Revert "feat: add gain control for USB and webapp microphone inputs with persistence - to be tested"
This reverts commit 0bf87c85b7.
This commit is contained in:
@@ -51,7 +51,6 @@ class AuracastBigConfig(BaseModel):
|
||||
program_info: str = 'Some Announcements'
|
||||
audio_source: str = 'file:./auracast/announcement_48_10_96000_en.wav'
|
||||
input_format: str = 'auto'
|
||||
input_gain: float | None = None # Parsed from audio_source for device inputs
|
||||
loop: bool = True
|
||||
precode_wav: bool = False
|
||||
iso_que_len: int = 64
|
||||
|
||||
@@ -452,135 +452,97 @@ class Streamer():
|
||||
lc3_frames = itertools.cycle(lc3_frames)
|
||||
big['lc3_frames'] = lc3_frames
|
||||
|
||||
# anything else, e.g. realtime stream from device (bumble) or non-precoded file
|
||||
# anything else, e.g. realtime stream from device (bumble)
|
||||
else:
|
||||
current_big_config = self.big_config[i]
|
||||
audio_source_str = str(current_big_config.audio_source) # Ensure string type
|
||||
input_format_str = current_big_config.input_format
|
||||
input_gain_val = current_big_config.input_gain
|
||||
|
||||
audio_filter_for_create = None
|
||||
effective_audio_source_for_create = audio_source_str
|
||||
|
||||
if audio_source_str.startswith('device:'):
|
||||
parts = audio_source_str.split(':', 1)
|
||||
if len(parts) > 1:
|
||||
device_specifier_with_potential_gain = parts[1]
|
||||
pure_device_name = device_specifier_with_potential_gain.split(',', 1)[0]
|
||||
effective_audio_source_for_create = f"device:{pure_device_name}"
|
||||
|
||||
gain_to_apply = input_gain_val if input_gain_val is not None else 1.0
|
||||
if abs(gain_to_apply - 1.0) > 0.01:
|
||||
audio_filter_for_create = f"volume={gain_to_apply:.2f}"
|
||||
logger.info(f"Applying FFmpeg volume filter for {effective_audio_source_for_create}: {audio_filter_for_create}")
|
||||
elif audio_source_str.startswith('file:'):
|
||||
gain_to_apply = input_gain_val if input_gain_val is not None else 1.0
|
||||
if abs(gain_to_apply - 1.0) > 0.01:
|
||||
audio_filter_for_create = f"volume={gain_to_apply:.2f}"
|
||||
logger.info(f"Applying FFmpeg volume filter for {audio_source_str}: {audio_filter_for_create}")
|
||||
|
||||
# Prepare the source string, potentially with an FFmpeg filter
|
||||
final_audio_source_spec = effective_audio_source_for_create
|
||||
if current_big_config.input_gain is not None and input_format_str == 'ffmpeg': # Apply gain only if ffmpeg is used
|
||||
audio_filter_value = f"volume={current_big_config.input_gain:.2f}"
|
||||
logging.info(f"Applying FFmpeg volume filter for {effective_audio_source_for_create}: {audio_filter_value}")
|
||||
# Append 'af' (audio filter) option to the source spec for FFmpeg
|
||||
if '?' in final_audio_source_spec: # if there are already ffmpeg options (e.g. sample_rate)
|
||||
final_audio_source_spec = f"{final_audio_source_spec}&af={audio_filter_value}"
|
||||
else: # if this is the first ffmpeg option
|
||||
final_audio_source_spec = f"{final_audio_source_spec},af={audio_filter_value}"
|
||||
|
||||
# Initial creation of audio_input
|
||||
audio_input = await audio_io.create_audio_input(
|
||||
final_audio_source_spec,
|
||||
input_format=input_format_str
|
||||
)
|
||||
big['audio_input'] = audio_input # Store early for potential cleanup
|
||||
|
||||
audio_input = await audio_io.create_audio_input(audio_source, input_format)
