app.py

import time
import streamlit as st
# from transformers import pipeline
import os
import torch
import datetime
import numpy as np
import soundfile
from wavmark.utils import file_reader
from audioseal import AudioSeal
import torchaudio
from pydub import AudioSegment

# pipeline = pipeline(task="image-classification", model="julien-c/hotdog-not-hotdog")

# st.title("Hot Dog? Or Not?")

# file_name = st.file_uploader("Upload a hot dog candidate image")

# if file_name is not None:
#     col1, col2 = st.columns(2)

#     image = Image.open(file_name)
#     col1.image(image, use_column_width=True)
#     predictions = pipeline(image)

#     col2.header("Probabilities")
#     for p in predictions:
#         col2.subheader(f"{ p['label'] }: { round(p['score'] * 100, 1)}%")

def read_as_single_channel_16k(audio_file, def_sr=16000, verbose=True, aim_second=None):
    assert os.path.exists(audio_file)

    file_extension = os.path.splitext(audio_file)[1].lower()

    if file_extension == ".mp3":
        data, origin_sr = librosa.load(audio_file, sr=None)
    elif file_extension in [".wav", ".flac"]:
        data, origin_sr = soundfile.read(audio_file)
    else:
        raise Exception("unsupported file:" + file_extension)

    # channel check
    if len(data.shape) == 2:
        left_channel = data[:, 0]
        if verbose:
            print("Warning! the input audio has multiple chanel, this tool only use the first channel!")
        data = left_channel

    # sample rate check
    if origin_sr != def_sr:
        data = resampy.resample(data, origin_sr, def_sr)
        if verbose:
            print("Warning! The original samplerate is not 16Khz; the watermarked audio will be re-sampled to 16KHz")

    sr = def_sr
    audio_length_second = 1.0 * len(data) / sr
    # if verbose:
    #     print("input length :%d second" % audio_length_second)

    if aim_second is not None:
        signal = data
        assert len(signal) > 0
        current_second = len(signal) / sr
        if current_second < aim_second:
            repeat_count = int(aim_second / current_second) + 1
            signal = np.repeat(signal, repeat_count)
        data = signal[0:sr * aim_second]

    return data, sr, audio_length_second

    
def my_read_file(audio_path, max_second):
    signal, sr, audio_length_second = read_as_single_channel_16k(audio_path, default_sr)
    if audio_length_second > max_second:
        signal = signal[0:default_sr * max_second]
        audio_length_second = max_second

    return signal, sr, audio_length_second

def create_default_value():
    if "def_value" not in st.session_state:
        def_val_npy = np.random.choice([0, 1], size=32 - len_start_bit)
        def_val_str = "".join([str(i) for i in def_val_npy])
        st.session_state.def_value = def_val_str

# Main web app
def main():
    create_default_value()

    # st.title("MDS07")
    # st.write("https://github.com/wavmark/wavmark")
    markdown_text = """
    # MDS07
    [AudioSeal](https://github.com/jcha0155/AudioSealEnhanced) is the next-generation watermarking tool driven by AI. 
    You can upload an audio file and encode a custom 16-bit watermark or perform decoding from a watermarked audio.
    
    This page is for demonstration usage and only process **the first minute** of the audio. 
    If you have longer files for processing, we recommend using [our python toolkit](https://github.com/jcha0155/AudioSealEnhanced).
    """

    # 使用st.markdown渲染Markdown文本
    st.markdown(markdown_text)

    audio_file = st.file_uploader("Upload Audio", type=["wav", "mp3"], accept_multiple_files=False)

    if audio_file:
        # 保存文件到本地：
        # tmp_input_audio_file = os.path.join("/tmp/", audio_file.name)
        # st.markdown(tmp_input_audio_file)
        # with open(tmp_input_audio_file, "wb") as f:
        #     f.write(audio_file.getbuffer())
        # st.audio(tmp_input_audio_file, format="mp3/wav")

        # Save file to local storage
        tmp_input_audio_file = os.path.join("/tmp/", audio_file.name)
        with open(tmp_input_audio_file, "wb") as f:
            f.write(audio_file.getbuffer())

        # Convert MP3 to WAV using pydub
        mp3_audio = AudioSegment.from_mp3(tmp_input_audio_file)
        wav_output_file = tmp_input_audio_file.replace(".mp3", ".wav")
        mp3_audio.export(wav_output_file, format="wav")

        # Load the WAV file using torchaudio
        wav, sample_rate = torchaudio.load(wav_output_file)
        st.markdown("Before unsquuezewav")
        st.markdown(wav)
        wav= wav.unsqueeze(0)

        # Play the audio file (WAV format)
        st.audio(wav_output_file, format="audio/wav")
        
        # wav, sample_rate = torchaudio.load(audio_file, format="mp3/wav")
        st.markdown("SR")
        st.markdown(sample_rate)
        st.markdown("after unsqueeze wav")
        st.markdown(wav)
        # 展示文件到页面上
        # st.audio(tmp_input_audio_file, format="audio/wav")

        action = st.selectbox("Select Action", ["Add Watermark", "Decode Watermark"])

        if action == "Add Watermark":
            watermark_text = st.text_input("The watermark (0, 1 list of length-16):", value=st.session_state.def_value)
            add_watermark_button = st.button("Add Watermark", key="add_watermark_btn")
            if add_watermark_button:  # 点击按钮后执行的
                if audio_file and watermark_text:
                    with st.spinner("Adding Watermark..."):
                        wav = my_read_file(wav,max_second_encode)
                        watermark = model.get_watermark(wav, default_sr)
                        # watermarked_audio, encode_time_cost = add_watermark(tmp_input_audio_file, watermark_text)
#                         st.write("Watermarked Audio:")
#                         print("watermarked_audio:", watermarked_audio)
#                         st.audio(watermarked_audio, format="audio/wav")
#                         st.write("Time Cost: %d seconds" % encode_time_cost)

#                         # st.button("Add Watermark", disabled=False)
#         elif action == "Decode Watermark":
#             if st.button("Decode"):
#                 with st.spinner("Decoding..."):
#                     decode_watermark(tmp_input_audio_file)


if __name__ == "__main__":
    default_sr = 16000
    max_second_encode = 60
    max_second_decode = 30
    len_start_bit = 16
    device = torch.device('cuda:0' if torch.cuda.is_available() else 'cpu')
    # model = wavmark.load_model().to(device)
    model = AudioSeal.load_generator("audioseal_wm_16bits")
    main()

    # audio_path = "/Users/my/Library/Mobile Documents/com~apple~CloudDocs/CODE/PycharmProjects/4_语音水印/419_huggingface水印/WavMark/example.wav"

    # decoded_watermark, decode_cost = decode_watermark(audio_path)
    # print(decoded_watermark)