mirror of
https://github.com/modelscope/FunASR
synced 2025-09-15 14:48:36 +08:00
Merge branch 'dev_infer' of https://github.com/alibaba/FunASR into dev_infer
This commit is contained in:
commit
334dec5d18
@ -68,10 +68,12 @@ Overview
|
||||
./runtime/onnxruntime_python.md
|
||||
./runtime/onnxruntime_cpp.md
|
||||
./runtime/libtorch_python.md
|
||||
./runtime/grpc_python.md
|
||||
./runtime/grpc_cpp.md
|
||||
./runtime/html5.md
|
||||
./runtime/websocket_python.md
|
||||
./runtime/websocket_cpp.md
|
||||
./runtime/grpc_python.md
|
||||
./runtime/grpc_cpp.md
|
||||
|
||||
|
||||
.. toctree::
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:maxdepth: 1
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||||
|
||||
1
docs/runtime/html5.md
Symbolic link
1
docs/runtime/html5.md
Symbolic link
@ -0,0 +1 @@
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||||
../../funasr/runtime/html5/readme.md
|
||||
@ -9,7 +9,7 @@ logger.setLevel(logging.CRITICAL)
|
||||
inference_pipeline = pipeline(
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||||
task=Tasks.punctuation,
|
||||
model='damo/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727',
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output_dir="./tmp/"
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model_revision = 'v1.0.2'
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||||
)
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||||
##################text二进制数据#####################
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@ -762,23 +762,6 @@ class Speech2TextParaformerOnline:
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feats_len = speech_lengths
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if feats.shape[1] != 0:
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if cache_en["is_final"]:
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||||
if feats.shape[1] + cache_en["chunk_size"][2] < cache_en["chunk_size"][1]:
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||||
cache_en["last_chunk"] = True
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else:
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# first chunk
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feats_chunk1 = feats[:, :cache_en["chunk_size"][1], :]
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feats_len = torch.tensor([feats_chunk1.shape[1]])
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results_chunk1 = self.infer(feats_chunk1, feats_len, cache)
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# last chunk
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cache_en["last_chunk"] = True
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feats_chunk2 = feats[:, -(feats.shape[1] + cache_en["chunk_size"][2] - cache_en["chunk_size"][1]):, :]
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feats_len = torch.tensor([feats_chunk2.shape[1]])
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results_chunk2 = self.infer(feats_chunk2, feats_len, cache)
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return [" ".join(results_chunk1 + results_chunk2)]
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results = self.infer(feats, feats_len, cache)
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return results
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@ -36,6 +36,8 @@ def main(args=None, cmd=None):
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from funasr.tasks.asr import ASRTaskParaformer as ASRTask
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if args.mode == "uniasr":
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from funasr.tasks.asr import ASRTaskUniASR as ASRTask
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if args.mode == "rnnt":
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from funasr.tasks.asr import ASRTransducerTask as ASRTask
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ASRTask.main(args=args, cmd=cmd)
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@ -19,12 +19,15 @@ from funasr.models.decoder.transformer_decoder import (
|
||||
)
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from funasr.models.decoder.transformer_decoder import ParaformerDecoderSAN
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from funasr.models.decoder.transformer_decoder import TransformerDecoder
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from funasr.models.decoder.rnnt_decoder import RNNTDecoder
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from funasr.models.joint_net.joint_network import JointNetwork
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from funasr.models.e2e_asr import ASRModel
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from funasr.models.e2e_asr_mfcca import MFCCA
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from funasr.models.e2e_asr_paraformer import Paraformer, ParaformerBert, BiCifParaformer, ContextualParaformer
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from funasr.models.e2e_tp import TimestampPredictor
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from funasr.models.e2e_uni_asr import UniASR
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from funasr.models.encoder.conformer_encoder import ConformerEncoder
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from funasr.models.e2e_asr_transducer import TransducerModel, UnifiedTransducerModel
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from funasr.models.encoder.conformer_encoder import ConformerEncoder, ConformerChunkEncoder
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from funasr.models.encoder.data2vec_encoder import Data2VecEncoder
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from funasr.models.encoder.mfcca_encoder import MFCCAEncoder
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from funasr.models.encoder.rnn_encoder import RNNEncoder
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@ -97,6 +100,7 @@ encoder_choices = ClassChoices(
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sanm_chunk_opt=SANMEncoderChunkOpt,
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data2vec_encoder=Data2VecEncoder,
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||||
mfcca_enc=MFCCAEncoder,
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chunk_conformer=ConformerChunkEncoder,
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||||
),
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default="rnn",
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)
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@ -171,6 +175,23 @@ stride_conv_choices = ClassChoices(
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default="stride_conv1d",
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||||
optional=True,
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||||
)
|
||||
rnnt_decoder_choices = ClassChoices(
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||||
name="rnnt_decoder",
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||||
classes=dict(
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||||
rnnt=RNNTDecoder,
|
||||
),
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||||
default="rnnt",
|
||||
optional=True,
|
||||
)
|
||||
joint_network_choices = ClassChoices(
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||||
name="joint_network",
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||||
classes=dict(
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||||
joint_network=JointNetwork,
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||||
),
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default="joint_network",
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optional=True,
|
||||
)
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class_choices_list = [
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# --frontend and --frontend_conf
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frontend_choices,
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@ -194,6 +215,10 @@ class_choices_list = [
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predictor_choices2,
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# --stride_conv and --stride_conv_conf
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stride_conv_choices,
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# --rnnt_decoder and --rnnt_decoder_conf
|
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rnnt_decoder_choices,
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# --joint_network and --joint_network_conf
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joint_network_choices,
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]
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|
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@ -342,6 +367,63 @@ def build_asr_model(args):
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token_list=token_list,
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||||
**args.model_conf,
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||||
)
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||||
elif args.model == "rnnt":
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# 5. Decoder
