat_after and "batch_stats" in model_stat_before:
+                batch_stats_before_map = {b["batch_size"]: b for b in model_stat_before["batch_stats"]}
+                for batch_stat_after in model_stat_after["batch_stats"]:
+                    bs = batch_stat_after["batch_size"]
+                    if bs in batch_stats_before_map:
+                        batch_stat_before = batch_stats_before_map[bs]
+                        for key in ["compute_input", "compute_infer", "compute_output"]:
+                            if key in batch_stat_after and key in batch_stat_before:
+                                count_after = int(batch_stat_after[key]["count"])
+                                count_before = int(batch_stat_before[key]["count"])
+                                batch_stat_after[key]["count"] = str(count_after - count_before)
+
+                                ns_after = int(batch_stat_after[key]["ns"])
+                                ns_before = int(batch_stat_before[key]["ns"])
+                                batch_stat_after[key]["ns"] = str(ns_after - ns_before)
+    return stats_diff
+
+
 def get_args():
     parser = argparse.ArgumentParser(formatter_class=argparse.ArgumentDefaultsHelpFormatter)
 
@@ -357,10 +451,10 @@ def run_sync_streaming_inference(
     """Helper function to run the blocking sync streaming call."""
     start_time_total = time.time()
     user_data.record_start_time()  # Record start time for first chunk latency calculation
+    # e.g. 08:47:34.827758
 
-    # Establish stream
-    sync_triton_client.start_stream(callback=functools.partial(callback, user_data))
-
+    print(f"Record start time in human readable: {datetime.now()}")
+    # input()
     # Send request
     sync_triton_client.async_stream_infer(
         model_name,
@@ -374,11 +468,11 @@ def run_sync_streaming_inference(
     audios = []
     while True:
         try:
-            result = user_data._completed_requests.get()  # Add timeout
+            result = user_data._completed_requests.get(timeout=20)  # Add timeout
             if isinstance(result, InferenceServerException):
                 print(f"Received InferenceServerException: {result}")
-                sync_triton_client.stop_stream()
-                return None, None, None  # Indicate error
+                # Don't stop the stream here, just return error
+                return None, None, None, None
             # Get response metadata
             response = result.get_response()
             final = response.parameters["triton_final_response"].bool_param
@@ -393,13 +487,13 @@ def run_sync_streaming_inference(
 
         except queue.Empty:
             print(f"Timeout waiting for response for request id {request_id}")
-            sync_triton_client.stop_stream()
-            return None, None, None  # Indicate error
+            # Don't stop stream here, just return error
+            return None, None, None, None
 
-    sync_triton_client.stop_stream()
     end_time_total = time.time()
     total_request_latency = end_time_total - start_time_total
     first_chunk_latency = user_data.get_first_chunk_latency()
+    second_chunk_latency = user_data.get_second_chunk_latency()
 
     # Reconstruct audio using cross-fade (from client_grpc_streaming.py)
     actual_duration = 0
@@ -448,7 +542,7 @@ def run_sync_streaming_inference(
         print("Warning: No audio chunks received.")
         actual_duration = 0
 
-    return total_request_latency, first_chunk_latency, actual_duration
+    return total_request_latency, first_chunk_latency, second_chunk_latency, actual_duration
 
 
 async def send_streaming(
@@ -468,10 +562,12 @@ async def send_streaming(
     latency_data = []
     task_id = int(name[5:])
     sync_triton_client = None  # Initialize client variable
+    user_data_map = {}
 
     try:  # Wrap in try...finally to ensure client closing
         print(f"{name}: Initializing sync client for streaming...")
         sync_triton_client = grpcclient_sync.InferenceServerClient(url=server_url, verbose=False)  # Create client here
+        sync_triton_client.start_stream(callback=functools.partial(stream_callback, user_data_map))
 
         print(f"{name}: Starting streaming processing for {len(manifest_item_list)} items.")
         for i, item in enumerate(manifest_item_list):
@@ -494,10 +590,11 @@ async def send_streaming(
 
                 request_id = str(uuid.uuid4())
                 user_data = UserData()
+                user_data_map[request_id] = user_data
 
