from fractions import Fraction
import torch
import av
from tqdm import tqdm
from PIL import Image


def _resample_audio(
    container: av.container.Container, audio_stream: av.audio.AudioStream, frame_in: av.AudioFrame
) -> None:
    cc = audio_stream.codec_context

    # Use the encoder's format/layout/rate as the *target*
    target_format = cc.format or "fltp"  # AAC → usually fltp
    target_layout = cc.layout or "stereo"
    target_rate = cc.sample_rate or frame_in.sample_rate

    audio_resampler = av.audio.resampler.AudioResampler(
        format=target_format,
        layout=target_layout,
        rate=target_rate,
    )

    audio_next_pts = 0
    for rframe in audio_resampler.resample(frame_in):
        if rframe.pts is None:
            rframe.pts = audio_next_pts
        audio_next_pts += rframe.samples
        rframe.sample_rate = frame_in.sample_rate
        container.mux(audio_stream.encode(rframe))

    # flush audio encoder
    for packet in audio_stream.encode():
        container.mux(packet)


def _write_audio(
    container: av.container.Container, audio_stream: av.audio.AudioStream, samples: torch.Tensor, audio_sample_rate: int
) -> None:
    if samples.ndim == 1:
        samples = samples[:, None]

    if samples.shape[1] != 2 and samples.shape[0] == 2:
        samples = samples.T

    if samples.shape[1] != 2:
        raise ValueError(f"Expected samples with 2 channels; got shape {samples.shape}.")

    # Convert to int16 packed for ingestion; resampler converts to encoder fmt.
    if samples.dtype != torch.int16:
        samples = torch.clip(samples, -1.0, 1.0)
        samples = (samples * 32767.0).to(torch.int16)

    frame_in = av.AudioFrame.from_ndarray(
        samples.contiguous().reshape(1, -1).cpu().numpy(),
        format="s16",
        layout="stereo",
    )
    frame_in.sample_rate = audio_sample_rate

    _resample_audio(container, audio_stream, frame_in)


def _prepare_audio_stream(container: av.container.Container, audio_sample_rate: int) -> av.audio.AudioStream:
    """
    Prepare the audio stream for writing.
    """
    audio_stream = container.add_stream("aac", rate=audio_sample_rate)
    audio_stream.codec_context.sample_rate = audio_sample_rate
    audio_stream.codec_context.layout = "stereo"
    audio_stream.codec_context.time_base = Fraction(1, audio_sample_rate)
    return audio_stream

def write_video_audio_ltx2(
    video: list[Image.Image],
    audio: torch.Tensor | None,
    output_path: str,
    fps: int = 24,
    audio_sample_rate: int | None = 24000,
) -> None:

    width, height = video[0].size
    container = av.open(output_path, mode="w")
    stream = container.add_stream("libx264", rate=int(fps))
    stream.width = width
    stream.height = height
    stream.pix_fmt = "yuv420p"
    
    if audio is not None:
        if audio_sample_rate is None:
            raise ValueError("audio_sample_rate is required when audio is provided")
        audio_stream = _prepare_audio_stream(container, audio_sample_rate)

    for frame in tqdm(video, total=len(video)):
        frame = av.VideoFrame.from_image(frame)
        for packet in stream.encode(frame):
            container.mux(packet)

    # Flush encoder
    for packet in stream.encode():
        container.mux(packet)

    if audio is not None:
        _write_audio(container, audio_stream, audio, audio_sample_rate)

    container.close()