Update app.py

app.py

@@ -1,377 +1,377 @@
-import gradio as gr
-import os
-from utils.default_models import ensure_default_models
-import sys
-import traceback
-from pathlib import Path
-from time import perf_counter as timer
-import numpy as np
-import torch
-from encoder import inference as encoder
-from synthesizer.inference import Synthesizer
-#from toolbox.utterance import Utterance
-from vocoder import inference as vocoder
-import time
-import librosa
-import numpy as np
-#import sounddevice as sd
-import soundfile as sf
-import argparse
-from utils.argutils import print_args
-
-parser = argparse.ArgumentParser(
-    formatter_class=argparse.ArgumentDefaultsHelpFormatter
-)
-parser.add_argument("-e", "--enc_model_fpath", type=Path,
-                    default="saved_models/default/encoder.pt",
-                    help="Path to a saved encoder")
-parser.add_argument("-s", "--syn_model_fpath", type=Path,
-                    default="saved_models/default/synthesizer.pt",
-                    help="Path to a saved synthesizer")
-parser.add_argument("-v", "--voc_model_fpath", type=Path,
-                    default="saved_models/default/vocoder.pt",
-                    help="Path to a saved vocoder")
-parser.add_argument("--cpu", action="store_true", help=\
-    "If True, processing is done on CPU, even when a GPU is available.")
-parser.add_argument("--no_sound", action="store_true", help=\
-    "If True, audio won't be played.")
-parser.add_argument("--seed", type=int, default=None, help=\
-    "Optional random number seed value to make toolbox deterministic.")
-args = parser.parse_args()
-arg_dict = vars(args)
-print_args(args, parser)
-
-# Maximum of generated wavs to keep on memory
-MAX_WAVS = 15
-utterances = set()
-current_generated = (None, None, None, None) # speaker_name, spec, breaks, wav
-synthesizer = None # type: Synthesizer
-current_wav = None
-waves_list = []
-waves_count = 0
-waves_namelist = []
-
-# Hide GPUs from Pytorch to force CPU processing
-if arg_dict.pop("cpu"):
-    os.environ["CUDA_VISIBLE_DEVICES"] = "-1"
-
-print("Running a test of your configuration...\n")
-
-if torch.cuda.is_available():
-    device_id = torch.cuda.current_device()
-    gpu_properties = torch.cuda.get_device_properties(device_id)
-    ## Print some environment information (for debugging purposes)
-    print("Found %d GPUs available. Using GPU %d (%s) of compute capability %d.%d with "
-        "%.1fGb total memory.\n" %
-        (torch.cuda.device_count(),
-        device_id,
-        gpu_properties.name,
-        gpu_properties.major,
-        gpu_properties.minor,
-        gpu_properties.total_memory / 1e9))
-else:
-    print("Using CPU for inference.\n")
-
-## Load the models one by one.
-print("Preparing the encoder, the synthesizer and the vocoder...")
-ensure_default_models(Path("saved_models"))
-#encoder.load_model(args.enc_model_fpath)
-#synthesizer = Synthesizer(args.syn_model_fpath)
-#vocoder.load_model(args.voc_model_fpath)
-
-def compute_embedding(in_fpath):
-
-    if not encoder.is_loaded():
-        model_fpath = args.enc_model_fpath
-        print("Loading the encoder %s... " % model_fpath)
-        start = time.time() 
-        encoder.load_model(model_fpath)
-        print("Done (%dms)." % int(1000 * (time.time() - start)), "append")
-
-
-    ## Computing the embedding
-    # First, we load the wav using the function that the speaker encoder provides. This is
-    
-    # Get the wav from the disk. We take the wav with the vocoder/synthesizer format for
-    # playback, so as to have a fair comparison with the generated audio
-    wav = Synthesizer.load_preprocess_wav(in_fpath)
-    
-    # important: there is preprocessing that must be applied.
