code format, test=doc

3 years ago · 9d0224460b
parent 4b111146dc
commit 9d0224460b
2 changed files with 42 additions and 37 deletions
--- a/paddlespeech/server/tests/tts/infer/run.sh
+++ b/paddlespeech/server/tests/tts/infer/run.sh
@ -1,6 +1,6 @@
 model_path=~/.paddlespeech/models/
-am_model_dir=$model_path/fastspeech2_csmsc-zh/fastspeech2_cnndecoder_csmsc_ckpt_1.0.0/    ## fastspeech2_c
-voc_model_dir=$model_path/mb_melgan_csmsc-zh/mb_melgan_csmsc_ckpt_0.1.1/    ## mb_melgan
+am_model_dir=$model_path/fastspeech2_csmsc-zh/fastspeech2_cnndecoder_csmsc_ckpt_1.0.0/   
+voc_model_dir=$model_path/mb_melgan_csmsc-zh/mb_melgan_csmsc_ckpt_0.1.1/    
 testdata=../../../../t2s/exps/csmsc_test.txt

 # get am file
@ -33,9 +33,13 @@ done


 # run test
-# am can choose fastspeech2_csmsc or fastspeech2-C_csmsc, where fastspeech2-C_csmsc supports streaming inference.
+# am can choose fastspeech2_csmsc or fastspeech2_cnndecoder_csmsc, where fastspeech2_cnndecoder_csmsc supports streaming inference.
 # voc can choose hifigan_csmsc and mb_melgan_csmsc, They can both support streaming inference.
-python test_online_tts.py --am fastspeech2-C_csmsc \
+# When am is fastspeech2_cnndecoder_csmsc and am_pad is set to 12, there is no diff between streaming and non-streaming inference results.
+# When voc is mb_melgan_csmsc and voc_pad is set to 14, there is no diff between streaming and non-streaming inference results.
+# When voc is hifigan_csmsc and voc_pad is set to 20, there is no diff between streaming and non-streaming inference results.
+
+python test_online_tts.py --am fastspeech2_cnndecoder_csmsc \
                          --am_config $am_model_dir/$am_config_file \
                          --am_ckpt $am_model_dir/$am_ckpt_file \
                          --am_stat $am_model_dir/$am_stat_file \
--- a/paddlespeech/server/tests/tts/infer/test_online_tts.py
+++ b/paddlespeech/server/tests/tts/infer/test_online_tts.py
@ -34,8 +34,8 @@ from paddlespeech.t2s.utils import str2bool

 mel_streaming = None
 wav_streaming = None
-stream_first_time = 0.0
-voc_stream_st = 0.0
+streaming_first_time = 0.0
+streaming_voc_st = 0.0
 sample_rate = 0


@ -65,7 +65,7 @@ def get_chunks(data, block_size, pad_size, step):
    return chunks


-def get_stream_am_inference(args, am_config):
+def get_streaming_am_inference(args, am_config):
    with open(args.phones_dict, "r") as f:
        phn_id = [line.strip().split() for line in f.readlines()]
    vocab_size = len(phn_id)
@ -99,8 +99,8 @@ def init(args):
    frontend = get_frontend(args)

    # acoustic model
-    if args.am == 'fastspeech2-C_csmsc':
-        am, am_mu, am_std = get_stream_am_inference(args, am_config)
+    if args.am == 'fastspeech2_cnndecoder_csmsc':
+        am, am_mu, am_std = get_streaming_am_inference(args, am_config)
        am_infer_info = [am, am_mu, am_std, am_config]
    else:
        am_inference, am_name, am_dataset = get_am_inference(args, am_config)
@ -139,7 +139,7 @@ def get_phone(args, frontend, sentence, merge_sentences, get_tone_ids):
 # 生成完整的mel
 def gen_mel(args, am_infer_info, part_phone_ids, part_tone_ids):
    # 如果是支持流式的AM模型
-    if args.am == 'fastspeech2-C_csmsc':
+    if args.am == 'fastspeech2_cnndecoder_csmsc':
        am, am_mu, am_std, am_config = am_infer_info
        orig_hs, h_masks = am.encoder_infer(part_phone_ids)
        if args.am_streaming:
@ -183,9 +183,9 @@ def gen_mel(args, am_infer_info, part_phone_ids, part_tone_ids):


@paddle.no_grad()
-def stream_voc_infer(args, voc_infer_info, mel_len):
+def streaming_voc_infer(args, voc_infer_info, mel_len):
    global mel_streaming
-    global stream_first_time
+    global streaming_first_time
    global wav_streaming
    voc_inference, voc_config = voc_infer_info
    block = args.voc_block
@ -203,7 +203,7 @@ def stream_voc_infer(args, voc_infer_info, mel_len):
    while valid_end <= mel_len:
        sub_wav = voc_inference(mel_chunk)
        if flag == 1:
-            stream_first_time = time.time()
+            streaming_first_time = time.time()
            flag = 0

        # get valid wav    
@ -233,7 +233,7 @@ def stream_voc_infer(args, voc_infer_info, mel_len):

@paddle.no_grad()
 # 非流式AM / 流式AM + 非流式Voc
-def am_nostream_voc(args, am_infer_info, voc_infer_info, part_phone_ids,
+def am_nonstreaming_voc(args, am_infer_info, voc_infer_info, part_phone_ids,
                        part_tone_ids):
    mel = gen_mel(args, am_infer_info, part_phone_ids, part_tone_ids)
    am_infer_time = time.time()
@ -248,10 +248,10 @@ def am_nostream_voc(args, am_infer_info, voc_infer_info, part_phone_ids,

