diff --git a/README.md b/README.md
index d7c6d86..ae7c95b 100644
--- a/README.md
+++ b/README.md
@@ -13,6 +13,8 @@ There are three ways (besides running Gradio in Colab):
 1. More flexible inference beyond Gradio UI in Google Colab. see [quickstart colab](#quickstart-colab)
 2. with docker. see [quickstart docker](#quickstart-docker)
 3. without docker. see [environment setup](#environment-setup). You can also run gradio locally if you choose this option
+4. As a standalone script that you can easily integrate into other projects.
+see [quickstart command line](#quickstart-command-line).
 
 When you are inside the docker image or you have installed all dependencies, Checkout [`inference_tts.ipynb`](./inference_tts.ipynb).
 
@@ -21,7 +23,7 @@ If you want to do model development such as training/finetuning, I recommend fol
 ## News
 :star: 04/22/2024: 330M/830M TTS Enhanced Models are up [here](https://huggingface.co/pyp1), load them through [`gradio_app.py`](./gradio_app.py) or [`inference_tts.ipynb`](./inference_tts.ipynb)! Replicate demo is up, major thanks to [@chenxwh](https://github.com/chenxwh)!
 
-:star: 04/11/2024: VoiceCraft Gradio is now available on HuggingFace Spaces [here](https://huggingface.co/spaces/pyp1/VoiceCraft_gradio)! Major thanks to [@zuev-stepan](https://github.com/zuev-stepan), [@Sewlell](https://github.com/Sewlell), [@pgsoar](https://github.com/pgosar) [@Ph0rk0z](https://github.com/Ph0rk0z). 
+:star: 04/11/2024: VoiceCraft Gradio is now available on HuggingFace Spaces [here](https://huggingface.co/spaces/pyp1/VoiceCraft_gradio)! Major thanks to [@zuev-stepan](https://github.com/zuev-stepan), [@Sewlell](https://github.com/Sewlell), [@pgsoar](https://github.com/pgosar) [@Ph0rk0z](https://github.com/Ph0rk0z).
 
 :star: 04/05/2024: I finetuned giga330M with the TTS objective on gigaspeech and 1/5 of librilight. Weights are [here](https://huggingface.co/pyp1/VoiceCraft/tree/main). Make sure maximal prompt + generation length <= 16 seconds (due to our limited compute, we had to drop utterances longer than 16s in training data). Even stronger models forthcomming, stay tuned!
 
@@ -37,11 +39,9 @@ If you want to do model development such as training/finetuning, I recommend fol
 - [x] Better guidance on training/finetuning
 - [x] Colab notebooks
 - [x] HuggingFace Spaces demo
-- [ ] Command line
+- [x] Command line
 - [ ] Improve efficiency
 
-
-
 ## QuickStart Colab
 
 :star: To try out speech editing or TTS Inference with VoiceCraft, the simplest way is using Google Colab.
@@ -50,6 +50,15 @@ Instructions to run are on the Colab itself.
 1. To try [Speech Editing](https://colab.research.google.com/drive/1FV7EC36dl8UioePY1xXijXTMl7X47kR_?usp=sharing)
 2. To try [TTS Inference](https://colab.research.google.com/drive/1lch_6it5-JpXgAQlUTRRI2z2_rk5K67Z?usp=sharing)
 
+## QuickStart Command Line
+
+:star: To use it as a standalone script, check out tts_demo.py and speech_editing_demo.py.
+Be sure to first [setup your environment](#environment-setup).
+Without arguments, they will run the standard demo arguments used as an example elsewhere
+in this repository. You can use the command line arguments to specify unique input audios,
+target transcripts, and inference hyperparameters. Run the help command for more information:
+`python3 tts_demo.py -h`
+
 ## QuickStart Docker
 :star: To try out TTS inference with VoiceCraft, you can also use docker. Thank [@ubergarm](https://github.com/ubergarm) and [@jayc88](https://github.com/jay-c88) for making this happen.
 