-
-    # The following two methods are equivalent:
-    # - Directly load from the filepath:
-    preprocessed_wav = encoder.preprocess_wav(wav)
-
-    # - If the wav is already loaded:
-    #original_wav, sampling_rate = librosa.load(str(in_fpath))
-    #preprocessed_wav = encoder.preprocess_wav(original_wav, sampling_rate)
-
-    # Compute the embedding
-    embed, partial_embeds, _ = encoder.embed_utterance(preprocessed_wav, return_partials=True)
-
-
-    print("Loaded file succesfully")
-
-    # Then we derive the embedding. There are many functions and parameters that the
-    # speaker encoder interfaces. These are mostly for in-depth research. You will typically
-    # only use this function (with its default parameters):
-    #embed = encoder.embed_utterance(preprocessed_wav)
-    
-    return embed 
-def create_spectrogram(text,embed):
-        # If seed is specified, reset torch seed and force synthesizer reload
-        if args.seed is not None:
-            torch.manual_seed(args.seed)
-            synthesizer = Synthesizer(args.syn_model_fpath)
-        
-        
-        # Synthesize the spectrogram
-        model_fpath = args.syn_model_fpath
-        print("Loading the synthesizer %s... " % model_fpath)
-        start = time.time()
-        synthesizer = Synthesizer(model_fpath)
-        print("Done (%dms)." % int(1000 * (time.time()- start)), "append")          
-        
-
-        # The synthesizer works in batch, so you need to put your data in a list or numpy array
-        texts = [text]
-        embeds = [embed]
-        # If you know what the attention layer alignments are, you can retrieve them here by
-        # passing return_alignments=True
-        specs = synthesizer.synthesize_spectrograms(texts, embeds)
-        breaks = [spec.shape[1] for spec in specs]
-        spec = np.concatenate(specs, axis=1)
-        sample_rate=synthesizer.sample_rate
-        return spec, breaks , sample_rate
-
-
-def generate_waveform(current_generated):
-
-        speaker_name, spec, breaks = current_generated
-        assert spec is not None
-
-        ## Generating the waveform
-        print("Synthesizing the waveform:")
-        # If seed is specified, reset torch seed and reload vocoder
-        if args.seed is not None:
-            torch.manual_seed(args.seed)
-            vocoder.load_model(args.voc_model_fpath)
-
-        model_fpath = args.voc_model_fpath
-        # Synthesize the waveform
-        if not vocoder.is_loaded():
-            print("Loading the vocoder %s... " % model_fpath)
-            start = time.time()
-            vocoder.load_model(model_fpath)
-            print("Done (%dms)." % int(1000 * (time.time()- start)), "append")    
-
-        current_vocoder_fpath= model_fpath
-        def vocoder_progress(i, seq_len, b_size, gen_rate):
-            real_time_factor = (gen_rate / Synthesizer.sample_rate) * 1000
-            line = "Waveform generation: %d/%d (batch size: %d, rate: %.1fkHz - %.2fx real time)" \
-                % (i * b_size, seq_len * b_size, b_size, gen_rate, real_time_factor)
-            print(line, "overwrite")       
-
-
-        # Synthesizing the waveform is fairly straightforward. Remember that the longer the
-        # spectrogram, the more time-efficient the vocoder.
-        if  current_vocoder_fpath is not None:
-            print("")
-            generated_wav = vocoder.infer_waveform(spec, progress_callback=vocoder_progress)
-        else:
-            print("Waveform generation with Griffin-Lim... ")
-            generated_wav = Synthesizer.griffin_lim(spec)
-
-        print(" Done!", "append")
-
-
-        ## Post-generation
-        # There's a bug with sounddevice that makes the audio cut one second earlier, so we
-        # pad it.