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encoder_output_size = encoder.output_size()
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|
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rnnt_decoder_class = rnnt_decoder_choices.get_class(args.rnnt_decoder)
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decoder = rnnt_decoder_class(
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vocab_size,
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**args.rnnt_decoder_conf,
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)
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decoder_output_size = decoder.output_size
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if getattr(args, "decoder", None) is not None:
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att_decoder_class = decoder_choices.get_class(args.decoder)
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|
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att_decoder = att_decoder_class(
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vocab_size=vocab_size,
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encoder_output_size=encoder_output_size,
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**args.decoder_conf,
|
||||
)
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else:
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att_decoder = None
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||||
# 6. Joint Network
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joint_network = JointNetwork(
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vocab_size,
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encoder_output_size,
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decoder_output_size,
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**args.joint_network_conf,
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||||
)
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||||
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||||
# 7. Build model
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||||
if hasattr(encoder, 'unified_model_training') and encoder.unified_model_training:
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model = UnifiedTransducerModel(
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||||
vocab_size=vocab_size,
|
||||
token_list=token_list,
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||||
frontend=frontend,
|
||||
specaug=specaug,
|
||||
normalize=normalize,
|
||||
encoder=encoder,
|
||||
decoder=decoder,
|
||||
att_decoder=att_decoder,
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||||
joint_network=joint_network,
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||||
**args.model_conf,
|
||||
)
|
||||
|
||||
else:
|
||||
model = TransducerModel(
|
||||
vocab_size=vocab_size,
|
||||
token_list=token_list,
|
||||
frontend=frontend,
|
||||
specaug=specaug,
|
||||
normalize=normalize,
|
||||
encoder=encoder,
|
||||
decoder=decoder,
|
||||
att_decoder=att_decoder,
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||||
joint_network=joint_network,
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||||
**args.model_conf,
|
||||
)
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||||
else:
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||||
raise NotImplementedError("Not supported model: {}".format(args.model))
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||||
|
||||
@ -349,4 +431,4 @@ def build_asr_model(args):
|
||||
if args.init is not None:
|
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initialize(model, args.init)
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||||
|
||||
return model
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||||
return model
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||||
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||||
@ -12,7 +12,7 @@ if __name__ == '__main__':
|
||||
return {'inputs': np.ones((1, text_length), dtype=np.int64),
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||||
'text_lengths': np.array([text_length,], dtype=np.int32),
|
||||
'vad_masks': np.ones((1, 1, text_length, text_length), dtype=np.float32),
|
||||
'sub_masks': np.tril(np.ones((text_length, text_length), dtype=np.float32))[None, None, :, :].astype(np.float32)
|
||||
'sub_masks': np.ones((1, 1, text_length, text_length), dtype=np.float32),
|
||||
}
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||||
|
||||
def _run(feed_dict):
|
||||
|
||||
@ -1078,7 +1078,7 @@ class ConformerChunkEncoder(AbsEncoder):
|
||||
limit_size,
|
||||
)
|
||||
|
||||
mask = make_source_mask(x_len)
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||||
mask = make_source_mask(x_len).to(x.device)
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||||
|
||||
if self.unified_model_training:
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chunk_size = self.default_chunk_size + torch.randint(-self.jitter_range, self.jitter_range+1, (1,)).item()
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||||
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||||
@ -355,18 +355,9 @@ class SANMEncoder(AbsEncoder):
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def _add_overlap_chunk(self, feats: np.ndarray, cache: dict = {}):
|
||||
if len(cache) == 0:
|
||||
return feats
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||||
# process last chunk
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||||
cache["feats"] = to_device(cache["feats"], device=feats.device)
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||||
overlap_feats = torch.cat((cache["feats"], feats), dim=1)
|
||||
if cache["is_final"]:
|
||||
cache["feats"] = overlap_feats[:, -cache["chunk_size"][0]:, :]
|
||||
if not cache["last_chunk"]:
|
||||
padding_length = sum(cache["chunk_size"]) - overlap_feats.shape[1]
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||||
overlap_feats = overlap_feats.transpose(1, 2)
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overlap_feats = F.pad(overlap_feats, (0, padding_length))
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||||
overlap_feats = overlap_feats.transpose(1, 2)
|
||||
else:
|
||||
cache["feats"] = overlap_feats[:, -(cache["chunk_size"][0] + cache["chunk_size"][2]):, :]
|
||||
cache["feats"] = overlap_feats[:, -(cache["chunk_size"][0] + cache["chunk_size"][2]):, :]
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||||
return overlap_feats
|
||||
|
||||
def forward_chunk(self,
|
||||
|
||||
@ -221,13 +221,14 @@ class CifPredictorV2(nn.Module):
|
||||
|
||||
if cache is not None and "chunk_size" in cache:
|
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alphas[:, :cache["chunk_size"][0]] = 0.0
|
||||
alphas[:, sum(cache["chunk_size"][:2]):] = 0.0
|
||||
if "is_final" in cache and not cache["is_final"]:
|
||||
alphas[:, sum(cache["chunk_size"][:2]):] = 0.0
|
||||
if cache is not None and "cif_alphas" in cache and "cif_hidden" in cache:
|
||||
cache["cif_hidden"] = to_device(cache["cif_hidden"], device=hidden.device)
|
||||
cache["cif_alphas"] = to_device(cache["cif_alphas"], device=alphas.device)
|
||||
hidden = torch.cat((cache["cif_hidden"], hidden), dim=1)
|
||||
alphas = torch.cat((cache["cif_alphas"], alphas), dim=1)
|
||||
if cache is not None and "last_chunk" in cache and cache["last_chunk"]:
|
||||
if cache is not None and "is_final" in cache and cache["is_final"]:
|
||||
tail_hidden = torch.zeros((batch_size, 1, hidden_size), device=hidden.device)
|
||||
tail_alphas = torch.tensor([[self.tail_threshold]], device=alphas.device)
|
||||
tail_alphas = torch.tile(tail_alphas, (batch_size, 1))
|
||||
|
||||
@ -9,70 +9,70 @@ pyOpenSSL
|
||||
```
|
||||
|
||||
### javascript
|
||||
[html5录音](https://github.com/xiangyuecn/Recorder)
|
||||
[html5 recorder.js](https://github.com/xiangyuecn/Recorder)
|
||||
```shell
|
||||
Recorder
|
||||
```
|
||||
|
||||
### demo页面如下
|
||||

|
||||
### demo
|
||||

|
||||
|
||||
## 两种ws_server_online连接模式
|
||||
### 1)直接连接模式,浏览器https麦克风 --> html5 demo服务 --> js wss接口 --> wss asr online srv(证书生成请往后看)
|
||||
## wss or ws protocol for ws_server_online
|
||||
1) wss: browser microphone data --> html5 demo server --> js wss api --> wss asr online srv #for certificate generation just look back
|
||||
|
||||
### 2)nginx中转,浏览器https麦克风 --> html5 demo服务 --> js wss接口 --> nginx服务 --> ws asr online srv
|
||||
2) ws: browser microphone data --> html5 demo server --> js wss api --> nginx wss server --> ws asr online srv
|
||||
|
||||
## 1.html5 demo服务启动
|
||||
### 启动html5服务,需要ssl证书(自己生成请往后看)
|
||||
## 1.html5 demo start
|
||||
### ssl certificate is required
|
||||
|
||||
```shell
|
||||
usage: h5Server.py [-h] [--host HOST] [--port PORT] [--certfile CERTFILE]
|
||||
[--keyfile KEYFILE]
|
||||
python h5Server.py --port 1337
|
||||
```
|
||||
## 2.启动ws or wss asr online srv
|
||||
[具体请看online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket)
|
||||
online asr提供两种ws和wss模式,wss模式可以直接启动,无需nginx中转。否则需要通过nginx将wss转发到该online asr的ws端口上
|
||||
### wss方式
|
||||
## 2.asr online srv start
|
||||
[detail for online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket)
|
||||
Online asr provides wss or ws way. if started in ws way, nginx is required for relay.
