def test_capabilities(self):
# audio
capabilities = RTCRtpSender.getCapabilities('audio')
self.assertTrue(isinstance(capabilities, RTCRtpCapabilities))
self.assertEqual(capabilities.codecs, [
RTCRtpCodecCapability(
mimeType='audio/opus', clockRate=48000, channels=2),
RTCRtpCodecCapability(
mimeType='audio/PCMU', clockRate=8000, channels=1),
RTCRtpCodecCapability(
mimeType='audio/PCMA', clockRate=8000, channels=1),
])
self.assertEqual(capabilities.headerExtensions, [
RTCRtpHeaderExtensionCapability(
uri='urn:ietf:params:rtp-hdrext:sdes:mid'),
])
# video
capabilities = RTCRtpSender.getCapabilities('video')
self.assertTrue(isinstance(capabilities, RTCRtpCapabilities))
self.assertEqual(capabilities.codecs, [
RTCRtpCodecCapability(mimeType='video/VP8', clockRate=90000),
RTCRtpCodecCapability(mimeType='video/rtx', clockRate=90000),
RTCRtpCodecCapability(mimeType='video/H264',
clockRate=90000,
parameters=OrderedDict(
[('packetization-mode', '1'),
('level-asymmetry-allowed', '1'),
('profile-level-id', '42001f')])),
RTCRtpCodecCapability(mimeType='video/H264',
clockRate=90000,
parameters=OrderedDict(
[('packetization-mode', '1'),
('level-asymmetry-allowed', '1'),
('profile-level-id', '42e01f')])),
])
self.assertEqual(
capabilities.headerExtensions,
[
RTCRtpHeaderExtensionCapability(
uri='urn:ietf:params:rtp-hdrext:sdes:mid'),
RTCRtpHeaderExtensionCapability(
uri=
'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time'
), # noqa
])
# bogus
capabilities = RTCRtpSender.getCapabilities('bogus')
self.assertIsNone(capabilities)
def test_codec_preferences(self):
transceiver = RTCRtpTransceiver("audio", None, None)
self.assertEqual(transceiver._preferred_codecs, [])
# set empty preferences
transceiver.setCodecPreferences([])
self.assertEqual(transceiver._preferred_codecs, [])
# set single codec
transceiver.setCodecPreferences(
[RTCRtpCodecCapability(mimeType="audio/PCMU", clockRate=8000, channels=1)]
)
self.assertEqual(
transceiver._preferred_codecs,
[RTCRtpCodecCapability(mimeType="audio/PCMU", clockRate=8000, channels=1)],
)
# set single codec (duplicated)
transceiver.setCodecPreferences(
[
RTCRtpCodecCapability(
mimeType="audio/PCMU", clockRate=8000, channels=1
),
RTCRtpCodecCapability(
mimeType="audio/PCMU", clockRate=8000, channels=1
),
]
)
self.assertEqual(
transceiver._preferred_codecs,
[RTCRtpCodecCapability(mimeType="audio/PCMU", clockRate=8000, channels=1)],
)
# set single codec (invalid)
with self.assertRaises(ValueError) as cm:
transceiver.setCodecPreferences(
[
RTCRtpCodecCapability(
mimeType="audio/bogus", clockRate=8000, channels=1
)
]
)
self.assertEqual(str(cm.exception), "Codec is not in capabilities")
gi.require_version('Gst', '1.0')
from gi.repository import Gst
RATE = 15
ROOT = os.path.dirname(__file__)
camera = None
capabilities = RTCRtpSender.getCapabilities("video")
codec_parameters = OrderedDict([
("packetization-mode", "1"),
("level-asymmetry-allowed", "1"),
("profile-level-id", "42001f"),
])
h264_capability = RTCRtpCodecCapability(mimeType="video/H264",
clockRate=90000,
channels=None,
parameters=codec_parameters)
preferences = [h264_capability]
rtsp_input = None
webcam_input = None
class GstH264Camera:
WEBCAM_PIPELINE = "v4l2src device=/dev/{} ! video/x-h264,width=1280,height=720,framerate={}/1 ! queue ! appsink emit-signals=True name=h264_sink"
RTSP_PIPELINE = "rtspsrc location={} latency=0 ! rtph264depay ! queue ! h264parse ! video/x-h264,alignment=nal,stream-format=byte-stream ! appsink emit-signals=True name=h264_sink"
#RTSP_PIPELINE = "rtspsrc location={} latency=0 ! rtph264depay ! queue ! video/x-h264,alignment=nal,stream-format=byte-stream ! appsink emit-signals=True name=h264_sink"
def __init__(self, rate, output, rtsp_input=None, webcam_input=None):
if rtsp_input is not None and webcam_input is not None:
raise Exception("Only one inupt can be used at once")
def test_capabilities(self):
# audio
capabilities = RTCRtpSender.getCapabilities("audio")
self.assertTrue(isinstance(capabilities, RTCRtpCapabilities))
self.assertEqual(
capabilities.codecs,
[
RTCRtpCodecCapability(
mimeType="audio/opus", clockRate=48000, channels=2),
RTCRtpCodecCapability(
mimeType="audio/PCMU", clockRate=8000, channels=1),
RTCRtpCodecCapability(
mimeType="audio/PCMA", clockRate=8000, channels=1),
],
)
self.assertEqual(
capabilities.headerExtensions,
[
RTCRtpHeaderExtensionCapability(
uri="urn:ietf:params:rtp-hdrext:sdes:mid")
],
)
# video
capabilities = RTCRtpSender.getCapabilities("video")
self.assertTrue(isinstance(capabilities, RTCRtpCapabilities))
self.assertEqual(
capabilities.codecs,
[
RTCRtpCodecCapability(mimeType="video/VP8", clockRate=90000),
RTCRtpCodecCapability(mimeType="video/rtx", clockRate=90000),
RTCRtpCodecCapability(
mimeType="video/H264",
clockRate=90000,
parameters=OrderedDict([
("packetization-mode", "1"),
("level-asymmetry-allowed", "1"),
("profile-level-id", "42001f"),
]),
),
RTCRtpCodecCapability(
mimeType="video/H264",
clockRate=90000,
parameters=OrderedDict([
("packetization-mode", "1"),
("level-asymmetry-allowed", "1"),
("profile-level-id", "42e01f"),
]),
),
],
)
self.assertEqual(
capabilities.headerExtensions,
[
RTCRtpHeaderExtensionCapability(
uri="urn:ietf:params:rtp-hdrext:sdes:mid"),
RTCRtpHeaderExtensionCapability(
uri=
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
),
],
)
# bogus
with self.assertRaises(ValueError):
RTCRtpSender.getCapabilities("bogus")