-        generated_wav = np.pad(generated_wav, (0, Synthesizer.sample_rate), mode="constant")
-
-        # Add breaks
-        b_ends = np.cumsum(np.array(breaks) * Synthesizer.hparams.hop_size)
-        b_starts = np.concatenate(([0], b_ends[:-1]))
-        wavs = [generated_wav[start:end] for start, end, in zip(b_starts, b_ends)]
-        breaks = [np.zeros(int(0.15 * Synthesizer.sample_rate))] * len(breaks)
-        generated_wav = np.concatenate([i for w, b in zip(wavs, breaks) for i in (w, b)])
-
-
-        # Trim excess silences to compensate for gaps in spectrograms (issue #53)
-        generated_wav = encoder.preprocess_wav(generated_wav)
-
-
-        return generated_wav
-
-
-def save_on_disk(generated_wav,sample_rate):
-        # Save it on the disk
-        filename = "cloned_voice.wav"
-        print(generated_wav.dtype)
-        #OUT=os.environ['OUT_PATH']
-        # Returns `None` if key doesn't exist
-        #OUT=os.environ.get('OUT_PATH')
-        #result = os.path.join(OUT, filename)
-        result = filename
-        print(" > Saving output to {}".format(result))
-        sf.write(result, generated_wav.astype(np.float32), sample_rate)
-        print("\nSaved output as %s\n\n" % result) 
-      
-        return  result     
-def play_audio(generated_wav,sample_rate):
-        # Play the audio (non-blocking)
-        if not args.no_sound:
-          
-            try:
-                sd.stop()
-                sd.play(generated_wav, sample_rate)
-            except sd.PortAudioError as e:
-                print("\nCaught exception: %s" % repr(e))
-                print("Continuing without audio playback. Suppress this message with the \"--no_sound\" flag.\n")
-            except:
-                raise
-         
-
-def clean_memory():
-    import gc
-    #import GPUtil
-    # To see memory usage
-    print('Before clean ')
-    #GPUtil.showUtilization()
-    #cleaning memory 1
-    gc.collect()
-    torch.cuda.empty_cache()
-    time.sleep(2)
-    print('After Clean GPU')
-    #GPUtil.showUtilization()
-
-def clone_voice(in_fpath, text):
-    try:       
-            speaker_name = "output"
-            # Compute embedding
-            embed=compute_embedding(in_fpath)
-            print("Created the embedding")
-            # Generating the spectrogram
-            spec, breaks, sample_rate = create_spectrogram(text,embed)
-            current_generated = (speaker_name, spec, breaks)
-            print("Created the mel spectrogram")
-
-            # Create waveform
-            generated_wav=generate_waveform(current_generated)
-            print("Created the the waveform ")
-
-            # Save it on the disk
-            save_on_disk(generated_wav,sample_rate)
-
-            #Play the audio 
-            #play_audio(generated_wav,sample_rate)
-
-            return        
-    except Exception as e:
-        print("Caught exception: %s" % repr(e))
-        print("Restarting\n")
-
-# Set environment variables
-home_dir = os.getcwd()
-OUT_PATH=os.path.join(home_dir, "out/")
-os.environ['OUT_PATH'] = OUT_PATH
-
-# create output path
-os.makedirs(OUT_PATH, exist_ok=True)
-
-USE_CUDA = torch.cuda.is_available()  
-
-os.system('pip install -q pydub ffmpeg-normalize')
-CONFIG_SE_PATH = "config_se.json"
-CHECKPOINT_SE_PATH = "SE_checkpoint.pth.tar"
-def greet(Text,Voicetoclone ,input_mic=None):
-    text= "%s" % (Text)
-    #reference_files= "%s" % (Voicetoclone)
-
-    clean_memory()
-    print(text,len(text),type(text))
-    print(Voicetoclone,type(Voicetoclone))
-
-    if  len(text) == 0 : 
-        print("Please add text to the program")
-        Text="Please add text to the program, thank you."
-        is_no_text=True
-    else:
-        is_no_text=False
-
-    
-    if Voicetoclone==None and input_mic==None:
-        print("There is no input audio")
-        Text="Please add audio input, to the program, thank you."
-        Voicetoclone='trump.mp3'
-        if  is_no_text:
-            Text="Please add text and audio, to the program, thank you."
-
-    if  input_mic != "" and input_mic != None :
-        # Get the wav file from the microphone
-        print('The value of MIC IS :',input_mic,type(input_mic))
-        Voicetoclone= input_mic
-
-    text= "%s" % (Text)
-    reference_files= Voicetoclone
-    print("path url")
-    print(Voicetoclone)
-    sample= str(Voicetoclone)
-    os.environ['sample'] = sample
-    size= len(reference_files)*sys.getsizeof(reference_files)
-    size2= size / 1000000
-    if (size2 > 0.012) or len(text)>2000:
-      message="File is greater than 30mb or Text inserted is longer than 2000 characters. Please re-try with smaller sizes."
-      print(message)
-      raise SystemExit("File is greater than 30mb. Please re-try or Text inserted is longer than 2000 characters. Please re-try with smaller sizes.")