|
||||
### wss way, ssl certificate is required
|
||||
```shell
|
||||
python ws_server_online.py --certfile server.crt --keyfile server.key --port 5921
|
||||
```
|
||||
### ws方式
|
||||
### ws way
|
||||
```shell
|
||||
python ws_server_online.py --port 5921
|
||||
```
|
||||
## 3.修改wsconnecter.js里asr接口地址
|
||||
wsconnecter.js里配置online asr服务地址路径,这里配置的是wss端口
|
||||
## 3.modify asr address in wsconnecter.js according to your environment
|
||||
asr address in wsconnecter.js must be wss, just like
|
||||
var Uri = "wss://xxx:xxx/"
|
||||
|
||||
## 4.浏览器打开地址测试
|
||||
https://127.0.0.1:1337/static/index.html
|
||||
## 4.open browser to access html5 demo
|
||||
https://youraddress:port/static/index.html
|
||||
|
||||
|
||||
|
||||
|
||||
## 自行生成证书
|
||||
生成证书(注意这种证书并不能被所有浏览器认可,部分手动授权可以访问,最好使用其他认证的官方ssl证书)
|
||||
## certificate generation by yourself
|
||||
generated certificate may not suitable for all browsers due to security concerns. you'd better buy or download an authenticated ssl certificate from authorized agency.
|
||||
|
||||
```shell
|
||||
### 1)生成私钥,按照提示填写内容
|
||||
### 1) Generate a private key
|
||||
openssl genrsa -des3 -out server.key 1024
|
||||
|
||||
### 2)生成csr文件 ,按照提示填写内容
|
||||
### 2) Generate a csr file
|
||||
openssl req -new -key server.key -out server.csr
|
||||
|
||||
### 去掉pass
|
||||
### 3) Remove pass
|
||||
cp server.key server.key.org
|
||||
openssl rsa -in server.key.org -out server.key
|
||||
|
||||
### 生成crt文件,有效期1年(365天)
|
||||
### 4) Generated a crt file, valid for 1 year
|
||||
openssl x509 -req -days 365 -in server.csr -signkey server.key -out server.crt
|
||||
```
|
||||
|
||||
## nginx配置说明(了解的可以跳过)
|
||||
h5打开麦克风需要https协议,同时后端的asr websocket也必须是wss协议,如果[online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket)以ws方式运行,我们可以通过nginx配置实现wss协议到ws协议的转换。
|
||||
|
||||
### nginx转发配置示例
|
||||
## nginx configuration (you can skip it if you known)
|
||||
https and wss protocol are required by browsers when want to open microphone and websocket.
|
||||
if [online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket) run in ws way, you should use nginx to convert wss to ws.
|
||||
### nginx wss->ws configuration example
|
||||
```shell
|
||||
events { [0/1548]
|
||||
worker_connections 1024;
|
||||
|
||||
111
funasr/runtime/html5/readme_cn.md
Normal file
111
funasr/runtime/html5/readme_cn.md
Normal file
@ -0,0 +1,111 @@
|
||||
# online asr demo for html5
|
||||
|
||||
## requirement
|
||||
### python
|
||||
```shell
|
||||
flask
|
||||
gevent
|
||||
pyOpenSSL
|
||||
```
|
||||
|
||||
### javascript
|
||||
[html5录音](https://github.com/xiangyuecn/Recorder)
|
||||
```shell
|
||||
Recorder
|
||||
```
|
||||
|
||||
### demo页面如下
|
||||

|
||||
|
||||
## 两种ws_server_online连接模式
|
||||
### 1)直接连接模式,浏览器https麦克风 --> html5 demo服务 --> js wss接口 --> wss asr online srv(证书生成请往后看)
|
||||
|
||||
### 2)nginx中转,浏览器https麦克风 --> html5 demo服务 --> js wss接口 --> nginx服务 --> ws asr online srv
|
||||
|
||||
## 1.html5 demo服务启动
|
||||
### 启动html5服务,需要ssl证书(自己生成请往后看)
|
||||
|
||||
```shell
|
||||
usage: h5Server.py [-h] [--host HOST] [--port PORT] [--certfile CERTFILE]
|
||||
[--keyfile KEYFILE]
|
||||
python h5Server.py --port 1337
|
||||
```
|
||||
## 2.启动ws or wss asr online srv
|
||||
[具体请看online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket)
|
||||
online asr提供两种ws和wss模式,wss模式可以直接启动,无需nginx中转。否则需要通过nginx将wss转发到该online asr的ws端口上
|
||||
### wss方式
|
||||
```shell
|
||||
python ws_server_online.py --certfile server.crt --keyfile server.key --port 5921
|
||||
```
|
||||
### ws方式
|
||||
```shell
|
||||
python ws_server_online.py --port 5921
|
||||
```
|
||||
## 3.修改wsconnecter.js里asr接口地址
|
||||
wsconnecter.js里配置online asr服务地址路径,这里配置的是wss端口
|
||||
var Uri = "wss://xxx:xxx/"
|
||||
|
||||
## 4.浏览器打开地址测试
|
||||
https://127.0.0.1:1337/static/index.html
|
||||
|
||||
|
||||
|
||||
|
||||
## 自行生成证书
|
||||
生成证书(注意这种证书并不能被所有浏览器认可,部分手动授权可以访问,最好使用其他认证的官方ssl证书)
|
||||
|
||||
```shell
|
||||
### 1)生成私钥,按照提示填写内容
|
||||
openssl genrsa -des3 -out server.key 1024
|
||||
|
||||
### 2)生成csr文件 ,按照提示填写内容
|
||||
openssl req -new -key server.key -out server.csr
|
||||
|
||||
### 去掉pass
|
||||
cp server.key server.key.org
|
||||
openssl rsa -in server.key.org -out server.key
|
||||
|
||||
### 生成crt文件,有效期1年(365天)
|
||||
openssl x509 -req -days 365 -in server.csr -signkey server.key -out server.crt
|
||||
```
|
||||
|
||||
## nginx配置说明(了解的可以跳过)
|
||||
h5打开麦克风需要https协议,同时后端的asr websocket也必须是wss协议,如果[online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket)以ws方式运行,我们可以通过nginx配置实现wss协议到ws协议的转换。
|
||||
|
||||
### nginx转发配置示例
|
||||
```shell
|
||||
events { [0/1548]
|
||||
worker_connections 1024;
|
||||
accept_mutex on;
|
||||
}
|
||||
http {
|
||||
error_log error.log;