                 audio_save_path = os.path.join(audio_save_dir, f"{item['target_audio_path']}.wav")
-
-                total_request_latency, first_chunk_latency, actual_duration = await asyncio.to_thread(
+                print("target_text: ", target_text, "time: ", datetime.now())
+                total_request_latency, first_chunk_latency, second_chunk_latency, actual_duration = await asyncio.to_thread(
                     run_sync_streaming_inference,
                     sync_triton_client,
                     model_name,
@@ -511,12 +608,18 @@ async def send_streaming(
                 )
 
                 if total_request_latency is not None:
-                    print(f"{name}: Item {i} - First Chunk Latency: {first_chunk_latency:.4f}s, Total Latency: {total_request_latency:.4f}s, Duration: {actual_duration:.4f}s")
-                    latency_data.append((total_request_latency, first_chunk_latency, actual_duration))
+                    print(
+                        f"{name}: Item {i} - First Chunk Latency: {first_chunk_latency:.4f}s, "
+                        f"Second Chunk Latency: {second_chunk_latency if second_chunk_latency is not None else 'N/A'}, "
+                        f"Total Latency: {total_request_latency:.4f}s, Duration: {actual_duration:.4f}s"
+                    )
+                    latency_data.append((total_request_latency, first_chunk_latency, second_chunk_latency, actual_duration))
                     total_duration += actual_duration
                 else:
                     print(f"{name}: Item {i} failed.")
 
+                del user_data_map[request_id]
+
             except FileNotFoundError:
                 print(f"Error: Audio file not found for item {i}: {item['audio_filepath']}")
             except Exception as e:
@@ -527,7 +630,8 @@ async def send_streaming(
     finally:  # Ensure client is closed
         if sync_triton_client:
             try:
-                print(f"{name}: Closing sync client...")
+                print(f"{name}: Closing stream and sync client...")
+                sync_triton_client.stop_stream()
                 sync_triton_client.close()
             except Exception as e:
                 print(f"{name}: Error closing sync client: {e}")
@@ -685,9 +789,22 @@ async def main():
                     "target_text": dataset[i]["target_text"],
                 }
             )
+        # manifest_item_list = manifest_item_list[:4]
     else:
         manifest_item_list = load_manifests(args.manifest_path)
 
+    # --- Statistics Fetching (Before) ---
+    stats_client = None
+    stats_before = None
+    try:
+        print("Initializing temporary async client for fetching stats...")
+        stats_client = grpcclient_aio.InferenceServerClient(url=url, verbose=False)
+        print("Fetching inference statistics before running tasks...")
+        stats_before = await stats_client.get_inference_statistics(model_name="", as_json=True)
+    except Exception as e:
+        print(f"Could not retrieve statistics before running tasks: {e}")
+    # --- End Statistics Fetching (Before) ---
+
     num_tasks = min(args.num_tasks, len(manifest_item_list))
     manifest_item_list = split_data(manifest_item_list, num_tasks)
 
@@ -776,8 +893,9 @@ async def main():
 
         elif args.mode == "streaming":
             # Calculate stats for total request latency and first chunk latency
-            total_latency_list = [total for (total, first, duration) in latency_data if total is not None]
-            first_chunk_latency_list = [first for (total, first, duration) in latency_data if first is not None]
+            total_latency_list = [total for (total, first, second, duration) in latency_data if total is not None]
+            first_chunk_latency_list = [first for (total, first, second, duration) in latency_data if first is not None]
+            second_chunk_latency_list = [second for (total, first, second, duration) in latency_data if second is not None]
 