|
||||
# Store early so stop_streaming can close even if open() fails
|
||||
big['audio_input'] = audio_input
|
||||
# SoundDeviceAudioInput (used for `mic:<device>` captures) has no `.rewind`.
|
||||
if hasattr(audio_input, "rewind"):
|
||||
audio_input.rewind = current_big_config.loop
|
||||
audio_input.rewind = big_config[i].loop
|
||||
|
||||
# Retry logic – ALSA sometimes keeps the device busy for a short time after the
|
||||
# previous stream has closed. Handle PortAudioError -9985 with back-off retries.
|
||||
import sounddevice as _sd
|
||||
max_attempts = 3
|
||||
pcm_format = None # Initialize pcm_format
|
||||
for attempt in range(1, max_attempts + 1):
|
||||
try:
|
||||
logging.info(f"Attempting to open audio input: {effective_audio_source_for_create} (attempt {attempt})")
|
||||
pcm_format = await audio_input.open()
|
||||
logging.info(f"Successfully opened audio input: {effective_audio_source_for_create}, PCM Format: {pcm_format}")
|
||||
break # success
|
||||
except _sd.PortAudioError as err:
|
||||
logging.error('Could not open audio device %s with error %s (attempt %d/%d)', effective_audio_source_for_create, err, attempt, max_attempts)
|
||||
code = getattr(err, 'errno', None) or (err.args[1] if len(err.args) > 1 and isinstance(err.args[1], int) else None)
|
||||
if code == -9985 and attempt < max_attempts: # paDeviceUnavailable
|
||||
backoff_ms = (2 ** (attempt - 1)) * 100 # exponential backoff
|
||||
logging.warning("PortAudio device busy. Retrying in %.1f ms…", backoff_ms)
|
||||
# -9985 == paDeviceUnavailable
|
||||
logging.error('Could not open audio device %s with error %s', audio_source, err)
|
||||
code = None
|
||||
if hasattr(err, 'errno'):
|
||||
code = err.errno
|
||||
elif len(err.args) > 1 and isinstance(err.args[1], int):
|
||||
code = err.args[1]
|
||||
if code == -9985 and attempt < max_attempts:
|
||||
backoff_ms = 200 * attempt
|
||||
logging.warning("PortAudio device busy (attempt %d/%d). Retrying in %.1f ms…", attempt, max_attempts, backoff_ms)
|
||||
# ensure device handle and PortAudio context are closed before retrying
|
||||
try:
|
||||
if hasattr(audio_input, "aclose"): await audio_input.aclose()
|
||||
elif hasattr(audio_input, "close"): audio_input.close()
|
||||
except Exception as close_err: logging.debug(f"Error closing audio_input during retry: {close_err}")
|
||||
if hasattr(_sd, "_terminate"): # sounddevice specific cleanup
|
||||
try: _sd._terminate()
|
||||
except Exception as term_err: logging.debug(f"Error terminating PortAudio: {term_err}")
|
||||
if hasattr(audio_input, "aclose"):
|
||||
await audio_input.aclose()
|
||||
elif hasattr(audio_input, "close"):
|
||||
audio_input.close()
|
||||
except Exception:
|
||||
pass
|
||||
# Fully terminate PortAudio to drop lingering handles (sounddevice quirk)
|
||||
if hasattr(_sd, "_terminate"):
|
||||
try:
|
||||
_sd._terminate()
|
||||
except Exception:
|
||||
pass
|
||||
# Small pause then re-initialize PortAudio
|
||||
await asyncio.sleep(0.1)
|
||||
if hasattr(_sd, "_initialize"): # sounddevice specific reinit
|
||||
try: _sd._initialize()
|
||||
except Exception as init_err: logging.debug(f"Error initializing PortAudio: {init_err}")
|
||||
await asyncio.sleep(backoff_ms / 1000)
|
||||
# Recreate audio_input for next attempt, using the potentially modified source spec
|
||||
audio_input = await audio_io.create_audio_input(
|
||||
final_audio_source_spec, # Use the spec that includes the filter if applicable
|
||||
input_format=input_format_str
|
||||
)
|
||||
big['audio_input'] = audio_input # Update stored reference
|
||||
if hasattr(audio_input, "rewind"):
|
||||
audio_input.rewind = current_big_config.loop
|
||||
continue
|
||||
raise # Re-raise if not paDeviceUnavailable or max_attempts reached
|
||||
except Exception as e:
|
||||
logging.error(f"Unexpected error opening audio device {effective_audio_source_for_create}: {e}")
|
||||
raise # Re-raise other unexpected errors
|
||||
else: # else for 'for' loop: if loop finished without break
|
||||
logging.error("Unable to open audio device '%s' after %d attempts – giving up.", effective_audio_source_for_create, max_attempts)
|
||||
return # Or handle error more gracefully, e.g. mark BIG as inactive
|
||||
if hasattr(_sd, "_initialize"):
|
||||
try:
|
||||
_sd._initialize()
|
||||
except Exception:
|
||||
pass
|
||||
|
||||
# Proceed with encoder setup if pcm_format was obtained
|
||||
if not pcm_format:
|
||||
logging.error(f"Failed to obtain PCM format for {effective_audio_source_for_create}. Cannot set up encoder.")
