"""Provides the worker thread needed for processing streams."""
from fractions import Fraction
import io
import logging

import av

from .const import AUDIO_SAMPLE_RATE
from .core import Segment, StreamBuffer

_LOGGER = logging.getLogger(__name__)


def generate_audio_frame():
    """Generate a blank audio frame."""

    audio_frame = av.AudioFrame(format="dbl", layout="mono", samples=1024)
    # audio_bytes = b''.join(b'\x00\x00\x00\x00\x00\x00\x00\x00'
    #                        for i in range(0, 1024))
    audio_bytes = b"\x00\x00\x00\x00\x00\x00\x00\x00" * 1024
    audio_frame.planes[0].update(audio_bytes)
    audio_frame.sample_rate = AUDIO_SAMPLE_RATE
    audio_frame.time_base = Fraction(1, AUDIO_SAMPLE_RATE)
    return audio_frame


def create_stream_buffer(stream_output, video_stream, audio_frame):
    """Create a new StreamBuffer."""

    a_packet = None
    segment = io.BytesIO()
    output = av.open(segment, mode="w", format=stream_output.format)
    vstream = output.add_stream(template=video_stream)
    # Check if audio is requested
    astream = None
    if stream_output.audio_codec:
        astream = output.add_stream(stream_output.audio_codec, AUDIO_SAMPLE_RATE)
        # Need to do it multiple times for some reason
        while not a_packet:
            a_packets = astream.encode(audio_frame)
            if a_packets:
                a_packet = a_packets[0]
    return (a_packet, StreamBuffer(segment, output, vstream, astream))


def stream_worker(hass, stream, quit_event):
    """Handle consuming streams."""

    container = av.open(stream.source, options=stream.options)
    try:
        video_stream = container.streams.video[0]
    except (KeyError, IndexError):
        _LOGGER.error("Stream has no video")
        return

    audio_frame = generate_audio_frame()

    first_packet = True
    # Holds the buffers for each stream provider
    outputs = {}
    # Keep track of the number of segments we've processed
    sequence = 1
    # Holds the generated silence that needs to be muxed into the output
    audio_packets = {}
    # The presentation timestamp of the first video packet we receive
    first_pts = 0
    # The decoder timestamp of the latest packet we processed
    last_dts = None
    # Keep track of consecutive packets without a dts to detect end of stream.
    last_packet_was_without_dts = False

    while not quit_event.is_set():
        try:
            packet = next(container.demux(video_stream))
            if packet.dts is None:
                if first_packet:
                    continue
                _LOGGER.error("Stream packet without dts detected, skipping...")
                # Allow a single packet without dts before terminating the stream.
                if last_packet_was_without_dts:
                    # If we get a "flushing" packet, the stream is done
                    raise StopIteration("No dts in consecutive packets")
                last_packet_was_without_dts = True
                continue
            last_packet_was_without_dts = False
        except (av.AVError, StopIteration) as ex:
            # End of stream, clear listeners and stop thread
            for fmt, _ in outputs.items():
                hass.loop.call_soon_threadsafe(stream.outputs[fmt].put, None)
            _LOGGER.error("Error demuxing stream: %s", str(ex))
            break

        # Skip non monotonically increasing dts in feed
        if not first_packet and last_dts >= packet.dts:
            continue
        last_dts = packet.dts

        # Reset timestamps from a 0 time base for this stream
        packet.dts -= first_pts
        packet.pts -= first_pts

        # Reset segment on every keyframe
        if packet.is_keyframe:
            # Calculate the segment duration by multiplying the presentation
            # timestamp by the time base, which gets us total seconds.
            # By then dividing by the sequence, we can calculate how long
            # each segment is, assuming the stream starts from 0.
            segment_duration = (packet.pts * packet.time_base) / sequence
            # Save segment to outputs
            for fmt, buffer in outputs.items():
                buffer.output.close()
                del audio_packets[buffer.astream]
                if stream.outputs.get(fmt):
                    hass.loop.call_soon_threadsafe(
                        stream.outputs[fmt].put,
                        Segment(sequence, buffer.segment, segment_duration),
                    )

            # Clear outputs and increment sequence
            outputs = {}
            if not first_packet:
                sequence += 1

            # Initialize outputs
            for stream_output in stream.outputs.values():
                if video_stream.name != stream_output.video_codec:
                    continue

                a_packet, buffer = create_stream_buffer(
                    stream_output, video_stream, audio_frame
                )
                audio_packets[buffer.astream] = a_packet
                outputs[stream_output.name] = buffer

        # First video packet tends to have a weird dts/pts
        if first_packet:
            # If we are attaching to a live stream that does not reset
            # timestamps for us, we need to do it ourselves by recording
            # the first presentation timestamp and subtracting it from
            # subsequent packets we receive.
            if (packet.pts * packet.time_base) > 1:
                first_pts = packet.pts
            packet.dts = 0
            packet.pts = 0
            first_packet = False

        # Store packets on each output
        for buffer in outputs.values():
            # Check if the format requires audio
            if audio_packets.get(buffer.astream):
                a_packet = audio_packets[buffer.astream]
                a_time_base = a_packet.time_base

                # Determine video start timestamp and duration
                video_start = packet.pts * packet.time_base
                video_duration = packet.duration * packet.time_base

                if packet.is_keyframe:
                    # Set first audio packet in sequence to equal video pts
                    a_packet.pts = int(video_start / a_time_base)
                    a_packet.dts = int(video_start / a_time_base)

                # Determine target end timestamp for audio
                target_pts = int((video_start + video_duration) / a_time_base)
                while a_packet.pts < target_pts:
                    # Mux audio packet and adjust points until target hit
                    buffer.output.mux(a_packet)
                    a_packet.pts += a_packet.duration
                    a_packet.dts += a_packet.duration
                    audio_packets[buffer.astream] = a_packet

            # Assign the video packet to the new stream & mux
            packet.stream = buffer.vstream
            buffer.output.mux(packet)

    # Close stream
    buffer.output.close()
    container.close()