|
||||
access_log access.log;
|
||||
server {
|
||||
|
||||
listen 5921 ssl http2; # nginx listen port for wss
|
||||
server_name www.test.com;
|
||||
|
||||
ssl_certificate /funasr/server.crt;
|
||||
ssl_certificate_key /funasr/server.key;
|
||||
ssl_protocols TLSv1 TLSv1.1 TLSv1.2;
|
||||
ssl_ciphers HIGH:!aNULL:!MD5;
|
||||
|
||||
location /wss/ {
|
||||
|
||||
|
||||
proxy_pass http://127.0.0.1:1111/; # asr online model ws address and port
|
||||
proxy_http_version 1.1;
|
||||
proxy_set_header Upgrade $http_upgrade;
|
||||
proxy_set_header Connection "upgrade";
|
||||
proxy_read_timeout 600s;
|
||||
|
||||
}
|
||||
}
|
||||
```
|
||||
### 修改wsconnecter.js里asr接口地址
|
||||
wsconnecter.js里配置online asr服务地址路径,这里配置的是wss端口
|
||||
var Uri = "wss://xxx:xxx/wss/"
|
||||
## Acknowledge
|
||||
1. This project is maintained by [FunASR community](https://github.com/alibaba-damo-academy/FunASR).
|
||||
2. We acknowledge [AiHealthx](http://www.aihealthx.com/) for contributing the html5 demo.
|
||||
@ -5,7 +5,7 @@
|
||||
/* 2021-2023 by zhaoming,mali aihealthx.com */
|
||||
|
||||
function WebSocketConnectMethod( config ) { //定义socket连接方法类
|
||||
var Uri = "wss://111.205.137.58:5821/wss/" //设置wss asr online接口地址 如 wss://X.X.X.X:port/wss/
|
||||
var Uri = "wss://30.220.136.139:5921/" // var Uri = "wss://30.221.177.46:5921/" //设置wss asr online接口地址 如 wss://X.X.X.X:port/wss/
|
||||
var speechSokt;
|
||||
var connKeeperID;
|
||||
|
||||
|
||||
@ -11,15 +11,11 @@ class VadModel {
|
||||
public:
|
||||
virtual ~VadModel(){};
|
||||
virtual void InitVad(const std::string &vad_model, const std::string &vad_cmvn, const std::string &vad_config, int thread_num)=0;
|
||||
virtual std::vector<std::vector<int>> Infer(const std::vector<float> &waves)=0;
|
||||
virtual std::vector<std::vector<int>> Infer(std::vector<float> &waves, bool input_finished=true)=0;
|
||||
virtual void ReadModel(const char* vad_model)=0;
|
||||
virtual void LoadConfigFromYaml(const char* filename)=0;
|
||||
virtual void FbankKaldi(float sample_rate, std::vector<std::vector<float>> &vad_feats,
|
||||
const std::vector<float> &waves)=0;
|
||||
virtual void LfrCmvn(std::vector<std::vector<float>> &vad_feats)=0;
|
||||
virtual void Forward(
|
||||
const std::vector<std::vector<float>> &chunk_feats,
|
||||
std::vector<std::vector<float>> *out_prob)=0;
|
||||
std::vector<float> &waves)=0;
|
||||
virtual void LoadCmvn(const char *filename)=0;
|
||||
virtual void InitCache()=0;
|
||||
};
|
||||
|
||||
@ -127,6 +127,8 @@ For example:
|
||||
### funasr-onnx-offline-rtf
|
||||
```shell
|
||||
./funasr-onnx-offline-rtf --model-dir <string> [--quantize <string>]
|
||||
[--vad-dir <string>] [--vad-quant <string>]
|
||||
[--punc-dir <string>] [--punc-quant <string>]
|
||||
--wav-path <string> --thread-num <int32_t>
|
||||
[--] [--version] [-h]
|
||||
Where:
|
||||
@ -136,6 +138,17 @@ Where:
|
||||
(required) the model path, which contains model.onnx, config.yaml, am.mvn
|
||||
--quantize <string>
|
||||
false (Default), load the model of model.onnx in model_dir. If set true, load the model of model_quant.onnx in model_dir
|
||||
|
||||
--vad-dir <string>
|
||||
the vad model path, which contains model.onnx, vad.yaml, vad.mvn
|
||||
--vad-quant <string>
|
||||
false (Default), load the model of model.onnx in vad_dir. If set true, load the model of model_quant.onnx in vad_dir
|
||||
|
||||
--punc-dir <string>
|
||||
the punc model path, which contains model.onnx, punc.yaml
|
||||
--punc-quant <string>
|
||||
false (Default), load the model of model.onnx in punc_dir. If set true, load the model of model_quant.onnx in punc_dir
|
||||
|
||||
--wav-path <string>
|
||||
(required) the input could be:
|
||||
wav_path, e.g.: asr_example.wav;
|
||||
|
||||
@ -162,17 +162,21 @@ void FsmnVad::Forward(
|
||||
}
|
||||
|
||||
// get 4 caches outputs,each size is 128*19
|
||||
for (int i = 1; i < 5; i++) {
|
||||
float* data = vad_ort_outputs[i].GetTensorMutableData<float>();
|
||||
memcpy(in_cache_[i-1].data(), data, sizeof(float) * 128*19);
|
||||
}
|
||||
// for (int i = 1; i < 5; i++) {
|
||||
// float* data = vad_ort_outputs[i].GetTensorMutableData<float>();
|
||||
// memcpy(in_cache_[i-1].data(), data, sizeof(float) * 128*19);
|
||||
// }
|
||||
}
|
||||
|
||||
void FsmnVad::FbankKaldi(float sample_rate, std::vector<std::vector<float>> &vad_feats,
|
||||
const std::vector<float> &waves) {
|
||||
std::vector<float> &waves) {
|
||||
knf::OnlineFbank fbank(fbank_opts);
|
||||
|
||||
fbank.AcceptWaveform(sample_rate, &waves[0], waves.size());
|
||||
std::vector<float> buf(waves.size());
|
||||
for (int32_t i = 0; i != waves.size(); ++i) {
|
||||
buf[i] = waves[i] * 32768;
|
||||
}
|
||||
fbank.AcceptWaveform(sample_rate, buf.data(), buf.size());