|
||||
# Back-off before next attempt
|
||||
await asyncio.sleep(backoff_ms / 1000)
|
||||
# Recreate audio_input fresh for next attempt
|
||||
audio_input = await audio_io.create_audio_input(audio_source, input_format)
|
||||
continue
|
||||
# Other errors or final attempt – re-raise so caller can abort gracefully
|
||||
raise
|
||||
else:
|
||||
# Loop exhausted without break
|
||||
logging.error("Unable to open audio device after %d attempts – giving up", max_attempts)
|
||||
return
|
||||
|
||||
if pcm_format.channels != 1:
|
||||
logging.info("Input device '%s' provides %d channels – will down-mix to mono for LC3", effective_audio_source_for_create, pcm_format.channels)
|
||||
# Downmixing is typically handled by FFmpeg if channels > 1 and output is mono
|
||||
# For LC3, we always want mono, so this is informational.
|
||||
|
||||
# Determine pcm_bit_depth for encoder based on pcm_format.sample_type
|
||||
if pcm_format.sample_type == audio_io.PcmFormat.SampleType.INT16:
|
||||
pcm_bit_depth = 16
|
||||
elif pcm_format.sample_type == audio_io.PcmFormat.SampleType.FLOAT32:
|
||||
pcm_bit_depth = None # LC3 encoder can handle float32 directly
|
||||
else:
|
||||
logging.error("Unsupported PCM sample type: %s for %s. Only INT16 and FLOAT32 are supported.", pcm_format.sample_type, effective_audio_source_for_create)
|
||||
return
|
||||
logging.info("Input device provides %d channels – will down-mix to mono for LC3", pcm_format.channels)
|
||||
if pcm_format.sample_type == audio_io.PcmFormat.SampleType.INT16:
|
||||
pcm_bit_depth = 16
|
||||
elif pcm_format.sample_type == audio_io.PcmFormat.SampleType.FLOAT32:
|
||||
pcm_bit_depth = None
|
||||
else:
|
||||
logging.error("Only INT16 and FLOAT32 sample types are supported")
|
||||
return
|
||||
encoder = lc3.Encoder(
|
||||
frame_duration_us=global_config.frame_duration_us,
|
||||
sample_rate_hz=global_config.auracast_sampling_rate_hz,
|
||||
num_channels=1,
|
||||
input_sample_rate_hz=pcm_format.sample_rate,
|
||||
)
|
||||
lc3_frame_samples = encoder.get_frame_samples() # number of the pcm samples per lc3 frame
|
||||
|
||||
encoder = lc3.Encoder(
|
||||
frame_duration_us=self.global_config.frame_duration_us,
|
||||
sample_rate_hz=self.global_config.auracast_sampling_rate_hz,
|
||||
num_channels=1, # LC3 is mono
|
||||
input_sample_rate_hz=pcm_format.sample_rate,
|
||||
)
|
||||
lc3_frame_samples = encoder.get_frame_samples()
|
||||
big['pcm_bit_depth'] = pcm_bit_depth
|
||||
big['lc3_frame_samples'] = lc3_frame_samples
|
||||
big['lc3_bytes_per_frame'] = self.global_config.octets_per_frame
|
||||
big['encoder'] = encoder
|
||||
big['precoded'] = False
|
||||
big['pcm_bit_depth'] = pcm_bit_depth
|
||||
big['channels'] = pcm_format.channels
|
||||
big['lc3_frame_samples'] = lc3_frame_samples
|
||||
big['lc3_bytes_per_frame'] = global_config.octets_per_frame
|
||||
big['audio_input'] = audio_input
|
||||
big['encoder'] = encoder
|
||||
big['precoded'] = False
|
||||
|
||||
|
||||
logging.info("Streaming audio...")