@@ -197,7 +206,7 @@ cd ./z_scripts
 bash e830M.sh
 ```
 
-It's the same procedure to prepare your own custom dataset. Make sure that if 
+It's the same procedure to prepare your own custom dataset. Make sure that if
 
 ## Finetuning
 You also need to do step 1-4 as Training, and I recommend to use AdamW for optimization if you finetune a pretrained model for better stability. checkout script `./z_scripts/e830M_ft.sh`.
diff --git a/demo/temp/mfa_alignments/5895_34622_000026_000002.csv b/demo/temp/mfa_alignments/5895_34622_000026_000002.csv
new file mode 100644
index 0000000..81a998d
--- /dev/null
+++ b/demo/temp/mfa_alignments/5895_34622_000026_000002.csv
@@ -0,0 +1,106 @@
+Begin,End,Label,Type,Speaker
+0.04,0.58,gwynplaine,words,temp
+0.58,0.94,had,words,temp
+0.94,1.45,besides,words,temp
+1.45,1.62,for,words,temp
+1.62,1.86,his,words,temp
+1.86,2.16,work,words,temp
+2.16,2.31,and,words,temp
+2.31,2.49,for,words,temp
+2.49,2.71,his,words,temp
+2.71,3.03,feats,words,temp
+3.03,3.12,of,words,temp
+3.12,3.61,strength,words,temp
+3.95,4.25,round,words,temp
+4.25,4.45,his,words,temp
+4.45,4.7,neck,words,temp
+4.7,4.81,and,words,temp
+4.81,5.04,over,words,temp
+5.04,5.22,his,words,temp
+5.22,5.83,shoulders,words,temp
+6.16,6.31,an,words,temp
+6.41,7.15,esclavine,words,temp
+7.15,7.29,of,words,temp
+7.29,7.7,leather,words,temp
+0.04,0.1,G,phones,temp
+0.1,0.13,W,phones,temp
+0.13,0.22,IH1,phones,temp
+0.22,0.3,N,phones,temp
+0.3,0.38,P,phones,temp
+0.38,0.42,L,phones,temp
+0.42,0.53,EY1,phones,temp
+0.53,0.58,N,phones,temp
+0.58,0.71,HH,phones,temp
+0.71,0.86,AE1,phones,temp
+0.86,0.94,D,phones,temp
+0.94,0.97,B,phones,temp
+0.97,1.01,IH0,phones,temp
+1.01,1.14,S,phones,temp
+1.14,1.34,AY1,phones,temp
+1.34,1.4,D,phones,temp
+1.4,1.45,Z,phones,temp
+1.45,1.52,F,phones,temp
+1.52,1.55,AO1,phones,temp
+1.55,1.62,R,phones,temp
+1.62,1.69,HH,phones,temp
+1.69,1.76,IH1,phones,temp
+1.76,1.86,Z,phones,temp
+1.86,1.95,W,phones,temp
+1.95,2.07,ER1,phones,temp
+2.07,2.16,K,phones,temp
+2.16,2.23,AH0,phones,temp
+2.23,2.26,N,phones,temp
+2.26,2.31,D,phones,temp
+2.31,2.38,F,phones,temp
+2.38,2.41,AO1,phones,temp
+2.41,2.49,R,phones,temp
+2.49,2.55,HH,phones,temp
+2.55,2.62,IH1,phones,temp
+2.62,2.71,Z,phones,temp
+2.71,2.8,F,phones,temp
+2.8,2.9,IY1,phones,temp
+2.9,2.98,T,phones,temp
+2.98,3.03,S,phones,temp
+3.03,3.07,AH0,phones,temp
+3.07,3.12,V,phones,temp
+3.12,3.2,S,phones,temp
+3.2,3.26,T,phones,temp
+3.26,3.32,R,phones,temp
+3.32,3.39,EH1,phones,temp
+3.39,3.48,NG,phones,temp
+3.48,3.53,K,phones,temp
+3.53,3.61,TH,phones,temp
+3.95,4.03,R,phones,temp
+4.03,4.16,AW1,phones,temp
+4.16,4.21,N,phones,temp
+4.21,4.25,D,phones,temp
+4.25,4.29,HH,phones,temp
+4.29,4.36,IH1,phones,temp
+4.36,4.45,Z,phones,temp
+4.45,4.53,N,phones,temp
+4.53,4.62,EH1,phones,temp
+4.62,4.7,K,phones,temp
+4.7,4.74,AH0,phones,temp
+4.74,4.77,N,phones,temp
+4.77,4.81,D,phones,temp
+4.81,4.92,OW1,phones,temp
+4.92,4.97,V,phones,temp
+4.97,5.04,ER0,phones,temp
+5.04,5.11,HH,phones,temp
+5.11,5.18,IH1,phones,temp
+5.18,5.22,Z,phones,temp
+5.22,5.34,SH,phones,temp
+5.34,5.47,OW1,phones,temp
+5.47,5.51,L,phones,temp
+5.51,5.58,D,phones,temp
+5.58,5.71,ER0,phones,temp
+5.71,5.83,Z,phones,temp
+6.16,6.23,AE1,phones,temp
+6.23,6.31,N,phones,temp
+6.41,7.15,spn,phones,temp
+7.15,7.21,AH0,phones,temp
+7.21,7.29,V,phones,temp
+7.29,7.36,L,phones,temp
+7.36,7.44,EH1,phones,temp
+7.44,7.49,DH,phones,temp
+7.49,7.7,ER0,phones,temp
diff --git a/tts_demo.py b/tts_demo.py
new file mode 100644
index 0000000..c1d97ce
--- /dev/null
+++ b/tts_demo.py
@@ -0,0 +1,216 @@
+"""
+This script will allow you to run TTS inference with Voicecraft