             s += "\n--- Total Request Latency ---\n"
             if total_latency_list:
@@ -804,6 +922,19 @@ async def main():
                 s += f"average_first_chunk_latency_ms: {avg_first_chunk_latency_ms:.2f}\n"
             else:
                 s += "No first chunk latency data collected (check for errors or if all requests failed before first chunk).\n"
+
+            s += "\n--- Second Chunk Latency ---\n"
+            if second_chunk_latency_list:
+                avg_second_chunk_latency_ms = sum(second_chunk_latency_list) / len(second_chunk_latency_list) * 1000.0
+                variance_second_chunk_latency = np.var(second_chunk_latency_list, dtype=np.float64) * 1000.0
+                s += f"second_chunk_latency_variance: {variance_second_chunk_latency:.2f}\n"
+                s += f"second_chunk_latency_50_percentile_ms: {np.percentile(second_chunk_latency_list, 50) * 1000.0:.2f}\n"
+                s += f"second_chunk_latency_90_percentile_ms: {np.percentile(second_chunk_latency_list, 90) * 1000.0:.2f}\n"
+                s += f"second_chunk_latency_95_percentile_ms: {np.percentile(second_chunk_latency_list, 95) * 1000.0:.2f}\n"
+                s += f"second_chunk_latency_99_percentile_ms: {np.percentile(second_chunk_latency_list, 99) * 1000.0:.2f}\n"
+                s += f"average_second_chunk_latency_ms: {avg_second_chunk_latency_ms:.2f}\n"
+            else:
+                s += "No second chunk latency data collected (check for errors or if all requests failed before second chunk).\n"
     else:
         s += "No latency data collected.\n"
     # --- End Statistics Reporting ---
@@ -822,20 +953,23 @@ async def main():
 
     # --- Statistics Fetching using temporary Async Client ---
     # Use a separate async client for fetching stats regardless of mode
-    stats_client = None
     try:
-        print("Initializing temporary async client for fetching stats...")
-        stats_client = grpcclient_aio.InferenceServerClient(url=url, verbose=False)
-        print("Fetching inference statistics...")
-        # Fetching for all models, filtering might be needed depending on server setup
-        stats = await stats_client.get_inference_statistics(model_name="", as_json=True)
-        print("Fetching model config...")
-        metadata = await stats_client.get_model_config(model_name=args.model_name, as_json=True)
+        if stats_client and stats_before:
+            print("Fetching inference statistics after running tasks...")
+            stats_after = await stats_client.get_inference_statistics(model_name="", as_json=True)
+
+            print("Calculating statistics difference...")
+            stats = subtract_stats(stats_after, stats_before)
 
-        write_triton_stats(stats, f"{args.log_dir}/stats_summary-{name}.txt")
+            print("Fetching model config...")
+            metadata = await stats_client.get_model_config(model_name=args.model_name, as_json=True)
 
-        with open(f"{args.log_dir}/model_config-{name}.json", "w") as f:
-            json.dump(metadata, f, indent=4)
+            write_triton_stats(stats, f"{args.log_dir}/stats_summary-{name}.txt")
+
+            with open(f"{args.log_dir}/model_config-{name}.json", "w") as f:
+                json.dump(metadata, f, indent=4)
+        else:
+            print("Stats client not available or initial stats were not fetched. Skipping stats reporting.")
 
     except Exception as e:
         print(f"Could not retrieve statistics or config: {e}")

+ 2 - 50
runtime/triton_trtllm/model_repo/cosyvoice2_dit/3/model.py

@@ -109,7 +109,6 @@ class TritonPythonModel:
         spk_info = torch.load(spk_info_path, map_location="cpu", weights_only=False)
         self.default_spk_info = spk_info["001"]
         self.http_client = httpx.AsyncClient()
-        self.runtime_cache = {}
 
     def _convert_speech_tokens_to_str(self, speech_tokens: Union[torch.Tensor, List]) -> str:
         """Converts a tensor or list of speech token IDs to a string representation."""
@@ -264,38 +263,11 @@ class TritonPythonModel:
         finalize_tensor = pb_utils.Tensor("finalize", np.array([[finalize]], dtype=np.bool_))
         inputs_tensor = [target_speech_tokens_tensor, reference_wav, reference_wav_len, finalize_tensor]
         
