add basic usb input mode functionallity
This commit is contained in:
@@ -44,6 +44,7 @@ from bumble.profiles import bass
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import bumble.device
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import bumble.transport
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import bumble.utils
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import numpy as np # for audio down-mix
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from bumble.device import Host, BIGInfoAdvertisement, AdvertisingChannelMap
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from bumble.audio import io as audio_io
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@@ -326,19 +327,37 @@ class Streamer():
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else:
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logging.warning('Streamer is already running')
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def stop_streaming(self):
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"""Stops the background task if running."""
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if self.is_streaming:
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self.is_streaming = False
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if self.task:
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self.task.cancel() # Cancel the task safely
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self.task = None
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async def stop_streaming(self):
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"""Gracefully stop streaming and release audio devices."""
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if not self.is_streaming and self.task is None:
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return
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# Ask the streaming loop to finish
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self.is_streaming = False
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if self.task is not None:
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self.task.cancel()
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try:
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await self.task
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except asyncio.CancelledError:
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pass
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self.task = None
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# Close audio inputs (await to ensure ALSA devices are released)
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close_tasks = []
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for big in self.bigs.values():
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ai = big.get("audio_input")
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if ai and hasattr(ai, "close"):
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close_tasks.append(ai.close())
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# Remove reference so a fresh one is created next time
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big.pop("audio_input", None)
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if close_tasks:
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await asyncio.gather(*close_tasks, return_exceptions=True)
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async def stream(self):
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bigs = self.bigs
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big_config = self.big_config
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global_config = self.global_config
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# init
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for i, big in enumerate(bigs.values()):
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audio_source = big_config[i].audio_source
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input_format = big_config[i].input_format
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@@ -356,6 +375,7 @@ class Streamer():
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lc3_frame_samples = encoder.get_frame_samples()
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big['pcm_bit_depth'] = 16
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big['lc3_frame_samples'] = lc3_frame_samples
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big['lc3_bytes_per_frame'] = global_config.octets_per_frame
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big['audio_input'] = audio_source
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big['encoder'] = encoder
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big['precoded'] = False
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@@ -371,6 +391,7 @@ class Streamer():
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lc3_frame_samples = encoder.get_frame_samples()
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big['pcm_bit_depth'] = 16
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big['lc3_frame_samples'] = lc3_frame_samples
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big['lc3_bytes_per_frame'] = global_config.octets_per_frame
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big['encoder'] = encoder
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big['precoded'] = False
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@@ -405,14 +426,14 @@ class Streamer():
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pcm_format = await audio_input.open()
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if pcm_format.channels != 1:
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print("Only 1 channels PCM configurations are supported")
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logging.error("Only 1 channels PCM configurations are supported")
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return
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if pcm_format.sample_type == audio_io.PcmFormat.SampleType.INT16:
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pcm_bit_depth = 16
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elif pcm_format.sample_type == audio_io.PcmFormat.SampleType.FLOAT32:
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pcm_bit_depth = None
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else:
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print("Only INT16 and FLOAT32 sample types are supported")
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logging.error("Only INT16 and FLOAT32 sample types are supported")
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return
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encoder = lc3.Encoder(
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frame_duration_us=global_config.frame_duration_us,
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@@ -437,73 +458,137 @@ class Streamer():
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# anything else, e.g. realtime stream from device (bumble)
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else:
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audio_input = await audio_io.create_audio_input(audio_source, input_format)
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audio_input.rewind = big_config[i].loop
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pcm_format = await audio_input.open()
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#try:
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if pcm_format.channels != 1:
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print("Only 1 channels PCM configurations are supported")
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return
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if pcm_format.sample_type == audio_io.PcmFormat.SampleType.INT16:
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pcm_bit_depth = 16
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elif pcm_format.sample_type == audio_io.PcmFormat.SampleType.FLOAT32:
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pcm_bit_depth = None
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else:
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print("Only INT16 and FLOAT32 sample types are supported")
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return
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encoder = lc3.Encoder(
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frame_duration_us=global_config.frame_duration_us,
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sample_rate_hz=global_config.auracast_sampling_rate_hz,
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num_channels=1,
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input_sample_rate_hz=pcm_format.sample_rate,
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)
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lc3_frame_samples = encoder.get_frame_samples() # number of the pcm samples per lc3 frame
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big['pcm_bit_depth'] = pcm_bit_depth
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big['lc3_frame_samples'] = lc3_frame_samples
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# Store early so stop_streaming can close even if open() fails
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big['audio_input'] = audio_input
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big['encoder'] = encoder
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big['precoded'] = False
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# SoundDeviceAudioInput (used for `mic:<device>` captures) has no `.rewind`.
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if hasattr(audio_input, "rewind"):
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audio_input.rewind = big_config[i].loop
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# Need for coded an uncoded audio
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lc3_frame_size = global_config.octets_per_frame #encoder.get_frame_bytes(bitrate)
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lc3_bytes_per_frame = lc3_frame_size #* 2 #multiplied by number of channels
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big['lc3_bytes_per_frame'] = lc3_bytes_per_frame
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# Retry logic – ALSA sometimes keeps the device busy for a short time after the
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# previous stream has closed. Handle PortAudioError -9985 with back-off retries.
