Merge pull request #13 from legop3/fix-laptoprover-audio

Fix laptoprover tts and horn including gtts
This commit is contained in:
legop3
2026-07-07 00:34:03 -04:00
committed by GitHub
3 changed files with 402 additions and 75 deletions
+71 -61
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@@ -1,88 +1,98 @@
# ALSA routing for the Debian laptop rover profile.
# Dedicated ALSA routing for the Debian laptop rover profile.
#
# This file intentionally mirrors the logical device names used by the Pi rover
# audio setup. roverd can keep sending horn audio to "horn", forwarded browser
# audio to "forward", and TTS to ALSA's default playback path without caring
# which physical sound card is underneath the profile.
# This intentionally mirrors pi/asound.conf as closely as a normal PC can:
# one fixed hardware card, one dmix playback engine, separate softvol controls
# for TTS/horn/forwarded audio, and a raw capture alias for the rover mic.
#
# This is NOT meant to preserve normal desktop audio behavior. The laptop rover
# installer disables PipeWire/PulseAudio so roverd owns the audio hardware like
# the Raspberry Pi rover does. If the laptop's real speaker/mic card is not ALSA
# card 0, change the hw:0,0/card 0 references below to the card shown by:
# aplay -l
# arecord -l
pcm.roverd_playback {
type plug
# Use the system's first normal ALSA playback device as the physical sink.
# This avoids referencing "default" here, because this file replaces
# pcm.!default below and using it as a slave would recurse.
slave.pcm "sysdefault"
}
pcm.roverd_capture {
type plug
# The media publisher records from "default"; with pcm.!default below that
# capture side resolves here. Keeping capture separate from playback lets the
# asym default expose ordinary microphone input while playback goes through
# the TTS softvol path.
slave.pcm "sysdefault"
# Mix multiple playback clients in software with a fixed low-cost format.
pcm.dmixer {
type dmix
ipc_key 1024
ipc_perm 0666
slave {
pcm "hw:0,0"
format S16_LE
rate 16000
channels 1
period_time 0
period_size 1024
buffer_size 4096
}
}
# TTS volume control (used by default playback path).
pcm.tts_softvol {
type softvol
slave.pcm "roverd_playback"
control {
name "TTSMaster"
card 0
}
min_dB -60.0
max_dB 12.0
type softvol
slave.pcm "dmixer"
control {
name "TTSMaster"
card 0
}
min_dB -60.0
max_dB 12.0
}
# Horn volume control.
pcm.horn_softvol {
type softvol
slave.pcm "roverd_playback"
control {
name "HornMaster"
card 0
}
min_dB -60.0
max_dB 12.0
type softvol
slave.pcm "dmixer"
control {
name "HornMaster"
card 0
}
min_dB -60.0
max_dB 12.0
}
# Forwarded audio volume control.
pcm.forward_softvol {
type softvol
slave.pcm "roverd_playback"
control {
name "ForwardMaster"
card 0
}
min_dB -60.0
max_dB 12.0
type softvol
slave.pcm "dmixer"
control {
name "ForwardMaster"
card 0
}
min_dB -60.0
max_dB 12.0
}
# Per-source playback PCMs.
pcm.tts {
type plug
slave.pcm "tts_softvol"
type plug
slave.pcm "tts_softvol"
}
pcm.horn {
type plug
slave.pcm "horn_softvol"
type plug
slave.pcm "horn_softvol"
}
pcm.forward {
type plug
slave.pcm "forward_softvol"
type plug
slave.pcm "forward_softvol"
}
pcm.!default {
type asym
# Capture alias used by laptop rover config defaults.
pcm.rovermic {
type plug
slave.pcm "hw:0,0"
}
# Existing TTS engines play to their default ALSA output, so default playback
# is intentionally the TTS path. This preserves the current TTS execution
# model while still making the TTS volume control meaningful on laptops.
playback.pcm "tts"
capture.pcm "roverd_capture"