@paddle.no_grad()
 # 非流式AM + 流式Voc
-def nostream_am_stream_voc(args, am_infer_info, voc_infer_info, part_phone_ids,
-                           part_tone_ids):
+def nonstreaming_am_streaming_voc(args, am_infer_info, voc_infer_info,
+                                  part_phone_ids, part_tone_ids):
    global mel_streaming
-    global stream_first_time
+    global streaming_first_time
    global wav_streaming

    mel = gen_mel(args, am_infer_info, part_phone_ids, part_tone_ids)
@ -260,8 +260,8 @@ def nostream_am_stream_voc(args, am_infer_info, voc_infer_info, part_phone_ids,
    # voc streaming
    mel_streaming = mel
    mel_len = mel.shape[0]
-    stream_voc_infer(args, voc_infer_info, mel_len)
-    first_response_time = stream_first_time
+    streaming_voc_infer(args, voc_infer_info, mel_len)
+    first_response_time = streaming_first_time
    wav = wav_streaming
    final_response_time = time.time()
    voc_infer_time = final_response_time
@ -271,12 +271,12 @@ def nostream_am_stream_voc(args, am_infer_info, voc_infer_info, part_phone_ids,

@paddle.no_grad()
 # 流式AM + 流式 Voc
-def stream_am_stream_voc(args, am_infer_info, voc_infer_info, part_phone_ids,
-                         part_tone_ids):
+def streaming_am_streaming_voc(args, am_infer_info, voc_infer_info,
+                               part_phone_ids, part_tone_ids):
    global mel_streaming
-    global stream_first_time
+    global streaming_first_time
    global wav_streaming
-    global voc_stream_st
+    global streaming_voc_st
    mel_streaming = None
    #用来表示开启流式voc的线程
    flag = 1
@ -311,15 +311,16 @@ def stream_am_stream_voc(args, am_infer_info, voc_infer_info, part_phone_ids,

        if flag and mel_streaming.shape[0] > args.voc_block + args.voc_pad:
            t = threading.Thread(
-                target=stream_voc_infer, args=(args, voc_infer_info, mel_len, ))
+                target=streaming_voc_infer,
+                args=(args, voc_infer_info, mel_len, ))
            t.start()
-            voc_stream_st = time.time()
+            streaming_voc_st = time.time()
            flag = 0

    t.join()
    final_response_time = time.time()
    voc_infer_time = final_response_time
-    first_response_time = stream_first_time
+    first_response_time = streaming_first_time
    wav = wav_streaming

    return am_infer_time, voc_infer_time, first_response_time, final_response_time, wav
@ -337,11 +338,11 @@ def warm_up(args, logger, frontend, am_infer_info, voc_infer_info):

    if args.voc_streaming:
        if args.am_streaming:
-            infer_func = stream_am_stream_voc
+            infer_func = streaming_am_streaming_voc
        else:
-            infer_func = nostream_am_stream_voc
+            infer_func = nonstreaming_am_streaming_voc
    else:
-        infer_func = am_nostream_voc
+        infer_func = am_nonstreaming_voc

    merge_sentences = True
    get_tone_ids = False
@ -376,11 +377,11 @@ def evaluate(args, logger, frontend, am_infer_info, voc_infer_info):
    # choose infer function
    if args.voc_streaming:
        if args.am_streaming:
-            infer_func = stream_am_stream_voc
+            infer_func = streaming_am_streaming_voc
        else:
-            infer_func = nostream_am_stream_voc
+            infer_func = nonstreaming_am_streaming_voc
    else:
-        infer_func = am_nostream_voc
+        infer_func = am_nonstreaming_voc

    final_up_duration = 0.0
    sentence_count = 0
@ -410,7 +411,7 @@ def evaluate(args, logger, frontend, am_infer_info, voc_infer_info):
            args, am_infer_info, voc_infer_info, part_phone_ids, part_tone_ids)
        am_time = am_infer_time - am_st
        if args.voc_streaming and args.am_streaming:
-            voc_time = voc_infer_time - voc_stream_st
+            voc_time = voc_infer_time - streaming_voc_st
        else:
            voc_time = voc_infer_time - am_infer_time

@ -482,8 +483,8 @@ def parse_args():
        '--am',
        type=str,
        default='fastspeech2_csmsc',
-        choices=['fastspeech2_csmsc', 'fastspeech2-C_csmsc'],
-        help='Choose acoustic model type of tts task. where fastspeech2-C_csmsc supports streaming inference'
+        choices=['fastspeech2_csmsc', 'fastspeech2_cnndecoder_csmsc'],
+        help='Choose acoustic model type of tts task. where fastspeech2_cnndecoder_csmsc supports streaming inference'
    )

    parser.add_argument(
@ -576,7 +577,7 @@ def main():
    args = parse_args()
    paddle.set_device(args.device)
    if args.am_streaming:
-        assert (args.am == 'fastspeech2-C_csmsc')
+        assert (args.am == 'fastspeech2_cnndecoder_csmsc')

    logger = logging.getLogger()
    fhandler = logging.FileHandler(filename=args.log_file, mode='w')