-    else:
-
-      env_var = 'sample'
-      if env_var in os.environ:
-            print(f'{env_var} value is {os.environ[env_var]}')
-      else:
-            print(f'{env_var} does not exist')
-      #os.system(f'ffmpeg-normalize {os.environ[env_var]} -nt rms -t=-27 -o {os.environ[env_var]} -ar 16000 -f')
-      in_fpath = Path(Voicetoclone)
-      #in_fpath= in_fpath.replace("\"", "").replace("\'", "")
-      
-      out_path=clone_voice(in_fpath, text)
-
-      print(" > text: {}".format(text))
-
-      print("Generated Audio")
-      return "cloned_voice.wav"
-
-demo = gr.Interface(
-    fn=greet, 
-    inputs=[gr.inputs.Textbox(label='What would you like the voice to say? (max. 2000 characters per request)'),
-            gr.Audio(
-            type="filepath",         
-            source="upload",
-            label='Please upload a voice to clone (max. 30mb)'),
-            gr.inputs.Audio(
-            source="microphone", 
-            label='or record',
-            type="filepath", 
-            optional=True)
-            ],
-    outputs="audio",
-
-    title = 'Clone Your Voice',
-            description = 'A simple application that Clone Your Voice.  Wait one minute to process.',
-            article = 
-                        '''<div>
-                            <p style="text-align: center"> All you need to do is record your voice, type what you want be say
-                            ,then wait for compiling. After that click on Play/Pause for listen the audio. The audio is saved in an wav format.
-                            For more information visit <a href="https://ruslanmv.com/">ruslanmv.com</a>
-                            </p>
-                        </div>''',
-
-           examples = [["I am the cloned version of Donald Trump. Well. I think what's happening to this country is unbelievably bad. We're no longer a respected country","trump.mp3","trump.mp3"],
-                        ["I am the cloned version of Elon Musk. Persistence is very important. You should not give up unless you are forced to give up.","musk.mp3","musk.mp3"] ,
-                        ["I am the cloned version of Elizabeth. It has always been easy to hate and destroy. To build and to cherish is much more difficult." ,"queen.mp3","queen.mp3"]                    
-                      ]      
-    
-    )
+import gradio as gr
+import os
+from utils.default_models import ensure_default_models
+import sys
+import traceback
+from pathlib import Path
+from time import perf_counter as timer
+import numpy as np
+import torch
+from encoder import inference as encoder
+from synthesizer.inference import Synthesizer
+#from toolbox.utterance import Utterance
+from vocoder import inference as vocoder
+import time
+import librosa
+import numpy as np
+#import sounddevice as sd
+import soundfile as sf
+import argparse
+from utils.argutils import print_args
+
+parser = argparse.ArgumentParser(
+    formatter_class=argparse.ArgumentDefaultsHelpFormatter
+)
+parser.add_argument("-e", "--enc_model_fpath", type=Path,
+                    default="saved_models/default/encoder.pt",
+                    help="Path to a saved encoder")
+parser.add_argument("-s", "--syn_model_fpath", type=Path,
+                    default="saved_models/default/synthesizer.pt",
+                    help="Path to a saved synthesizer")
+parser.add_argument("-v", "--voc_model_fpath", type=Path,
+                    default="saved_models/default/vocoder.pt",
+                    help="Path to a saved vocoder")
+parser.add_argument("--cpu", action="store_true", help=\
+    "If True, processing is done on CPU, even when a GPU is available.")
+parser.add_argument("--no_sound", action="store_true", help=\
+    "If True, audio won't be played.")
+parser.add_argument("--seed", type=int, default=None, help=\
+    "Optional random number seed value to make toolbox deterministic.")