|
||||
int32_t frames = fbank.NumFramesReady();
|
||||
for (int32_t i = 0; i != frames; ++i) {
|
||||
const float *frame = fbank.GetFrame(i);
|
||||
@ -267,7 +271,7 @@ void FsmnVad::LfrCmvn(std::vector<std::vector<float>> &vad_feats) {
|
||||
}
|
||||
|
||||
std::vector<std::vector<int>>
|
||||
FsmnVad::Infer(const std::vector<float> &waves) {
|
||||
FsmnVad::Infer(std::vector<float> &waves, bool input_finished) {
|
||||
std::vector<std::vector<float>> vad_feats;
|
||||
std::vector<std::vector<float>> vad_probs;
|
||||
FbankKaldi(vad_sample_rate_, vad_feats, waves);
|
||||
|
||||
@ -21,7 +21,7 @@ public:
|
||||
~FsmnVad();
|
||||
void Test();
|
||||
void InitVad(const std::string &vad_model, const std::string &vad_cmvn, const std::string &vad_config, int thread_num);
|
||||
std::vector<std::vector<int>> Infer(const std::vector<float> &waves);
|
||||
std::vector<std::vector<int>> Infer(std::vector<float> &waves, bool input_finished=true);
|
||||
void Reset();
|
||||
|
||||
private:
|
||||
@ -34,7 +34,7 @@ private:
|
||||
std::vector<const char *> *in_names, std::vector<const char *> *out_names);
|
||||
|
||||
void FbankKaldi(float sample_rate, std::vector<std::vector<float>> &vad_feats,
|
||||
const std::vector<float> &waves);
|
||||
std::vector<float> &waves);
|
||||
|
||||
void LfrCmvn(std::vector<std::vector<float>> &vad_feats);
|
||||
|
||||
|
||||
@ -39,7 +39,7 @@ void runReg(FUNASR_HANDLE asr_handle, vector<string> wav_list,
|
||||
// warm up
|
||||
for (size_t i = 0; i < 1; i++)
|
||||
{
|
||||
FUNASR_RESULT result=FunASRInfer(asr_handle, wav_list[0].c_str(), RASR_NONE, NULL, 16000);
|
||||
FUNASR_RESULT result=FunOfflineInfer(asr_handle, wav_list[0].c_str(), RASR_NONE, NULL, 16000);
|
||||
}
|
||||
|
||||
while (true) {
|
||||
@ -50,7 +50,7 @@ void runReg(FUNASR_HANDLE asr_handle, vector<string> wav_list,
|
||||
}
|
||||
|
||||
gettimeofday(&start, NULL);
|
||||
FUNASR_RESULT result=FunASRInfer(asr_handle, wav_list[i].c_str(), RASR_NONE, NULL, 16000);
|
||||
FUNASR_RESULT result=FunOfflineInfer(asr_handle, wav_list[i].c_str(), RASR_NONE, NULL, 16000);
|
||||
|
||||
gettimeofday(&end, NULL);
|
||||
seconds = (end.tv_sec - start.tv_sec);
|
||||
@ -102,12 +102,20 @@ int main(int argc, char *argv[])
|
||||
TCLAP::CmdLine cmd("funasr-onnx-offline-rtf", ' ', "1.0");
|
||||
TCLAP::ValueArg<std::string> model_dir("", MODEL_DIR, "the model path, which contains model.onnx, config.yaml, am.mvn", true, "", "string");
|
||||
TCLAP::ValueArg<std::string> quantize("", QUANTIZE, "false (Default), load the model of model.onnx in model_dir. If set true, load the model of model_quant.onnx in model_dir", false, "false", "string");
|
||||
TCLAP::ValueArg<std::string> vad_dir("", VAD_DIR, "the vad model path, which contains model.onnx, vad.yaml, vad.mvn", false, "", "string");
|
||||
TCLAP::ValueArg<std::string> vad_quant("", VAD_QUANT, "false (Default), load the model of model.onnx in vad_dir. If set true, load the model of model_quant.onnx in vad_dir", false, "false", "string");
|
||||
TCLAP::ValueArg<std::string> punc_dir("", PUNC_DIR, "the punc model path, which contains model.onnx, punc.yaml", false, "", "string");
|
||||
TCLAP::ValueArg<std::string> punc_quant("", PUNC_QUANT, "false (Default), load the model of model.onnx in punc_dir. If set true, load the model of model_quant.onnx in punc_dir", false, "false", "string");
|
||||
|
||||
TCLAP::ValueArg<std::string> wav_path("", WAV_PATH, "the input could be: wav_path, e.g.: asr_example.wav; pcm_path, e.g.: asr_example.pcm; wav.scp, kaldi style wav list (wav_id \t wav_path)", true, "", "string");
|
||||
TCLAP::ValueArg<std::int32_t> thread_num("", THREAD_NUM, "multi-thread num for rtf", true, 0, "int32_t");
|
||||
|
||||
cmd.add(model_dir);
|
||||
cmd.add(quantize);
|
||||
cmd.add(vad_dir);
|
||||
cmd.add(vad_quant);
|
||||
cmd.add(punc_dir);
|
||||
cmd.add(punc_quant);
|
||||
cmd.add(wav_path);
|
||||
cmd.add(thread_num);
|
||||
cmd.parse(argc, argv);
|
||||
@ -115,11 +123,15 @@ int main(int argc, char *argv[])
|
||||
std::map<std::string, std::string> model_path;
|
||||
GetValue(model_dir, MODEL_DIR, model_path);
|
||||
GetValue(quantize, QUANTIZE, model_path);
|
||||
GetValue(vad_dir, VAD_DIR, model_path);
|
||||
GetValue(vad_quant, VAD_QUANT, model_path);
|
||||
GetValue(punc_dir, PUNC_DIR, model_path);
|
||||
GetValue(punc_quant, PUNC_QUANT, model_path);
|
||||
GetValue(wav_path, WAV_PATH, model_path);
|
||||
|
||||
struct timeval start, end;
|
||||
gettimeofday(&start, NULL);
|
||||
FUNASR_HANDLE asr_handle=FunASRInit(model_path, 1);
|
||||
FUNASR_HANDLE asr_handle=FunOfflineInit(model_path, 1);
|
||||
|
||||
if (!asr_handle)
|
||||
{
|
||||
@ -132,7 +144,7 @@ int main(int argc, char *argv[])
|
||||