|
||||
bigs = self.bigs
|
||||
self.is_streaming = True
|
||||
logging.info("Entering main streaming loop...")
|
||||
# One streamer fits all
|
||||
while self.is_streaming:
|
||||
stream_finished = [False for _ in range(len(bigs))]
|
||||
@@ -595,9 +557,7 @@ class Streamer():
|
||||
stream_finished[i] = True
|
||||
continue
|
||||
else: # code lc3 on the fly
|
||||
logging.debug(f"BIG {i} ({big.get('name', 'N/A')}): Attempting to read pcm_frame.")
|
||||
pcm_frame = await anext(big['audio_input'].frames(big['lc3_frame_samples']), None)
|
||||
logging.debug(f"BIG {i} ({big.get('name', 'N/A')}): Read pcm_frame: {'None' if pcm_frame is None else f'type {type(pcm_frame)}, len {len(pcm_frame)} bytes' if isinstance(pcm_frame, bytes) else f'type {type(pcm_frame)}, shape {pcm_frame.shape}' if hasattr(pcm_frame, 'shape') else f'type {type(pcm_frame)}'}")
|
||||
|
||||
if pcm_frame is None: # Not all streams may stop at the same time
|
||||
stream_finished[i] = True
|
||||
|
||||
@@ -14,27 +14,14 @@ PTIME = 40
|
||||
BACKEND_URL = "http://localhost:5000"
|
||||
|
||||
# Try loading persisted settings from backend
|
||||
# This is the correct place to define saved_settings before it's used for defaults
|
||||
saved_settings = {}
|
||||
try:
|
||||
resp = requests.get(f"{BACKEND_URL}/status", timeout=1)
|
||||
if resp.status_code == 200:
|
||||
saved_settings = resp.json()
|
||||
except Exception:
|
||||
# If backend is not available or error, saved_settings will be empty dict
|
||||
# Defaults will be used for gain values in this case.
|
||||
saved_settings = {}
|
||||
|
||||
# Initialize gain session states
|
||||
# This must come AFTER saved_settings is populated.
|
||||
default_webapp_gain = float(saved_settings.get('webapp_mic_gain', 1.0))
|
||||
if 'webapp_mic_gain' not in st.session_state:
|
||||
st.session_state.webapp_mic_gain = default_webapp_gain
|
||||
|
||||
default_usb_gain = float(saved_settings.get('usb_mic_gain', 1.0))
|
||||
if 'usb_mic_gain' not in st.session_state:
|
||||
st.session_state.usb_mic_gain = default_usb_gain
|
||||
|
||||
st.title("🎙️ Auracast Audio Mode Control")
|
||||
|
||||
# Audio mode selection with persisted default
|
||||
@@ -66,12 +53,9 @@ if audio_mode in ["Webapp", "USB"]:
|
||||
language = st.text_input("Language (ISO 639-3)", value=default_lang)
|
||||
# Gain slider for Webapp mode
|
||||
if audio_mode == "Webapp":
|
||||
st.session_state.webapp_mic_gain = st.slider(
|
||||
"Microphone Gain", 0.0, 4.0, st.session_state.webapp_mic_gain, 0.1,
|
||||
help="Adjust microphone volume sent to Auracast (applied by browser)"
|
||||
)
