2bcf87b980
* Change the API boundary between stream and camera Shift more of the stream lifecycle management to the camera. The motivation is to support stream urls that expire giving the camera the ability to change the stream once it is created. * Document stream lifecycle and simplify stream/camera interaction * Reorder create_stream function to reduce diffs * Increase test coverage for camera_sdm.py * Fix ffmpeg typo. * Add a stream identifier for each stream, managed by camera * Remove stream record service * Update homeassistant/components/stream/__init__.py Co-authored-by: Paulus Schoutsen <paulus@home-assistant.io> * Unroll changes to Stream interface back into camera component * Fix preload stream to actually start the background worker * Reduce unncessary diffs for readability * Remove redundant camera stream start code Co-authored-by: Paulus Schoutsen <paulus@home-assistant.io>
92 lines
3.1 KiB
Python
92 lines
3.1 KiB
Python
"""Collection of test helpers."""
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from fractions import Fraction
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import io
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import av
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import numpy as np
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AUDIO_SAMPLE_RATE = 8000
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def generate_h264_video(container_format="mp4", audio_codec=None):
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"""
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Generate a test video.
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See: http://docs.mikeboers.com/pyav/develop/cookbook/numpy.html
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"""
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def generate_audio_frame(pcm_mulaw=False):
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"""Generate a blank audio frame."""
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if pcm_mulaw:
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audio_frame = av.AudioFrame(format="s16", layout="mono", samples=1)
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audio_bytes = b"\x00\x00"
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else:
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audio_frame = av.AudioFrame(format="dbl", layout="mono", samples=1024)
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audio_bytes = b"\x00\x00\x00\x00\x00\x00\x00\x00" * 1024
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audio_frame.planes[0].update(audio_bytes)
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audio_frame.sample_rate = AUDIO_SAMPLE_RATE
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audio_frame.time_base = Fraction(1, AUDIO_SAMPLE_RATE)
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return audio_frame
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duration = 5
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fps = 24
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total_frames = duration * fps
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output = io.BytesIO()
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output.name = "test.mov" if container_format == "mov" else "test.mp4"
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container = av.open(output, mode="w", format=container_format)
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stream = container.add_stream("libx264", rate=fps)
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stream.width = 480
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stream.height = 320
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stream.pix_fmt = "yuv420p"
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a_packet = None
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last_a_dts = -1
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if audio_codec is not None:
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if audio_codec == "empty": # empty we add a stream but don't mux any audio
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astream = container.add_stream("aac", AUDIO_SAMPLE_RATE)
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else:
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astream = container.add_stream(audio_codec, AUDIO_SAMPLE_RATE)
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# Need to do it multiple times for some reason
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while not a_packet:
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a_packets = astream.encode(
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generate_audio_frame(pcm_mulaw=audio_codec == "pcm_mulaw")
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)
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if a_packets:
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a_packet = a_packets[0]
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for frame_i in range(total_frames):
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img = np.empty((480, 320, 3))
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img[:, :, 0] = 0.5 + 0.5 * np.sin(2 * np.pi * (0 / 3 + frame_i / total_frames))
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img[:, :, 1] = 0.5 + 0.5 * np.sin(2 * np.pi * (1 / 3 + frame_i / total_frames))
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img[:, :, 2] = 0.5 + 0.5 * np.sin(2 * np.pi * (2 / 3 + frame_i / total_frames))
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img = np.round(255 * img).astype(np.uint8)
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img = np.clip(img, 0, 255)
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frame = av.VideoFrame.from_ndarray(img, format="rgb24")
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for packet in stream.encode(frame):
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container.mux(packet)
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if a_packet is not None:
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a_packet.pts = int(frame_i / (fps * a_packet.time_base))
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while a_packet.pts * a_packet.time_base * fps < frame_i + 1:
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a_packet.dts = a_packet.pts
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if (
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a_packet.dts > last_a_dts
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): # avoid writing same dts twice in case of rounding
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container.mux(a_packet)
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last_a_dts = a_packet.dts
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a_packet.pts += a_packet.duration
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# Flush stream
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for packet in stream.encode():
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container.mux(packet)
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# Close the file
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container.close()
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output.seek(0)
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return output
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