long modle_init_micros = ((seconds * 1000000) + end.tv_usec) - (start.tv_usec);
|
||||
LOG(INFO) << "Model initialization takes " << (double)modle_init_micros / 1000000 << " s";
|
||||
|
||||
// read wav_scp
|
||||
// read wav_path
|
||||
vector<string> wav_list;
|
||||
string wav_path_ = model_path.at(WAV_PATH);
|
||||
if(is_target_file(wav_path_, "wav") || is_target_file(wav_path_, "pcm")){
|
||||
@ -179,6 +191,6 @@ int main(int argc, char *argv[])
|
||||
LOG(INFO) << "total_rtf " << (double)total_time/ (total_length*1000000);
|
||||
LOG(INFO) << "speedup " << 1.0/((double)total_time/ (total_length*1000000));
|
||||
|
||||
FunASRUninit(asr_handle);
|
||||
FunOfflineUninit(asr_handle);
|
||||
return 0;
|
||||
}
|
||||
|
||||
@ -69,7 +69,11 @@ void Paraformer::Reset()
|
||||
|
||||
vector<float> Paraformer::FbankKaldi(float sample_rate, const float* waves, int len) {
|
||||
knf::OnlineFbank fbank_(fbank_opts);
|
||||
fbank_.AcceptWaveform(sample_rate, waves, len);
|
||||
std::vector<float> buf(len);
|
||||
for (int32_t i = 0; i != len; ++i) {
|
||||
buf[i] = waves[i] * 32768;
|
||||
}
|
||||
fbank_.AcceptWaveform(sample_rate, buf.data(), buf.size());
|
||||
//fbank_->InputFinished();
|
||||
int32_t frames = fbank_.NumFramesReady();
|
||||
int32_t feature_dim = fbank_opts.mel_opts.num_bins;
|
||||
|
||||
@ -186,11 +186,12 @@ class CT_Transformer_VadRealtime(CT_Transformer):
|
||||
mini_sentence = cache_sent + mini_sentence
|
||||
mini_sentence_id = np.concatenate((cache_sent_id, mini_sentence_id), axis=0,dtype='int32')
|
||||
text_length = len(mini_sentence_id)
|
||||
vad_mask = self.vad_mask(text_length, len(cache))[None, None, :, :].astype(np.float32)
|
||||
data = {
|
||||
"input": mini_sentence_id[None,:],
|
||||
"text_lengths": np.array([text_length], dtype='int32'),
|
||||
"vad_mask": self.vad_mask(text_length, len(cache))[None, None, :, :].astype(np.float32),
|
||||
"sub_masks": np.tril(np.ones((text_length, text_length), dtype=np.float32))[None, None, :, :].astype(np.float32)
|
||||
"vad_mask": vad_mask,
|
||||
"sub_masks": vad_mask
|
||||
}
|
||||
try:
|
||||
outputs = self.infer(data['input'], data['text_lengths'], data['vad_mask'], data["sub_masks"])
|
||||
|
||||
@ -32,15 +32,29 @@ inference_pipeline_asr_online = pipeline(
|
||||
ncpu=args.ncpu,
|
||||
model_revision='v1.0.4')
|
||||
|
||||
# vad
|
||||
inference_pipeline_vad = pipeline(
|
||||
task=Tasks.voice_activity_detection,
|
||||
model=args.vad_model,
|
||||
model_revision=None,
|
||||
output_dir=None,
|
||||
batch_size=1,
|
||||
mode='online',
|
||||
ngpu=args.ngpu,
|
||||
ncpu=1,
|
||||
)
|
||||
|
||||
print("model loaded")
|
||||
|
||||
|
||||
|
||||
async def ws_serve(websocket, path):
|
||||
frames = []
|
||||
frames_asr_online = []
|
||||
global websocket_users
|
||||
websocket_users.add(websocket)
|
||||
websocket.param_dict_asr_online = {"cache": dict()}
|
||||
websocket.param_dict_vad = {'in_cache': dict()}
|
||||
websocket.wav_name = "microphone"
|
||||
print("new user connected",flush=True)
|
||||
try:
|
||||
@ -53,9 +67,10 @@ async def ws_serve(websocket, path):
|
||||
if "is_speaking" in messagejson:
|
||||
websocket.is_speaking = messagejson["is_speaking"]
|
||||
websocket.param_dict_asr_online["is_final"] = not websocket.is_speaking
|
||||
websocket.param_dict_vad["is_final"] = not websocket.is_speaking
|
||||
# need to fire engine manually if no data received any more
|
||||
if not websocket.is_speaking:
|
||||
await async_asr_online(websocket,b"")
|
||||
await async_asr_online(websocket, b"")
|
||||
if "chunk_interval" in messagejson:
|
||||
websocket.chunk_interval=messagejson["chunk_interval"]
|
||||
if "wav_name" in messagejson:
|
||||
@ -64,14 +79,18 @@ async def ws_serve(websocket, path):
|
||||
websocket.param_dict_asr_online["chunk_size"] = messagejson["chunk_size"]
|
||||
# if has bytes in buffer or message is bytes
|
||||
if len(frames_asr_online) > 0 or not isinstance(message, str):
|
||||
if not isinstance(message,str):
|
||||
if not isinstance(message, str):
|
||||
frames_asr_online.append(message)
|
||||
# frames.append(message)
|
||||
# duration_ms = len(message) // 32
|
||||
# websocket.vad_pre_idx += duration_ms
|
||||
speech_start_i, speech_end_i = await async_vad(websocket, message)
|
||||
websocket.is_speaking = not speech_end_i
|
||||
|
||||
if len(frames_asr_online) % websocket.chunk_interval == 0 or not websocket.is_speaking:
|
||||
websocket.param_dict_asr_online["is_final"] = not websocket.is_speaking
|
||||
audio_in = b"".join(frames_asr_online)
|
||||
# if not websocket.is_speaking:
|
||||
#padding 0.5s at end gurantee that asr engine can fire out last word
|
||||
# audio_in=audio_in+b''.join(np.zeros(int(16000*0.5),dtype=np.int16))
|
||||