-        # optional cache inputs
-        if self.runtime_cache[request_id]["conformer_cnn_cache"] is not None:
-            # inputs_tensor.extend([
-            #     pb_utils.Tensor("conformer_cnn_cache", self.runtime_cache[request_id]["conformer_cnn_cache"].as_numpy()),
-            #     pb_utils.Tensor("conformer_att_cache", self.runtime_cache[request_id]["conformer_att_cache"].as_numpy()),
-            #     pb_utils.Tensor("estimator_cnn_cache", self.runtime_cache[request_id]["estimator_cnn_cache"].as_numpy()),
-            #     pb_utils.Tensor("estimator_att_cache", self.runtime_cache[request_id]["estimator_att_cache"].as_numpy()),
-            #     pb_utils.Tensor("mel", self.runtime_cache[request_id]["mel"].as_numpy()),
-            #     pb_utils.Tensor("source", self.runtime_cache[request_id]["source"].as_numpy()),
-            #     pb_utils.Tensor("speech", self.runtime_cache[request_id]["speech"].as_numpy()),
-            # ])
-            inputs_tensor.extend([
-                self.runtime_cache[request_id]["conformer_cnn_cache"],
-                self.runtime_cache[request_id]["conformer_att_cache"],
-                self.runtime_cache[request_id]["estimator_cnn_cache"],
-                self.runtime_cache[request_id]["estimator_att_cache"],
-                self.runtime_cache[request_id]["mel"],
-                self.runtime_cache[request_id]["source"],
-                self.runtime_cache[request_id]["speech"],
-            ])
         # Create and execute inference request
         inference_request = pb_utils.InferenceRequest(
             model_name='token2wav_dit',
             requested_output_names=[
                 "waveform",
-                "conformer_cnn_cache",
-                "conformer_att_cache",
-                "estimator_cnn_cache",
-                "estimator_att_cache",
-                "mel",
-                "source",
-                "speech",
             ],
             inputs=inputs_tensor,
             request_id=request_id,
@@ -306,14 +278,6 @@ class TritonPythonModel:
         if inference_response.has_error():
             raise pb_utils.TritonModelException(inference_response.error().message())
 
-        self.runtime_cache[request_id]["conformer_cnn_cache"] = pb_utils.get_output_tensor_by_name(inference_response, "conformer_cnn_cache")
-        self.runtime_cache[request_id]["conformer_att_cache"] = pb_utils.get_output_tensor_by_name(inference_response, "conformer_att_cache")
-        self.runtime_cache[request_id]["estimator_cnn_cache"] = pb_utils.get_output_tensor_by_name(inference_response, "estimator_cnn_cache")
-        self.runtime_cache[request_id]["estimator_att_cache"] = pb_utils.get_output_tensor_by_name(inference_response, "estimator_att_cache")
-        self.runtime_cache[request_id]["mel"] = pb_utils.get_output_tensor_by_name(inference_response, "mel")
-        self.runtime_cache[request_id]["source"] = pb_utils.get_output_tensor_by_name(inference_response, "source")
-        self.runtime_cache[request_id]["speech"] = pb_utils.get_output_tensor_by_name(inference_response, "speech")
-
         # Extract and convert output waveform
         waveform = pb_utils.get_output_tensor_by_name(inference_response, 'waveform')
         waveform = torch.utils.dlpack.from_dlpack(waveform.to_dlpack()).cpu()
@@ -339,16 +303,6 @@ class TritonPythonModel:
 
     async def _process_request(self, request):
         request_id = request.request_id()
-        if request_id not in self.runtime_cache:
-            self.runtime_cache[request_id] = {
-                "conformer_cnn_cache": None,
-                "conformer_att_cache": None,
-                "estimator_cnn_cache": None,
-                "estimator_att_cache": None,
-                "mel": None,
-                "source": None,
-                "speech": None,
-            }
         # Extract input tensors
         wav = pb_utils.get_input_tensor_by_name(request, "reference_wav")
 
@@ -369,7 +323,7 @@ class TritonPythonModel:
 
             reference_text = pb_utils.get_input_tensor_by_name(request, "reference_text").as_numpy()
             reference_text = reference_text[0][0].decode('utf-8')
-            prompt_spk_embedding = self.forward_speaker_embedding(wav_tensor)
+            # prompt_spk_embedding = self.forward_speaker_embedding(wav_tensor)
 
             # reference_text = self.default_spk_info["prompt_text"]
             # prompt_speech_tokens = self.default_spk_info["speech_token"] + ORIGINAL_VOCAB_SIZE
@@ -453,9 +407,7 @@ class TritonPythonModel:
             audio_tensor = pb_utils.Tensor.from_dlpack("waveform", to_dlpack(sub_tts_speech))
             inference_response = pb_utils.InferenceResponse(output_tensors=[audio_tensor])
             response_sender.send(inference_response)
-            if request_id in self.runtime_cache:
-                del self.runtime_cache[request_id]
-                self.logger.log_info(f"Deleted cache for request_id: {request_id}")
+
             response_sender.send(flags=pb_utils.TRITONSERVER_RESPONSE_COMPLETE_FINAL)
             self.logger.log_info("send tritonserver_response_complete_final to end")
         else:

+ 1 - 1
runtime/triton_trtllm/model_repo/cosyvoice2_dit/config.pbtxt

@@ -31,7 +31,7 @@ parameters [
    value: {string_value:"${model_dir}"}
   }
 ]
-parameters: { key: "FORCE_CPU_ONLY_INPUT_TENSORS" value: {string_value:"no"}}
+
 input [
   {
     name: "reference_wav"

+ 30 - 74
runtime/triton_trtllm/model_repo/token2wav_dit/1/model.py

@@ -103,91 +103,47 @@ class TritonPythonModel:
             List of inference responses containing generated waveforms
         """
         responses = []
+        # Process each request in batch
         for request in requests:
-            request_id = request.request_id()
-
-            # Get inputs
-            target_speech_tokens_tensor = pb_utils.get_input_tensor_by_name(request, "target_speech_tokens")
-            target_speech_tokens = torch.utils.dlpack.from_dlpack(target_speech_tokens_tensor.to_dlpack())
+            target_speech_tokens_tensor = pb_utils.get_input_tensor_by_name(request, "target_speech_tokens").as_numpy()
+            target_speech_tokens = torch.from_numpy(target_speech_tokens_tensor)#.to(self.device)
+            # shift the speech tokens according to the original vocab size
+            target_speech_tokens = target_speech_tokens - ORIGINAL_VOCAB_SIZE
             target_speech_tokens = target_speech_tokens.squeeze().tolist()
 
+            # We set token_offset as an optional input to support streaming/offline tts. It has to be None when offline tts.
+           
             finalize = pb_utils.get_input_tensor_by_name(request, "finalize").as_numpy().item()
-            wav_array = pb_utils.get_input_tensor_by_name(request, "reference_wav").as_numpy()
-            wav_len = pb_utils.get_input_tensor_by_name(request, "reference_wav_len").as_numpy().item()
-            wav = torch.from_numpy(wav_array)[:, :wav_len].squeeze(0)
+                
+            request_id = request.request_id()
+               
+
+            wav_array = pb_utils.get_input_tensor_by_name(
+                request, "reference_wav").as_numpy()
+            wav_len = pb_utils.get_input_tensor_by_name(
+                request, "reference_wav_len").as_numpy().item()
+
+            wav_array = torch.from_numpy(wav_array)
+            # Prepare inputs
+            wav = wav_array[:, :wav_len].squeeze(0)
+
             spk_id = get_spk_id_from_prompt_audio(wav)
+            # wav = wav.to(self.device)
 
-            # Handle cache
-            conformer_cnn_cache = pb_utils.get_input_tensor_by_name(request, "conformer_cnn_cache")
-            if conformer_cnn_cache is not None:
-                self.token2wav_model.streaming_flow_cache[request_id]['conformer_cnn_cache'] = torch.utils.dlpack.from_dlpack(conformer_cnn_cache.to_dlpack())
-                
-                conformer_att_cache_np = pb_utils.get_input_tensor_by_name(request, "conformer_att_cache")
-                self.token2wav_model.streaming_flow_cache[request_id]['conformer_att_cache'] = torch.utils.dlpack.from_dlpack(conformer_att_cache_np.to_dlpack()).transpose(0,1)
-                
-                estimator_cnn_cache_np = pb_utils.get_input_tensor_by_name(request, "estimator_cnn_cache")
-                self.token2wav_model.streaming_flow_cache[request_id]['estimator_cnn_cache'] = torch.utils.dlpack.from_dlpack(estimator_cnn_cache_np.to_dlpack()).squeeze(0)
+            # update cache before forward
+            # self.token2wav_model.streaming_flow_cache[request_id]
+            # self.token2wav_model.hift_cache_dict[request_id]
 
-                estimator_att_cache_np = pb_utils.get_input_tensor_by_name(request, "estimator_att_cache")
-                self.token2wav_model.streaming_flow_cache[request_id]['estimator_att_cache'] = torch.utils.dlpack.from_dlpack(estimator_att_cache_np.to_dlpack()).squeeze(0)
+            audio_hat = self.token2wav_model.forward_streaming(target_speech_tokens, finalize, request_id=request_id, speaker_id=f"{spk_id}", prompt_audio=wav, prompt_audio_sample_rate=16000)
 