+Before getting started, be sure to follow the environment setup.
+"""
+
+from inference_tts_scale import inference_one_sample
+from models import voicecraft
+from data.tokenizer import (
+    AudioTokenizer,
+    TextTokenizer,
+)
+import argparse
+import random
+import numpy as np
+import torchaudio
+import torch
+import os
+os.environ["USER"] = "me"  # TODO change this to your username
+
+device = "cuda" if torch.cuda.is_available() else "cpu"
+
+
+def parse_arguments():
+    parser = argparse.ArgumentParser(
+        description="VoiceCraft TTS Inference: see the script for more information on the options")
+
+    parser.add_argument("-m", "--model_name", type=str, default="giga830M", choices=[
+                        "giga330M", "giga830M", "giga330M_TTSEnhanced", "giga830M_TTSEnhanced"],
+                        help="VoiceCraft model to use")
+    parser.add_argument("-st", "--silence_tokens", type=int, nargs="*",
+                        default=[1388, 1898, 131], help="Silence token IDs")
+    parser.add_argument("-casr", "--codec_audio_sr", type=int,
+                        default=16000, help="Codec audio sample rate.")
+    parser.add_argument("-csr", "--codec_sr", type=int, default=50,
+                        help="Codec sample rate.")
+
+    parser.add_argument("-k", "--top_k", type=float,
+                        default=0, help="Top k value.")
+    parser.add_argument("-p", "--top_p", type=float,
+                        default=0.8, help="Top p value.")
+    parser.add_argument("-t", "--temperature", type=float,
+                        default=1, help="Temperature value.")
+    parser.add_argument("-kv", "--kvcache", type=float, choices=[0, 1],
+                        default=0, help="Kvcache value.")
+    parser.add_argument("-sr", "--stop_repetition", type=int,
+                        default=-1, help="Stop repetition for generation")
+    parser.add_argument("--sample_batch_size", type=int,
+                        default=3, help="Batch size for sampling")
+    parser.add_argument("-s", "--seed", type=int,
+                        default=1, help="Seed value.")
+    parser.add_argument("-bs", "--beam_size", type=int, default=50,
+                        help="beam size for MFA alignment")
+    parser.add_argument("-rbs", "--retry_beam_size", type=int, default=200,
+                        help="retry beam size for MFA alignment")
+    parser.add_argument("--output_dir", type=str, default="./generated_tts",
+                        help="directory to save generated audio")
+    parser.add_argument("-oa", "--original_audio", type=str,
+                        default="./demo/5895_34622_000026_000002.wav", help="location of audio file")
+    parser.add_argument("-ot", "--original_transcript", type=str,
+                        default="Gwynplaine had, besides, for his work and for his feats of strength, round his neck and over his shoulders, an esclavine of leather.",
+                        help="original transcript")
+    parser.add_argument("-tt", "--target_transcript", type=str,
+                        default="I cannot believe that the same model can also do text to speech synthesis too!",
+                        help="target transcript")
+    parser.add_argument("-co", "--cut_off_sec", type=float, default=3.6,
+                        help="cut off point in seconds for input prompt")
+    parser.add_argument("-ma", "--margin", type=float, default=0.04,
+                    help="margin in seconds between the end of the cutoff words and the start of the next word. If the next word is not immediately following the cutoff word, the algorithm is more tolerant to word alignment errors")