+args = parser.parse_args()
+arg_dict = vars(args)
+print_args(args, parser)
+
+# Maximum of generated wavs to keep on memory
+MAX_WAVS = 15
+utterances = set()
+current_generated = (None, None, None, None) # speaker_name, spec, breaks, wav
+synthesizer = None # type: Synthesizer
+current_wav = None
+waves_list = []
+waves_count = 0
+waves_namelist = []
+
+# Hide GPUs from Pytorch to force CPU processing
+if arg_dict.pop("cpu"):
+    os.environ["CUDA_VISIBLE_DEVICES"] = "-1"
+
+print("Running a test of your configuration...\n")
+
+if torch.cuda.is_available():
+    device_id = torch.cuda.current_device()
+    gpu_properties = torch.cuda.get_device_properties(device_id)
+    ## Print some environment information (for debugging purposes)
+    print("Found %d GPUs available. Using GPU %d (%s) of compute capability %d.%d with "
+        "%.1fGb total memory.\n" %
+        (torch.cuda.device_count(),
+        device_id,
+        gpu_properties.name,
+        gpu_properties.major,
+        gpu_properties.minor,
+        gpu_properties.total_memory / 1e9))
+else:
+    print("Using CPU for inference.\n")
+
+## Load the models one by one.
+print("Preparing the encoder, the synthesizer and the vocoder...")
+ensure_default_models(Path("saved_models"))
+#encoder.load_model(args.enc_model_fpath)
+#synthesizer = Synthesizer(args.syn_model_fpath)
+#vocoder.load_model(args.voc_model_fpath)
+
+def compute_embedding(in_fpath):
+
+    if not encoder.is_loaded():
+        model_fpath = args.enc_model_fpath
+        print("Loading the encoder %s... " % model_fpath)
+        start = time.time() 
+        encoder.load_model(model_fpath)
+        print("Done (%dms)." % int(1000 * (time.time() - start)), "append")
+
+
+    ## Computing the embedding
+    # First, we load the wav using the function that the speaker encoder provides. This is
+    
+    # Get the wav from the disk. We take the wav with the vocoder/synthesizer format for
+    # playback, so as to have a fair comparison with the generated audio
+    wav = Synthesizer.load_preprocess_wav(in_fpath)
+    
+    # important: there is preprocessing that must be applied.
+
+    # The following two methods are equivalent:
+    # - Directly load from the filepath:
+    preprocessed_wav = encoder.preprocess_wav(wav)
+
+    # - If the wav is already loaded:
+    #original_wav, sampling_rate = librosa.load(str(in_fpath))
+    #preprocessed_wav = encoder.preprocess_wav(original_wav, sampling_rate)
+
+    # Compute the embedding
+    embed, partial_embeds, _ = encoder.embed_utterance(preprocessed_wav, return_partials=True)
+
+
+    print("Loaded file succesfully")
+
+    # Then we derive the embedding. There are many functions and parameters that the
+    # speaker encoder interfaces. These are mostly for in-depth research. You will typically
+    # only use this function (with its default parameters):
+    #embed = encoder.embed_utterance(preprocessed_wav)
+    
+    return embed 
+def create_spectrogram(text,embed):
+        # If seed is specified, reset torch seed and force synthesizer reload
+        if args.seed is not None:
+            torch.manual_seed(args.seed)
+            synthesizer = Synthesizer(args.syn_model_fpath)
+        
+        
+        # Synthesize the spectrogram
+        model_fpath = args.syn_model_fpath
+        print("Loading the synthesizer %s... " % model_fpath)
+        start = time.time()
+        synthesizer = Synthesizer(model_fpath)
+        print("Done (%dms)." % int(1000 * (time.time()- start)), "append")          
+        
+
+        # The synthesizer works in batch, so you need to put your data in a list or numpy array
+        texts = [text]
+        embeds = [embed]
+        # If you know what the attention layer alignments are, you can retrieve them here by
+        # passing return_alignments=True
+        specs = synthesizer.synthesize_spectrograms(texts, embeds)
+        breaks = [spec.shape[1] for spec in specs]
+        spec = np.concatenate(specs, axis=1)
+        sample_rate=synthesizer.sample_rate
+        return spec, breaks , sample_rate
+
+
+def generate_waveform(current_generated):
+
+        speaker_name, spec, breaks = current_generated
+        assert spec is not None
+
+        ## Generating the waveform
+        print("Synthesizing the waveform:")
+        # If seed is specified, reset torch seed and reload vocoder
+        if args.seed is not None:
+            torch.manual_seed(args.seed)
+            vocoder.load_model(args.voc_model_fpath)
+
+        model_fpath = args.voc_model_fpath
+        # Synthesize the waveform
+        if not vocoder.is_loaded():
+            print("Loading the vocoder %s... " % model_fpath)
+            start = time.time()
+            vocoder.load_model(model_fpath)
+            print("Done (%dms)." % int(1000 * (time.time()- start)), "append")    
+
+        current_vocoder_fpath= model_fpath
+        def vocoder_progress(i, seq_len, b_size, gen_rate):
+            real_time_factor = (gen_rate / Synthesizer.sample_rate) * 1000
+            line = "Waveform generation: %d/%d (batch size: %d, rate: %.1fkHz - %.2fx real time)" \
+                % (i * b_size, seq_len * b_size, b_size, gen_rate, real_time_factor)
+            print(line, "overwrite")       
+
+
+        # Synthesizing the waveform is fairly straightforward. Remember that the longer the
+        # spectrogram, the more time-efficient the vocoder.