|
||||
# For USB mode, gain slider is defined below.
|
||||
# The variable 'mic_gain' for JS is sourced from st.session_state.webapp_mic_gain within Webapp mode logic.
|
||||
mic_gain = st.slider("Microphone Gain", 0.0, 4.0, 1.0, 0.1, help="Adjust microphone volume sent to Auracast")
|
||||
else:
|
||||
mic_gain = 1.0
|
||||
|
||||
# Input device selection for USB mode
|
||||
if audio_mode == "USB":
|
||||
@@ -103,16 +87,6 @@ if audio_mode in ["Webapp", "USB"]:
|
||||
st.rerun()
|
||||
# We send only the numeric/card identifier (before :) or 'default'
|
||||
input_device = selected_option.split(":", 1)[0] if ":" in selected_option else selected_option
|
||||
|
||||
# USB Microphone Gain Slider
|
||||
st.session_state.usb_mic_gain = st.slider(
|
||||
"Microphone Gain (USB)",
|
||||
min_value=0.0,
|
||||
max_value=4.0,
|
||||
value=st.session_state.usb_mic_gain, # Use session state value
|
||||
step=0.1,
|
||||
help="Adjust microphone volume for USB input (applied by server)"
|
||||
)
|
||||
else:
|
||||
input_device = None
|
||||
start_stream = st.button("Start Auracast")
|
||||
@@ -122,7 +96,7 @@ if audio_mode in ["Webapp", "USB"]:
|
||||
if audio_mode == "Webapp" and st.session_state.get('stream_started'):
|
||||
update_js = f"""
|
||||
<script>
|
||||
if (window.gainNode) {{ window.gainNode.gain.value = {st.session_state.webapp_mic_gain}; }}
|
||||
if (window.gainNode) {{ window.gainNode.gain.value = {mic_gain}; }}
|
||||
</script>
|
||||
"""
|
||||
st.components.v1.html(update_js, height=0)
|
||||
@@ -148,17 +122,6 @@ if audio_mode in ["Webapp", "USB"]:
|
||||
import time; time.sleep(1)
|
||||
# Prepare config using the model (do NOT send qos_config, only relevant fields)
|
||||
q = quality_map[quality]
|
||||
|
||||
# Determine audio_source based on mode and gain settings
|
||||
if audio_mode == "USB":
|
||||
current_usb_gain = st.session_state.get('usb_mic_gain', 1.0) # Use .get for safety
|
||||
audio_source_str = f"device:{input_device},gain={current_usb_gain}"
|
||||
elif audio_mode == "Webapp":
|
||||
audio_source_str = "webrtc"
|
||||
# Webapp gain is handled client-side by JS using st.session_state.webapp_mic_gain
|
||||
else: # Assuming a 'network' mode or other future modes
|
||||
audio_source_str = "network" # Default or handle other modes
|
||||
|
||||
config = auracast_config.AuracastConfigGroup(
|
||||
auracast_sampling_rate_hz=q['rate'],
|
||||
octets_per_frame=q['octets'],
|
||||
@@ -168,7 +131,11 @@ if audio_mode in ["Webapp", "USB"]:
|
||||
name=stream_name,
|
||||
program_info=f"{stream_name} {quality}",
|
||||
language=language,
|
||||
audio_source=audio_source_str, # Use the constructed string
|
||||
audio_source=(
|
||||
f"device:{input_device}" if audio_mode == "USB" else (
|
||||
"webrtc" if audio_mode == "Webapp" else "network"
|
||||
)
|
||||
),
|
||||
input_format=(f"int16le,{q['rate']},1" if audio_mode == "USB" else "auto"),
|
||||
iso_que_len=1, # TODO: this should be way less to decrease delay
|
||||
sampling_frequency=q['rate'],
|
||||
@@ -193,7 +160,7 @@ if audio_mode in ["Webapp", "USB"]:
|
||||
(async () => {{
|
||||
if (window.webrtc_started) return; // Prevent re-init on rerun
|
||||
window.webrtc_started = true;
|
||||
const GAIN_VALUE = {st.session_state.webapp_mic_gain};
|
||||
const GAIN_VALUE = {mic_gain};
|
||||
const pc = new RTCPeerConnection(); // No STUN needed for localhost
|
||||
const micStream = await navigator.mediaDevices.getUserMedia({{audio:true}});
|
||||
// Create Web Audio gain processing
|
||||
|
||||
@@ -4,7 +4,7 @@ import logging as log
|
||||
import uuid
|
||||
import json
|
||||
import sys
|
||||
from datetime import datetime, timezone
|
||||
from datetime import datetime
|
||||
import asyncio
|
||||
import numpy as np