await async_asr_online(websocket,audio_in)
|
||||
await async_asr_online(websocket, audio_in)
|
||||
frames_asr_online = []
|
||||
|
||||
|
||||
@ -85,7 +104,7 @@ async def ws_serve(websocket, path):
|
||||
|
||||
|
||||
async def async_asr_online(websocket,audio_in):
|
||||
if len(audio_in) >=0:
|
||||
if len(audio_in) >= 0:
|
||||
audio_in = load_bytes(audio_in)
|
||||
rec_result = inference_pipeline_asr_online(audio_in=audio_in,
|
||||
param_dict=websocket.param_dict_asr_online)
|
||||
@ -97,16 +116,30 @@ async def async_asr_online(websocket,audio_in):
|
||||
await websocket.send(message)
|
||||
|
||||
|
||||
async def async_vad(websocket, audio_in):
|
||||
segments_result = inference_pipeline_vad(audio_in=audio_in, param_dict=websocket.param_dict_vad)
|
||||
|
||||
speech_start = False
|
||||
speech_end = False
|
||||
|
||||
if len(segments_result) == 0 or len(segments_result["text"]) > 1:
|
||||
return speech_start, speech_end
|
||||
if segments_result["text"][0][0] != -1:
|
||||
speech_start = segments_result["text"][0][0]
|
||||
if segments_result["text"][0][1] != -1:
|
||||
speech_end = True
|
||||
return speech_start, speech_end
|
||||
|
||||
if len(args.certfile)>0:
|
||||
ssl_context = ssl.SSLContext(ssl.PROTOCOL_TLS_SERVER)
|
||||
|
||||
# Generate with Lets Encrypt, copied to this location, chown to current user and 400 permissions
|
||||
ssl_cert = args.certfile
|
||||
ssl_key = args.keyfile
|
||||
|
||||
ssl_context.load_cert_chain(ssl_cert, keyfile=ssl_key)
|
||||
start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None,ssl=ssl_context)
|
||||
ssl_context = ssl.SSLContext(ssl.PROTOCOL_TLS_SERVER)
|
||||
|
||||
# Generate with Lets Encrypt, copied to this location, chown to current user and 400 permissions
|
||||
ssl_cert = args.certfile
|
||||
ssl_key = args.keyfile
|
||||
|
||||
ssl_context.load_cert_chain(ssl_cert, keyfile=ssl_key)
|
||||
start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None,ssl=ssl_context)
|
||||
else:
|
||||
start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None)
|
||||
start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None)
|
||||
asyncio.get_event_loop().run_until_complete(start_server)
|
||||
asyncio.get_event_loop().run_forever()
|
||||
@ -290,6 +290,8 @@ class ASRTask(AbsTask):
|
||||
predictor_choices2,
|
||||
# --stride_conv and --stride_conv_conf
|
||||
stride_conv_choices,
|
||||
# --rnnt_decoder and --rnnt_decoder_conf
|
||||
rnnt_decoder_choices,
|
||||
]
|
||||
|
||||
# If you need to modify train() or eval() procedures, change Trainer class here
|
||||
@ -1360,7 +1362,7 @@ class ASRTaskAligner(ASRTaskParaformer):
|
||||
return retval
|
||||
|
||||
|
||||
class ASRTransducerTask(AbsTask):
|
||||
class ASRTransducerTask(ASRTask):
|
||||
"""ASR Transducer Task definition."""
|
||||
|
||||
num_optimizers: int = 1
|
||||
@ -1371,244 +1373,11 @@ class ASRTransducerTask(AbsTask):
|
||||
normalize_choices,
|
||||
encoder_choices,
|
||||
rnnt_decoder_choices,
|
||||
joint_network_choices,
|
||||
]
|
||||
|
||||
trainer = Trainer
|
||||
|
||||
@classmethod
|
||||
def add_task_arguments(cls, parser: argparse.ArgumentParser):
|
||||
"""Add Transducer task arguments.
|
||||
Args:
|
||||
cls: ASRTransducerTask object.
|
||||
parser: Transducer arguments parser.
|
||||
"""
|
||||
group = parser.add_argument_group(description="Task related.")
|
||||
|
||||
# required = parser.get_default("required")
|
||||
# required += ["token_list"]
|
||||
|
||||
group.add_argument(
|
||||
"--token_list",
|
||||
type=str_or_none,
|
||||
default=None,
|
||||
help="Integer-string mapper for tokens.",
|
||||
)
|
||||
group.add_argument(
|
||||
"--split_with_space",
|
||||
type=str2bool,
|
||||
default=True,
|
||||
help="whether to split text using <space>",
|
||||
)
|
||||
group.add_argument(
|
||||
"--input_size",
|
||||
type=int_or_none,
|
||||
default=None,
|
||||
help="The number of dimensions for input features.",
|
||||
)
|
||||
group.add_argument(
|
||||
"--init",
|
||||
type=str_or_none,
|
||||
default=None,
|
||||
help="Type of model initialization to use.",
|
||||
)
|
||||
group.add_argument(
|
||||
"--model_conf",
|
||||
action=NestedDictAction,
|
||||
default=get_default_kwargs(TransducerModel),
|
||||
help="The keyword arguments for the model class.",
|
||||
)
|
||||
# group.add_argument(
|
||||
# "--encoder_conf",
|
||||
# action=NestedDictAction,
|
||||
# default={},
|
||||
# help="The keyword arguments for the encoder class.",
|
||||
# )
|
||||
group.add_argument(
|
||||
"--joint_network_conf",
|
||||
action=NestedDictAction,
|
||||
default={},
|
||||
help="The keyword arguments for the joint network class.",
|
||||
)
|
||||
group = parser.add_argument_group(description="Preprocess related.")