+        if  current_vocoder_fpath is not None:
+            print("")
+            generated_wav = vocoder.infer_waveform(spec, progress_callback=vocoder_progress)
+        else:
+            print("Waveform generation with Griffin-Lim... ")
+            generated_wav = Synthesizer.griffin_lim(spec)
+
+        print(" Done!", "append")
+
+
+        ## Post-generation
+        # There's a bug with sounddevice that makes the audio cut one second earlier, so we
+        # pad it.
+        generated_wav = np.pad(generated_wav, (0, Synthesizer.sample_rate), mode="constant")
+
+        # Add breaks
+        b_ends = np.cumsum(np.array(breaks) * Synthesizer.hparams.hop_size)
+        b_starts = np.concatenate(([0], b_ends[:-1]))
+        wavs = [generated_wav[start:end] for start, end, in zip(b_starts, b_ends)]
+        breaks = [np.zeros(int(0.15 * Synthesizer.sample_rate))] * len(breaks)
+        generated_wav = np.concatenate([i for w, b in zip(wavs, breaks) for i in (w, b)])
+
+
+        # Trim excess silences to compensate for gaps in spectrograms (issue #53)
+        generated_wav = encoder.preprocess_wav(generated_wav)
+
+
+        return generated_wav
+
+
+def save_on_disk(generated_wav,sample_rate):
+        # Save it on the disk
+        filename = "cloned_voice.wav"
+        print(generated_wav.dtype)
+        #OUT=os.environ['OUT_PATH']
+        # Returns `None` if key doesn't exist
+        #OUT=os.environ.get('OUT_PATH')
+        #result = os.path.join(OUT, filename)
+        result = filename
+        print(" > Saving output to {}".format(result))
+        sf.write(result, generated_wav.astype(np.float32), sample_rate)
+        print("\nSaved output as %s\n\n" % result) 
+      
+        return  result     
+def play_audio(generated_wav,sample_rate):
+        # Play the audio (non-blocking)
+        if not args.no_sound:
+          
+            try:
+                sd.stop()
+                sd.play(generated_wav, sample_rate)
+            except sd.PortAudioError as e:
+                print("\nCaught exception: %s" % repr(e))
+                print("Continuing without audio playback. Suppress this message with the \"--no_sound\" flag.\n")
+            except:
+                raise
+         
+
+def clean_memory():
+    import gc
+    #import GPUtil
+    # To see memory usage
+    print('Before clean ')
+    #GPUtil.showUtilization()
+    #cleaning memory 1
+    gc.collect()
+    torch.cuda.empty_cache()
+    time.sleep(2)
+    print('After Clean GPU')
+    #GPUtil.showUtilization()
+
+def clone_voice(in_fpath, text):
+    try:       
+            speaker_name = "output"
+            # Compute embedding
+            embed=compute_embedding(in_fpath)
+            print("Created the embedding")
+            # Generating the spectrogram
+            spec, breaks, sample_rate = create_spectrogram(text,embed)
+            current_generated = (speaker_name, spec, breaks)
+            print("Created the mel spectrogram")
+
+            # Create waveform
+            generated_wav=generate_waveform(current_generated)
+            print("Created the the waveform ")
+
+            # Save it on the disk
+            save_on_disk(generated_wav,sample_rate)
+
+            #Play the audio 
+            #play_audio(generated_wav,sample_rate)
+
+            return        
+    except Exception as e:
+        print("Caught exception: %s" % repr(e))
+        print("Restarting\n")
+
+# Set environment variables
+home_dir = os.getcwd()
+OUT_PATH=os.path.join(home_dir, "out/")
+os.environ['OUT_PATH'] = OUT_PATH
+
+# create output path
+os.makedirs(OUT_PATH, exist_ok=True)
+
+USE_CUDA = torch.cuda.is_available()  
+
+os.system('pip install -q pydub ffmpeg-normalize')
+CONFIG_SE_PATH = "config_se.json"
+CHECKPOINT_SE_PATH = "SE_checkpoint.pth.tar"
+def greet(Text,Voicetoclone ,input_mic=None):
+    text= "%s" % (Text)
+    #reference_files= "%s" % (Voicetoclone)
+
+    clean_memory()
+    print(text,len(text),type(text))
+    print(Voicetoclone,type(Voicetoclone))
+
+    if  len(text) == 0 : 
+        print("Please add text to the program")
+        Text="Please add text to the program, thank you."