+    parser.add_argument("-cuttol", "--cutoff_tolerance", type=float, default=1, help="tolerance in seconds for the cutoff time, if given cut_off_sec plus the tolerance, we still are not able to find the next word, we will use the best cutoff time found, i.e. likely no margin or very small margin between the end of the cutoff word and the start of the next word")
+
+    args = parser.parse_args()
+    return args
+
+
+args = parse_arguments()
+voicecraft_name = args.model_name
+# hyperparameters for inference
+codec_audio_sr = args.codec_audio_sr
+codec_sr = args.codec_sr
+top_k = args.top_k
+top_p = args.top_p  # defaults to 0.9 can also try 0.8, but 0.9 seems to work better
+temperature = args.temperature
+silence_tokens = args.silence_tokens
+kvcache = args.kvcache  # NOTE if OOM, change this to 0, or try the 330M model
+
+# NOTE adjust the below three arguments if the generation is not as good
+# NOTE if the model generate long silence, reduce the stop_repetition to 3, 2 or even 1
+stop_repetition = args.stop_repetition
+
+# NOTE: if the if there are long silence or unnaturally strecthed words,
+# increase sample_batch_size to 4 or higher. What this will do to the model is that the
+# model will run sample_batch_size examples of the same audio, and pick the one that's the shortest.
+# So if the speech rate of the generated is too fast change it to a smaller number.
+sample_batch_size = args.sample_batch_size
+seed = args.seed  # change seed if you are still unhappy with the result
+
+# load the model
+if voicecraft_name == "330M":
+    voicecraft_name = "giga330M"
+elif voicecraft_name == "830M":
+    voicecraft_name = "giga830M"
+elif voicecraft_name == "330M_TTSEnhanced":
+    voicecraft_name = "330M_TTSEnhanced"
+elif voicecraft_name == "830M_TTSEnhanced":
+    voicecraft_name = "830M_TTSEnhanced"
+model = voicecraft.VoiceCraft.from_pretrained(
+    f"pyp1/VoiceCraft_{voicecraft_name.replace('.pth', '')}")
+phn2num = model.args.phn2num
+config = vars(model.args)
+model.to(device)
+
+encodec_fn = "./pretrained_models/encodec_4cb2048_giga.th"
+if not os.path.exists(encodec_fn):
+    os.system(
+        f"wget https://huggingface.co/pyp1/VoiceCraft/resolve/main/encodec_4cb2048_giga.th -O ./pretrained_models/encodec_4cb2048_giga.th")
+# will also put the neural codec model on gpu
+audio_tokenizer = AudioTokenizer(signature=encodec_fn, device=device)
+
+text_tokenizer = TextTokenizer(backend="espeak")
+
+# Prepare your audio
+# point to the original audio whose speech you want to clone
+# write down the transcript for the file, or run whisper to get the transcript (and you can modify it if it's not accurate), save it as a .txt file
+orig_audio = args.original_audio
+orig_transcript = args.original_transcript
+
+# move the audio and transcript to temp folder
+temp_folder = "./demo/temp"
+os.makedirs(temp_folder, exist_ok=True)
+os.system(f"cp {orig_audio} {temp_folder}")
+filename = os.path.splitext(orig_audio.split("/")[-1])[0]
+with open(f"{temp_folder}/{filename}.txt", "w") as f:
+    f.write(orig_transcript)
+# run MFA to get the alignment
+align_temp = f"{temp_folder}/mfa_alignments"
+beam_size = args.beam_size
+retry_beam_size = args.retry_beam_size
+alignments = f"{temp_folder}/mfa_alignments/{filename}.csv"
+if not os.path.isfile(alignments):
+    os.system(f"mfa align -v --clean -j 1 --output_format csv {temp_folder} \
+            english_us_arpa english_us_arpa {align_temp} --beam {beam_size} --retry_beam {retry_beam_size}")
+# if the above fails, it could be because the audio is too hard for the alignment model,