|
||||
group.add_argument(
|
||||
"--use_preprocessor",
|
||||
type=str2bool,
|
||||
default=True,
|
||||
help="Whether to apply preprocessing to input data.",
|
||||
)
|
||||
group.add_argument(
|
||||
"--token_type",
|
||||
type=str,
|
||||
default="bpe",
|
||||
choices=["bpe", "char", "word", "phn"],
|
||||
help="The type of tokens to use during tokenization.",
|
||||
)
|
||||
group.add_argument(
|
||||
"--bpemodel",
|
||||
type=str_or_none,
|
||||
default=None,
|
||||
help="The path of the sentencepiece model.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--non_linguistic_symbols",
|
||||
type=str_or_none,
|
||||
help="The 'non_linguistic_symbols' file path.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--cleaner",
|
||||
type=str_or_none,
|
||||
choices=[None, "tacotron", "jaconv", "vietnamese"],
|
||||
default=None,
|
||||
help="Text cleaner to use.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--g2p",
|
||||
type=str_or_none,
|
||||
choices=g2p_choices,
|
||||
default=None,
|
||||
help="g2p method to use if --token_type=phn.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--speech_volume_normalize",
|
||||
type=float_or_none,
|
||||
default=None,
|
||||
help="Normalization value for maximum amplitude scaling.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--rir_scp",
|
||||
type=str_or_none,
|
||||
default=None,
|
||||
help="The RIR SCP file path.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--rir_apply_prob",
|
||||
type=float,
|
||||
default=1.0,
|
||||
help="The probability of the applied RIR convolution.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--noise_scp",
|
||||
type=str_or_none,
|
||||
default=None,
|
||||
help="The path of noise SCP file.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--noise_apply_prob",
|
||||
type=float,
|
||||
default=1.0,
|
||||
help="The probability of the applied noise addition.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--noise_db_range",
|
||||
type=str,
|
||||
default="13_15",
|
||||
help="The range of the noise decibel level.",
|
||||
)
|
||||
for class_choices in cls.class_choices_list:
|
||||
# Append --<name> and --<name>_conf.
|
||||
# e.g. --decoder and --decoder_conf
|
||||
class_choices.add_arguments(group)
|
||||
|
||||
@classmethod
|
||||
def build_collate_fn(
|
||||
cls, args: argparse.Namespace, train: bool
|
||||
) -> Callable[
|
||||
[Collection[Tuple[str, Dict[str, np.ndarray]]]],
|
||||
Tuple[List[str], Dict[str, torch.Tensor]],
|
||||
]:
|
||||
"""Build collate function.
|
||||
Args:
|
||||
cls: ASRTransducerTask object.
|
||||
args: Task arguments.
|
||||
train: Training mode.
|
||||
Return:
|
||||
: Callable collate function.
|
||||
"""
|
||||
assert check_argument_types()
|
||||
|
||||
return CommonCollateFn(float_pad_value=0.0, int_pad_value=-1)
|
||||
|
||||
@classmethod
|
||||
def build_preprocess_fn(
|
||||
cls, args: argparse.Namespace, train: bool
|
||||
) -> Optional[Callable[[str, Dict[str, np.array]], Dict[str, np.ndarray]]]:
|
||||
"""Build pre-processing function.
|
||||
Args:
|
||||
cls: ASRTransducerTask object.
|
||||
args: Task arguments.
|
||||
train: Training mode.
|
||||
Return:
|
||||
: Callable pre-processing function.
|
||||
"""
|
||||
assert check_argument_types()
|
||||
|
||||
if args.use_preprocessor:
|
||||
retval = CommonPreprocessor(
|
||||
train=train,
|
||||
token_type=args.token_type,
|
||||
token_list=args.token_list,
|
||||
bpemodel=args.bpemodel,
|
||||
non_linguistic_symbols=args.non_linguistic_symbols,
|
||||
text_cleaner=args.cleaner,
|
||||
g2p_type=args.g2p,
|
||||
split_with_space=args.split_with_space if hasattr(args, "split_with_space") else False,
|
||||
rir_scp=args.rir_scp if hasattr(args, "rir_scp") else None,
|
||||
rir_apply_prob=args.rir_apply_prob
|
||||
if hasattr(args, "rir_apply_prob")
|
||||
else 1.0,
|
||||
noise_scp=args.noise_scp if hasattr(args, "noise_scp") else None,
|
||||
noise_apply_prob=args.noise_apply_prob
|
||||
if hasattr(args, "noise_apply_prob")
|
||||
else 1.0,
|
||||
noise_db_range=args.noise_db_range
|
||||
if hasattr(args, "noise_db_range")
|
||||
else "13_15",
|
||||
speech_volume_normalize=args.speech_volume_normalize
|
||||
if hasattr(args, "rir_scp")
|
||||
else None,
|
||||
)
|
||||
else:
|
||||
retval = None
|
||||
|
||||
assert check_return_type(retval)
|
||||
return retval
|
||||
|
||||
@classmethod
|
||||
def required_data_names(
|
||||
cls, train: bool = True, inference: bool = False
|
||||
) -> Tuple[str, ...]:
|
||||
"""Required data depending on task mode.
|
||||
Args:
|
||||
cls: ASRTransducerTask object.
|
||||
train: Training mode.
|
||||
inference: Inference mode.
|
||||
Return:
|
||||
retval: Required task data.
|
||||
"""
|
||||
if not inference:
|
||||
retval = ("speech", "text")
|
||||
else:
|
||||
retval = ("speech",)
|
||||
|
||||
return retval
|
||||
|
||||
@classmethod
|
||||
def optional_data_names(
|
||||
cls, train: bool = True, inference: bool = False
|
||||
) -> Tuple[str, ...]:
|
||||
"""Optional data depending on task mode.
|
||||
Args:
|
||||
cls: ASRTransducerTask object.
|
||||
train: Training mode.
|
||||
inference: Inference mode.
|
||||
Return:
|
||||
retval: Optional task data.
|
||||
"""
|
||||
retval = ()
|
||||
assert check_return_type(retval)
|
||||
|
||||
return retval
|
||||
|
||||
@classmethod
|
||||
def build_model(cls, args: argparse.Namespace) -> TransducerModel:
|
||||
"""Required data depending on task mode.
|
||||
|
||||
Loading…
Reference in New Issue
Block a user