|
||||
from pydantic import BaseModel
|
||||
@@ -88,58 +88,25 @@ async def initialize(conf: auracast_config.AuracastConfigGroup):
|
||||
|
||||
# initialize the streams dict
|
||||
# persist stream settings for later retrieval
|
||||
# Derive audio_mode from first BIG audio_source and parse gain for all device sources
|
||||
audio_mode_persist = 'Network' # Default
|
||||
input_device_persist = None # Default for saving settings
|
||||
|
||||
if conf.bigs:
|
||||
first_big = conf.bigs[0]
|
||||
# Determine audio_mode for saving settings based on the first BIG
|
||||
if first_big.audio_source.startswith('device:'):
|
||||
audio_mode_persist = 'USB'
|
||||
# For saving settings, just get the device ID part from the first BIG
|
||||
device_id_part = first_big.audio_source.split(':', 1)[1].split(',', 1)[0]
|
||||
input_device_persist = device_id_part
|
||||
elif first_big.audio_source == 'webrtc':
|
||||
audio_mode_persist = 'Webapp'
|
||||
|
||||
# Parse gain for all BIGs that are device inputs
|
||||
for big_config in conf.bigs:
|
||||
if big_config.audio_source.startswith('device:'):
|
||||
parts = big_config.audio_source.split(':', 1)[1].split(',')
|
||||
device_id = parts[0]
|
||||
gain_value = 1.0 # Default gain
|
||||
if len(parts) > 1:
|
||||
for part in parts[1:]:
|
||||
if part.startswith('gain='):
|
||||
try:
|
||||
gain_value = float(part.split('=')[1])
|
||||
except ValueError:
|
||||
log.warning(f"Invalid gain value in audio_source: {part}. Using default 1.0.")
|
||||
gain_value = 1.0
|
||||
break # Found gain, no need to check other parts
|
||||
big_config.input_gain = gain_value
|
||||
# Update audio_source to only contain the device ID for Multicaster compatibility if needed
|
||||
# For now, let's assume Multicaster will handle the full string or we adapt it later.
|
||||
# big_config.audio_source = f"device:{device_id}" # Optional: simplify for downstream if it doesn't parse gain
|
||||
# Derive audio_mode from first BIG audio_source
|
||||
first_source = conf.bigs[0].audio_source if conf.bigs else ''
|
||||
if first_source.startswith('device:'):
|
||||
audio_mode_persist = 'USB'
|
||||
|
||||
input_device = first_source.split(':', 1)[1] if ':' in first_source else 'default'
|
||||
elif first_source == 'webrtc':
|
||||
audio_mode_persist = 'Webapp'
|
||||
input_device = None
|
||||
else:
|
||||
audio_mode_persist = 'Network'
|
||||
input_device = None
|
||||
save_stream_settings({
|
||||
'channel_names': [big.name for big in conf.bigs],
|
||||
'languages': [big.language for big in conf.bigs],
|
||||
'audio_mode': audio_mode_persist,
|
||||
'input_device': input_device_persist, # Use the parsed device ID for saving
|
||||
'webapp_mic_gain': load_stream_settings().get('webapp_mic_gain', 1.0), # Preserve existing webapp gain
|
||||
'usb_mic_gain': load_stream_settings().get('usb_mic_gain', 1.0), # Preserve existing usb gain
|
||||
'timestamp': datetime.now(timezone.utc).isoformat()
|
||||
'input_device': input_device,
|
||||
'timestamp': datetime.utcnow().isoformat()
|
||||
})
|
||||
|
||||
# Persist the specific gain value that was just used for USB mode if applicable
|
||||
if audio_mode_persist == 'USB' and conf.bigs and conf.bigs[0].input_gain is not None:
|
||||
current_settings = load_stream_settings()
|
||||
current_settings['usb_mic_gain'] = conf.bigs[0].input_gain
|
||||
# Ensure timestamp is also updated if we are re-saving
|
||||
current_settings['timestamp'] = datetime.now(timezone.utc).isoformat()
|
||||
save_stream_settings(current_settings)
|
||||
global_config_group = conf
|
||||
# If there is an existing multicaster, cleanly shut it down first so audio devices are released
|
||||
if multicaster is not None:
|
||||
|
||||
Reference in New Issue
Block a user