-                mel_np = pb_utils.get_input_tensor_by_name(request, "mel")
-                self.token2wav_model.streaming_flow_cache[request_id]['mel'] = torch.utils.dlpack.from_dlpack(mel_np.to_dlpack())
-                
-                source_np = pb_utils.get_input_tensor_by_name(request, "source")
-                self.token2wav_model.hift_cache_dict[request_id]['source'] = torch.utils.dlpack.from_dlpack(source_np.to_dlpack())
-                
-                speech_np = pb_utils.get_input_tensor_by_name(request, "speech")
-                self.token2wav_model.hift_cache_dict[request_id]['speech'] = torch.utils.dlpack.from_dlpack(speech_np.to_dlpack())
-
-            # Forward pass
-            audio_hat = self.token2wav_model.forward_streaming(
-                target_speech_tokens, 
-                finalize, 
-                request_id=request_id, 
-                speaker_id=f"{spk_id}", 
-                prompt_audio=wav, 
-                prompt_audio_sample_rate=16000
-            )
-            
-            # Prepare outputs
+            # get the cache after forward
             outputs = []
+
+            generated_wave = audio_hat.squeeze(0).cpu().numpy()
+
             wav_tensor = pb_utils.Tensor.from_dlpack("waveform", to_dlpack(audio_hat))
             outputs.append(wav_tensor)
-            
-            if request_id in self.token2wav_model.streaming_flow_cache:
-                cache = self.token2wav_model.streaming_flow_cache[request_id]
-                hifigan_cache = self.token2wav_model.hift_cache_dict[request_id]
-                conformer_cnn_cache = cache['conformer_cnn_cache']
-                conformer_att_cache = cache['conformer_att_cache'].transpose(0,1)
-                estimator_cnn_cache = cache['estimator_cnn_cache'].unsqueeze(0)
-                estimator_att_cache = cache['estimator_att_cache'].unsqueeze(0)
-                mel = hifigan_cache['mel']
-                source = hifigan_cache['source']
-                speech = hifigan_cache['speech']
-
-                outputs.extend([
-                    pb_utils.Tensor.from_dlpack("conformer_cnn_cache", to_dlpack(conformer_cnn_cache)),
-                    pb_utils.Tensor.from_dlpack("conformer_att_cache", to_dlpack(conformer_att_cache)),
-                    pb_utils.Tensor.from_dlpack("estimator_cnn_cache", to_dlpack(estimator_cnn_cache)),
-                    pb_utils.Tensor.from_dlpack("estimator_att_cache", to_dlpack(estimator_att_cache)),
-                    pb_utils.Tensor.from_dlpack("mel", to_dlpack(mel)),
-                    pb_utils.Tensor.from_dlpack("source", to_dlpack(source)),
-                    pb_utils.Tensor.from_dlpack("speech", to_dlpack(speech)),
-                ])
-            else:
-                outputs.extend([pb_utils.Tensor("conformer_cnn_cache", np.array([], dtype=np.float16)),
-                pb_utils.Tensor("conformer_att_cache", np.array([], dtype=np.float16)),
-                pb_utils.Tensor("estimator_cnn_cache", np.array([], dtype=np.float16)),
-                pb_utils.Tensor("estimator_att_cache", np.array([], dtype=np.float16)),
-                pb_utils.Tensor("mel", np.array([], dtype=np.float32)),
-                pb_utils.Tensor("source", np.array([], dtype=np.float32)),
-                pb_utils.Tensor("speech", np.array([], dtype=np.float32)),
-                ])
-
             inference_response = pb_utils.InferenceResponse(output_tensors=outputs)
             responses.append(inference_response)
-        return responses
 
-    def finalize(self):
-        self.logger.log_info("Finalizing Token2WavDiT model")
+        return responses