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import sounddevice as _sd
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max_attempts = 3
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for attempt in range(1, max_attempts + 1):
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try:
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pcm_format = await audio_input.open()
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break # success
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except _sd.PortAudioError as err:
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# -9985 == paDeviceUnavailable
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logging.error('Could not open audio device %s with error %s', audio_source, err)
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code = None
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if hasattr(err, 'errno'):
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code = err.errno
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elif len(err.args) > 1 and isinstance(err.args[1], int):
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code = err.args[1]
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if code == -9985 and attempt < max_attempts:
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backoff_ms = 200 * attempt
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logging.warning("PortAudio device busy (attempt %d/%d). Retrying in %.1f ms…", attempt, max_attempts, backoff_ms)
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# ensure device handle and PortAudio context are closed before retrying
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try:
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if hasattr(audio_input, "aclose"):
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await audio_input.aclose()
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elif hasattr(audio_input, "close"):
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audio_input.close()
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except Exception:
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pass
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# Fully terminate PortAudio to drop lingering handles (sounddevice quirk)
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if hasattr(_sd, "_terminate"):
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try:
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_sd._terminate()
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except Exception:
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pass
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# Small pause then re-initialize PortAudio
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await asyncio.sleep(0.1)
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if hasattr(_sd, "_initialize"):
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try:
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_sd._initialize()
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except Exception:
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pass
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# TODO: Maybe do some pre buffering so the stream is stable from the beginning. One half iso queue would be appropriate
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logging.info("Streaming audio...")
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bigs = self.bigs
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self.is_streaming = True
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# One streamer fits all
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while self.is_streaming:
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stream_finished = [False for _ in range(len(bigs))]
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for i, big in enumerate(bigs.values()):
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# Back-off before next attempt
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await asyncio.sleep(backoff_ms / 1000)
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# Recreate audio_input fresh for next attempt
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audio_input = await audio_io.create_audio_input(audio_source, input_format)
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continue
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# Other errors or final attempt – re-raise so caller can abort gracefully
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raise
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else:
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# Loop exhausted without break
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logging.error("Unable to open audio device after %d attempts – giving up", max_attempts)
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return
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if big['precoded']:# everything was already lc3 coded beforehand
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lc3_frame = bytes(
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itertools.islice(big['lc3_frames'], big['lc3_bytes_per_frame'])
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if pcm_format.channels != 1:
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logging.info("Input device provides %d channels – will down-mix to mono for LC3", pcm_format.channels)
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if pcm_format.sample_type == audio_io.PcmFormat.SampleType.INT16:
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pcm_bit_depth = 16
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elif pcm_format.sample_type == audio_io.PcmFormat.SampleType.FLOAT32:
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pcm_bit_depth = None
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else:
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logging.error("Only INT16 and FLOAT32 sample types are supported")
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return
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encoder = lc3.Encoder(
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frame_duration_us=global_config.frame_duration_us,
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sample_rate_hz=global_config.auracast_sampling_rate_hz,
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num_channels=1,
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input_sample_rate_hz=pcm_format.sample_rate,
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)
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lc3_frame_samples = encoder.get_frame_samples() # number of the pcm samples per lc3 frame
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big['pcm_bit_depth'] = pcm_bit_depth
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big['channels'] = pcm_format.channels
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big['lc3_frame_samples'] = lc3_frame_samples
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big['lc3_bytes_per_frame'] = global_config.octets_per_frame
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big['audio_input'] = audio_input
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big['encoder'] = encoder
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big['precoded'] = False
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logging.info("Streaming audio...")
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bigs = self.bigs
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self.is_streaming = True
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# One streamer fits all
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while self.is_streaming:
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stream_finished = [False for _ in range(len(bigs))]
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for i, big in enumerate(bigs.values()):
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if big['precoded']:# everything was already lc3 coded beforehand
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lc3_frame = bytes(
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itertools.islice(big['lc3_frames'], big['lc3_bytes_per_frame'])
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)
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if lc3_frame == b'': # Not all streams may stop at the same time
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stream_finished[i] = True
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continue
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else: # code lc3 on the fly
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pcm_frame = await anext(big['audio_input'].frames(big['lc3_frame_samples']), None)
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if pcm_frame is None: # Not all streams may stop at the same time
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stream_finished[i] = True
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continue
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# Down-mix multi-channel PCM to mono for LC3 encoder if needed
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if big.get('channels', 1) > 1:
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if isinstance(pcm_frame, np.ndarray):
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if pcm_frame.ndim > 1:
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mono = pcm_frame.mean(axis=1).astype(pcm_frame.dtype)
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pcm_frame = mono
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else:
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# Convert raw bytes to numpy, average channels, convert back
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dtype = np.int16 if big['pcm_bit_depth'] == 16 else np.float32
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samples = np.frombuffer(pcm_frame, dtype=dtype)
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samples = samples.reshape(-1, big['channels']).mean(axis=1)
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pcm_frame = samples.astype(dtype).tobytes()
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lc3_frame = big['encoder'].encode(
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pcm_frame, num_bytes=big['lc3_bytes_per_frame'], bit_depth=big['pcm_bit_depth']
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)
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if lc3_frame == b'': # Not all streams may stop at the same time
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stream_finished[i] = True
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continue
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else: # code lc3 on the fly
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pcm_frame = await anext(big['audio_input'].frames(big['lc3_frame_samples']), None)
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await big['iso_queue'].write(lc3_frame)
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if pcm_frame is None: # Not all streams may stop at the same time
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stream_finished[i] = True
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continue
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lc3_frame = big['encoder'].encode(
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pcm_frame, num_bytes=big['lc3_bytes_per_frame'], bit_depth=big['pcm_bit_depth']
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)
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await big['iso_queue'].write(lc3_frame)
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if all(stream_finished): # Take into account that multiple files have different lengths
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logging.info('All streams finished, stopping streamer')
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self.is_streaming = False
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break
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if all(stream_finished): # Take into account that multiple files have different lengths
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logging.info('All streams finished, stopping streamer')
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self.is_streaming = False
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break
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# -----------------------------------------------------------------------------
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