Quick fix for audio glitches in streaming. TODO fix latency.
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@@ -88,7 +88,7 @@ class ModSoundDeviceAudioInput(audio_io.SoundDeviceAudioInput):
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self.counter=0
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self.max_avail=0
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self.logfile_name="available_samples.txt"
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self.blocksize = 120
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self.blocksize = 240
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if os.path.exists(self.logfile_name):
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os.remove(self.logfile_name)
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@@ -99,7 +99,7 @@ class ModSoundDeviceAudioInput(audio_io.SoundDeviceAudioInput):
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channels=self._pcm_format.channels,
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dtype='int16',
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blocksize=self.blocksize,
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latency=0.004,
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latency=0.020,
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)
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self._stream.start()
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@@ -114,7 +114,7 @@ class ModSoundDeviceAudioInput(audio_io.SoundDeviceAudioInput):
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"""Read PCM samples from the stream."""
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#if self.counter % 50 == 0:
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frame_size = frame_size + 1 # consume samples a little faster to avoid latency akkumulation
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# frame_size = frame_size + 1 # consume samples a little faster to avoid latency akkumulation
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pcm_buffer, overflowed = self._stream.read(frame_size)
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if overflowed:
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@@ -385,8 +385,7 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
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if input_device_name in dante_channels:
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conf.auracast_sampling_rate_hz = 48000
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# Also update octets per frame for 48000 Hz
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if conf.octets_per_frame == 60: # 24000 Hz setting
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conf.octets_per_frame = 120 # 48000 Hz setting
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conf.octets_per_frame = 120 # 48000 Hz setting
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conf.qos_config.max_transport_latency_ms = int(conf.qos_config.number_of_retransmissions) * 10 + 3
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@@ -49,8 +49,8 @@ pcm.dante_asrc_shared6 {
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rate 48000
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format S16_LE
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period_size 256
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buffer_size 1024
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period_size 240
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buffer_size 960
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}
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hint { show on ; description "DEP RX (via ASRC) shared 6ch (loopback+dsnoop)" }
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