+ 0 - 78
runtime/triton_trtllm/model_repo/token2wav_dit/config.pbtxt

@@ -22,7 +22,6 @@ dynamic_batching {
     default_priority_level: 10
 }
 
-parameters: { key: "FORCE_CPU_ONLY_INPUT_TENSORS" value: {string_value:"no"}}
 parameters [
   {
    key: "model_dir",
@@ -52,48 +51,6 @@ input [
     dims: [ 1 ]
     reshape: { shape: [ ] }
     optional: true
-  },
-  {
-    name: "conformer_cnn_cache"
-    data_type: TYPE_FP16
-    dims: [ 512, -1 ]
-    optional: true
-  },
-  {
-    name: "conformer_att_cache"
-    data_type: TYPE_FP16
-    dims: [ 10, 8, -1, 128 ]
-    optional: true
-  },
-  {
-    name: "estimator_cnn_cache"
-    data_type: TYPE_FP16
-    dims: [ 10, 16, -1, 1024, 2 ]
-    optional: true
-  },
-  {
-    name: "estimator_att_cache"
-    data_type: TYPE_FP16
-    dims: [ 10, 16, -1, 8, -1, 128 ]
-    optional: true
-  },
-  {
-    name: "mel"
-    data_type: TYPE_FP32
-    dims: [ 80, -1 ]
-    optional: true
-  },
-  {
-    name: "source"
-    data_type: TYPE_FP32
-    dims: [ 1, -1 ]
-    optional: true
-  },
-  {
-    name: "speech"
-    data_type: TYPE_FP32
-    dims: [ -1 ]
-    optional: true
   }
 ]
 output [
@@ -101,41 +58,6 @@ output [
     name: "waveform"
     data_type: TYPE_FP32
     dims: [ -1 ]
-  },
-  {
-    name: "conformer_cnn_cache"
-    data_type: TYPE_FP16
-    dims: [ 512, -1 ]
-  },
-  {
-    name: "conformer_att_cache"
-    data_type: TYPE_FP16
-    dims: [ 10, 8, -1, 128 ]
-  },
-  {
-    name: "estimator_cnn_cache"
-    data_type: TYPE_FP16
-    dims: [ 10, 16, -1, 1024, 2 ]
-  },
-  {
-    name: "estimator_att_cache"
-    data_type: TYPE_FP16
-    dims: [ 10, 16, -1, 8, -1, 128 ]
-  },
-  {
-    name: "mel"
-    data_type: TYPE_FP32
-    dims: [ 80, -1 ]
-  },
-  {
-    name: "source"
-    data_type: TYPE_FP32
-    dims: [ 1, -1 ]
-  },
-  {
-    name: "speech"
-    data_type: TYPE_FP32
-    dims: [ -1 ]
   }
 ]
 

+ 5 - 5
runtime/triton_trtllm/run_stepaudio2_dit_token2wav.sh

@@ -1,6 +1,6 @@
 #!/bin/bash
 # Copyright (c) 2025 NVIDIA (authors: Yuekai Zhang)
-export CUDA_VISIBLE_DEVICES=1
+export CUDA_VISIBLE_DEVICES=0
 cosyvoice_path=/workspace/CosyVoice
 cosyvoice_path=/workspace_yuekai/tts/CosyVoice
 stepaudio2_path=/workspace_yuekai/tts/Step-Audio2
@@ -112,7 +112,7 @@ if [ $stage -le 2 ] && [ $stop_stage -ge 2 ]; then
     MODEL_DIR=$model_scope_model_local_dir
     LLM_TOKENIZER_DIR=$huggingface_model_local_dir
     BLS_INSTANCE_NUM=4
-    TRITON_MAX_BATCH_SIZE=32
+    TRITON_MAX_BATCH_SIZE=1
     DECOUPLED_MODE=True # True for streaming, False for offline
     STEP_AUDIO_MODEL_DIR=/workspace_yuekai/tts/CosyVoice/runtime/triton_trtllm/Step-Audio-2-mini/token2wav
 