+        is_no_text=True
+    else:
+        is_no_text=False
+
+    
+    if Voicetoclone==None and input_mic==None:
+        print("There is no input audio")
+        Text="Please add audio input, to the program, thank you."
+        Voicetoclone='trump.mp3'
+        if  is_no_text:
+            Text="Please add text and audio, to the program, thank you."
+
+    if  input_mic != "" and input_mic != None :
+        # Get the wav file from the microphone
+        print('The value of MIC IS :',input_mic,type(input_mic))
+        Voicetoclone= input_mic
+
+    text= "%s" % (Text)
+    reference_files= Voicetoclone
+    print("path url")
+    print(Voicetoclone)
+    sample= str(Voicetoclone)
+    os.environ['sample'] = sample
+    size= len(reference_files)*sys.getsizeof(reference_files)
+    size2= size / 1000000
+    if (size2 > 0.012) or len(text)>2000:
+      message="File is greater than 30mb or Text inserted is longer than 2000 characters. Please re-try with smaller sizes."
+      print(message)
+      raise SystemExit("File is greater than 30mb. Please re-try or Text inserted is longer than 2000 characters. Please re-try with smaller sizes.")
+    else:
+
+      env_var = 'sample'
+      if env_var in os.environ:
+            print(f'{env_var} value is {os.environ[env_var]}')
+      else:
+            print(f'{env_var} does not exist')
+      #os.system(f'ffmpeg-normalize {os.environ[env_var]} -nt rms -t=-27 -o {os.environ[env_var]} -ar 16000 -f')
+      in_fpath = Path(Voicetoclone)
+      #in_fpath= in_fpath.replace("\"", "").replace("\'", "")
+      
+      out_path=clone_voice(in_fpath, text)
+
+      print(" > text: {}".format(text))
+
+      print("Generated Audio")
+      return "cloned_voice.wav"
+
+demo = gr.Interface(
+    fn=greet, 
+    inputs=[gr.inputs.Textbox(label='What would you like the voice to say? (max. 2000 characters per request)'),
+            gr.Audio(
+            type="filepath",         
+            source="upload",
+            label='Please upload a voice to clone (max. 30mb)'),
+            gr.inputs.Audio(
+            source="microphone", 
+            label='or record',
+            type="filepath", 
+            optional=True)
+            ],
+    outputs="audio",
+
+    title = 'Clone Your Voice',
+            description = 'A simple application that Clone Your Voice.  Wait one minute to process.',
+            article = 
+                        '''<div>
+                            <p style="text-align: center"> All you need to do is record your voice, type what you want be say
+                            ,then wait for compiling. After that click on Play/Pause for listen the audio. The audio is saved in an wav format.
+                            For more information visit <a href="https://ruslanmv.com/">ruslanmv.com</a>
+                            </p>
+                        </div>''',
+
+           #examples = [["I am the cloned version of Donald Trump. Well. I think what's happening to this country is unbelievably bad. We're no longer a respected country","trump.mp3","trump.mp3"],
+           #             ["I am the cloned version of Elon Musk. Persistence is very important. You should not give up unless you are forced to give up.","musk.mp3","musk.mp3"] ,
+           #             ["I am the cloned version of Elizabeth. It has always been easy to hate and destroy. To build and to cherish is much more difficult." ,"queen.mp3","queen.mp3"]                    
+           #           ]      
+    
+    )
 demo.launch()