+# increasing the beam_size and retry_beam_size usually solves the issue
+
+def find_closest_word_boundary(alignments, cut_off_sec, margin, cutoff_tolerance = 1):
+    with open(alignments, 'r') as file:
+        # skip header
+        next(file)
+        cutoff_time = None
+        cutoff_index = None
+        cutoff_time_best = None
+        cutoff_index_best = None
+        lines = [l for l in file.readlines()]
+        for i, line in enumerate(lines):
+            end = float(line.strip().split(',')[1])
+            if end >= cut_off_sec and cutoff_time == None:
+                cutoff_time = end
+                cutoff_index = i
+            if end >= cut_off_sec and end < cut_off_sec + cutoff_tolerance and float(lines[i+1].strip().split(',')[0]) - end >= margin:
+                    cutoff_time_best = end + margin * 2 / 3
+                    cutoff_index_best = i
+                    break
+        if cutoff_time_best != None:
+            cutoff_time = cutoff_time_best
+            cutoff_index = cutoff_index_best
+        return cutoff_time, cutoff_index
+
+# take a look at demo/temp/mfa_alignment, decide which part of the audio to use as prompt
+# NOTE: according to forced-alignment file demo/temp/mfa_alignments/5895_34622_000026_000002.wav, the word "strength" stop as 3.561 sec, so we use first 3.6 sec as the prompt. this should be different for different audio
+cut_off_sec = args.cut_off_sec
+margin = args.margin
+audio_fn = f"{temp_folder}/{filename}.wav"
+
+cut_off_sec, cut_off_word_idx = find_closest_word_boundary(alignments, cut_off_sec, margin, args.cutoff_tolerance)
+target_transcript = " ".join(orig_transcript.split(" ")[:cut_off_word_idx+1]) + " " + args.target_transcript
+# NOTE: 3 sec of reference is generally enough for high quality voice cloning, but longer is generally better, try e.g. 3~6 sec.
+info = torchaudio.info(audio_fn)
+audio_dur = info.num_frames / info.sample_rate
+
+assert cut_off_sec < audio_dur, f"cut_off_sec {cut_off_sec} is larger than the audio duration {audio_dur}"
+prompt_end_frame = int(cut_off_sec * info.sample_rate)
+
+
+def seed_everything(seed):
+    os.environ['PYTHONHASHSEED'] = str(seed)
+    random.seed(seed)
+    np.random.seed(seed)
+    torch.manual_seed(seed)
+    torch.cuda.manual_seed(seed)
+    torch.backends.cudnn.benchmark = False
+    torch.backends.cudnn.deterministic = True
+
+
+seed_everything(seed)
+
+# inference
+decode_config = {'top_k': top_k, 'top_p': top_p, 'temperature': temperature, 'stop_repetition': stop_repetition, 'kvcache': kvcache,
+                 "codec_audio_sr": codec_audio_sr, "codec_sr": codec_sr, "silence_tokens": silence_tokens, "sample_batch_size": sample_batch_size}
+concated_audio, gen_audio = inference_one_sample(model, argparse.Namespace(
+    **config), phn2num, text_tokenizer, audio_tokenizer, audio_fn, target_transcript, device, decode_config, prompt_end_frame)
+
+# save segments for comparison
+concated_audio, gen_audio = concated_audio[0].cpu(), gen_audio[0].cpu()
+# logging.info(f"length of the resynthesize orig audio: {orig_audio.shape}")
+
+# save the audio
+# output_dir
+output_dir = args.output_dir
+os.makedirs(output_dir, exist_ok=True)
+seg_save_fn_gen = f"{output_dir}/{os.path.basename(audio_fn)[:-4]}_gen_seed{seed}.wav"
+seg_save_fn_concat = f"{output_dir}/{os.path.basename(audio_fn)[:-4]}_concat_seed{seed}.wav"
+
+torchaudio.save(seg_save_fn_gen, gen_audio, codec_audio_sr)
+torchaudio.save(seg_save_fn_concat, concated_audio, codec_audio_sr)
+
+# you might get warnings like WARNING:phonemizer:words count mismatch on 300.0% of the lines (3/1), this can be safely ignored