@@ -154,7 +154,7 @@ if [ $stage -le 5 ] && [ $stop_stage -ge 5 ]; then
         --num-tasks $num_task \
         --mode $mode \
         --huggingface-dataset yuekai/seed_tts_cosy2 \
-        --log-dir ./log_concurrent_tasks_${num_task}_${mode}_bls_${BLS_INSTANCE_NUM}_no_att_cnn_cache_new
+        --log-dir ./log_debug_concurrent_tasks_${num_task}_${mode}_bls_${BLS_INSTANCE_NUM}
 fi
 
 if [ $stage -le 6 ] && [ $stop_stage -ge 6 ]; then
@@ -185,14 +185,14 @@ fi
 
 if [ $stage -le 7 ] && [ $stop_stage -ge 7 ]; then
 
-   python3 streaming_inference.py
+   CUDA_VISIBLE_DEVICES=2 python3 streaming_inference.py --enable-trt --strategy exponential
 
 
 fi
 
 
 if [ $stage -le 8 ] && [ $stop_stage -ge 8 ]; then
-    mpirun -np 1 --allow-run-as-root --oversubscribe trtllm-serve serve --tokenizer $huggingface_model_local_dir $trt_engines_dir --max_batch_size 16 
+    CUDA_VISIBLE_DEVICES=0 mpirun -np 1 --allow-run-as-root --oversubscribe trtllm-serve serve --tokenizer $huggingface_model_local_dir $trt_engines_dir --max_batch_size 16  --kv_cache_free_gpu_memory_fraction 0.4
     
 fi
 

+ 17 - 5
runtime/triton_trtllm/streaming_inference.py

@@ -31,6 +31,7 @@ def get_args():
     parser.add_argument("--output-dir", type=str, default="generated_wavs")
     parser.add_argument("--huggingface-dataset-split", type=str, default="wenetspeech4tts")
     parser.add_argument("--dataset-name", type=str, default="yuekai/seed_tts_cosy2")
+    parser.add_argument("--strategy", type=str, default="equal", choices=["equal", "exponential"])
     return parser.parse_args()
 
 
@@ -53,12 +54,14 @@ if __name__ == "__main__":
     token2wav_model = CosyVoice2_Token2Wav(model_dir=args.model_dir, enable_trt=args.enable_trt, streaming=True)
     
     flow_pre_lookahead_len = 3
-    CHUNK_SIZE = 25
+    CHUNK_SIZE = 15
+    token_frame_rate = 25
     OVERLAP_SIZE = 0
 
     warmup_times = 3
     for _ in range(warmup_times):
         start_time = time.time()
+        total_forward_count = 0
         for batch in data_loader:
             tts_speech_list = []
             ids, generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate, prompt_speech_tokens_list, prompt_text_list = batch
@@ -83,17 +86,26 @@ if __name__ == "__main__":
     
             buffer = generated_speech_tokens
             output_wavs = []
+            chunk_index = 0
             while True:
+                if args.strategy == "equal":
+                    this_chunk_size = CHUNK_SIZE
+                elif args.strategy == "exponential":
+                    this_chunk_size = token_frame_rate * (2 ** chunk_index)
 
-                if len(buffer) >= CHUNK_SIZE + token2wav_model.flow.pre_lookahead_len:
-                    wavs = token2wav_model.forward_streaming(buffer[:CHUNK_SIZE + token2wav_model.flow.pre_lookahead_len], False, request_id=id, speaker_id=f"{id}", prompt_audio=prompt_audio, prompt_audio_sample_rate=prompt_audio_sample_rate)
-                    buffer = buffer[CHUNK_SIZE - OVERLAP_SIZE:]
+                if len(buffer) >= this_chunk_size + token2wav_model.flow.pre_lookahead_len:
+                    wavs = token2wav_model.forward_streaming(buffer[:this_chunk_size + token2wav_model.flow.pre_lookahead_len], False, request_id=id, speaker_id=f"{id}", prompt_audio=prompt_audio, prompt_audio_sample_rate=prompt_audio_sample_rate)
+                    buffer = buffer[this_chunk_size - OVERLAP_SIZE:]
 
                     output_wavs.append(wavs)
+                    total_forward_count += 1
+                    chunk_index += 1
 
                 else:
                     wavs = token2wav_model.forward_streaming(buffer, True, request_id=id, speaker_id=f"{id}", prompt_audio=prompt_audio, prompt_audio_sample_rate=prompt_audio_sample_rate)
                     output_wavs.append(wavs)
+                    total_forward_count += 1
+                    # chunk_index += 1
                     break
 
             for i, wav in enumerate(output_wavs):
@@ -112,4 +124,4 @@ if __name__ == "__main__":
         if _ == 0:
             token2wav_model.speaker_cache = {}
         print(f"Warmup time: {end_time - start_time} seconds")
-
+        print(f"Total forward count: {total_forward_count}")