# Defaults: TTS direct playback + raw capture on the dedicated laptop sound card.
pcm.!default {
type asym
playback.pcm "tts"
capture.pcm "rovermic"
}
ctl.!default {
type hw
card 0
type hw
card 0
}
+202
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@@ -0,0 +1,202 @@
#!/usr/bin/env python3
import ctypes
import ctypes.util
import json
import os
import struct
import subprocess
import sys
ASSET_ROOT = "/opt/roverd/googletts"
LIB_PATH = os.path.join(ASSET_ROOT, "libchrometts.so")
VOICE_DIR = os.path.join(ASSET_ROOT, "en-us-x-multi-r30")
PIPELINE = "pipeline.pb"
PLAYBACK_DEVICE = "tts"
SAMPLE_RATE = "24000"
MAX_TEXT_CHARS = 512
VOICES = {
"sfg": "female",
"iob": "female",
"iog": "female",
"iol": "male",
"iom": "male",
"tpc": "female",
"tpd": "male",
"tpf": "female",
}
DEFAULT_VOICE = "tpf"
DEFAULT_PITCH = 1.0
DEFAULT_SPEED = 1.0
MIN_PITCH = 0.5
MAX_PITCH = 2.0
MIN_SPEED = 0.5
MAX_SPEED = 2.0
_runtime_handles = []
def load_shared_library(path):
mode = ctypes.RTLD_GLOBAL | getattr(os, "RTLD_NOW", 0)
return ctypes.CDLL(path, mode=mode)
def preload_runtime_libraries():
# Laptop-only workaround: some ChromeOS libchrometts builds reference
# compiler helper symbols such as __udivmodti4 without declaring the runtime
# library as an ELF dependency. Loading common compiler runtimes globally
# first makes those symbols visible before ctypes loads libchrometts.so.
for name in ("gcc_s", "atomic", "stdc++", "c++", "c++abi"):
lib = ctypes.util.find_library(name)
if not lib:
continue
try:
_runtime_handles.append(load_shared_library(lib))
except OSError:
pass
preload_runtime_libraries()
def varint(value):
out = bytearray()
while value >= 0x80:
out.append((value & 0x7F) | 0x80)
value >>= 7
out.append(value)
return bytes(out)
def field_bytes(number, payload):
return varint((number << 3) | 2) + varint(len(payload)) + payload
def field_float(number, value):
return varint((number << 3) | 5) + struct.pack("<f", float(value))
def build_utterance(text, pitch=1.0, speed=1.0):
params = field_float(2, pitch) + field_float(3, speed)
msg_b = field_bytes(1, text.encode("utf-8")) + field_bytes(20, params)
msg_a = field_bytes(1, msg_b)
return field_bytes(1, msg_a)
def build_speaker(name, gender):
return field_bytes(1, name.encode("utf-8")) + field_bytes(2, gender.encode("utf-8"))
class ChromeTTS:
def __init__(self):
self.lib = load_shared_library(LIB_PATH)
self.lib.GoogleTtsInit.argtypes = [ctypes.c_char_p, ctypes.c_char_p]
self.lib.GoogleTtsInit.restype = ctypes.c_bool
self.lib.GoogleTtsInitBuffered.argtypes = [ctypes.c_char_p, ctypes.c_char_p, ctypes.c_int, ctypes.c_int]
self.lib.GoogleTtsInitBuffered.restype = ctypes.c_bool
self.lib.GoogleTtsGetFramesInAudioBuffer.argtypes = []
self.lib.GoogleTtsGetFramesInAudioBuffer.restype = ctypes.c_size_t
self.lib.GoogleTtsReadBuffered.argtypes = [
ctypes.POINTER(ctypes.c_float),
ctypes.POINTER(ctypes.c_size_t),
]
self.lib.GoogleTtsReadBuffered.restype = ctypes.c_int
self.lib.GoogleTtsShutdown.argtypes = []
self.lib.GoogleTtsShutdown.restype = None
voice_dir = os.path.abspath(VOICE_DIR) + os.sep
pipeline = os.path.join(voice_dir, PIPELINE)
if not self.lib.GoogleTtsInit(pipeline.encode("utf-8"), voice_dir.encode("utf-8")):
raise RuntimeError("GoogleTtsInit failed")
self.frames = int(self.lib.GoogleTtsGetFramesInAudioBuffer())
if self.frames <= 0:
raise RuntimeError("invalid Google TTS audio buffer size")
self.buffer = (ctypes.c_float * self.frames)()
def speak_to_aplay(self, text, voice, pitch=DEFAULT_PITCH, speed=DEFAULT_SPEED):
voice = voice if voice in VOICES else DEFAULT_VOICE
pitch = clamp_float(pitch, MIN_PITCH, MAX_PITCH, DEFAULT_PITCH)
speed = clamp_float(speed, MIN_SPEED, MAX_SPEED, DEFAULT_SPEED)
text = text.strip()
if not text:
raise ValueError("text required")
text = text[:MAX_TEXT_CHARS]
utterance = build_utterance(text, pitch=pitch, speed=speed)
speaker = build_speaker(voice, VOICES[voice])
if not self.lib.GoogleTtsInitBuffered(utterance, speaker, len(utterance), len(speaker)):
raise RuntimeError("GoogleTtsInitBuffered failed")
player = subprocess.Popen(
["aplay", "-q", "-D", PLAYBACK_DEVICE, "-r", SAMPLE_RATE, "-f", "FLOAT_LE", "-c", "1"],
stdin=subprocess.PIPE,
)
try:
frames_written = ctypes.c_size_t(0)
while self.lib.GoogleTtsReadBuffered(self.buffer, ctypes.byref(frames_written)) > 0:
frames = int(frames_written.value)
if frames > 0:
player.stdin.write(ctypes.string_at(self.buffer, frames * ctypes.sizeof(ctypes.c_float)))
player.stdin.close()
rc = player.wait()
if rc != 0:
raise RuntimeError(f"aplay exited with {rc}")
finally:
if player.poll() is None:
player.kill()
player.wait()
def shutdown(self):
self.lib.GoogleTtsShutdown()
def respond(payload):
sys.stdout.write(json.dumps(payload, separators=(",", ":")) + "\n")
sys.stdout.flush()
def clamp_float(value, minimum, maximum, fallback):
try:
value = float(value)
except (TypeError, ValueError):
return fallback
if value <= 0:
return fallback
if value < minimum:
return minimum
if value > maximum:
return maximum
return value
def main():
try:
tts = ChromeTTS()
except Exception as exc:
respond({"ok": False, "error": str(exc)})
return 1
respond({"ok": True, "ready": True})
try:
for line in sys.stdin:
line = line.strip()
if not line:
continue
try:
request = json.loads(line)
tts.speak_to_aplay(
str(request.get("text") or ""),
str(request.get("voice") or DEFAULT_VOICE),
request.get("pitch", DEFAULT_PITCH),
request.get("speed", DEFAULT_SPEED),
)
respond({"ok": True})
except Exception as exc:
respond({"ok": False, "error": str(exc)})
finally:
tts.shutdown()
return 0
if __name__ == "__main__":
raise SystemExit(main())
+129 -14
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@@ -1,23 +1,63 @@
#!/usr/bin/env bash
install_debian_laptop_deps() {
# This profile is deliberately Debian-only. Using apt directly is simpler
# than adding a fake cross-distro layer, and it keeps the installed package
# set easy to inspect on the actual rover laptop.
# The laptop rover is a dedicated appliance, not a normal desktop/laptop audio
# install. Keep this package set intentionally close to the Pi profile so the
# same roverd TTS/playback/capture code paths are available on both targets.
if command -v ffmpeg >/dev/null 2>&1 \
&& command -v arecord >/dev/null 2>&1 \
&& command -v aplay >/dev/null 2>&1 \
&& command -v amixer >/dev/null 2>&1 \
&& command -v v4l2-ctl >/dev/null 2>&1 \
&& command -v flite >/dev/null 2>&1 \
&& command -v espeak >/dev/null 2>&1; then
log "Debian laptop media/audio dependencies already installed; skipping apt install"
&& command -v espeak >/dev/null 2>&1 \
&& command -v python3 >/dev/null 2>&1 \
&& command -v curl >/dev/null 2>&1 \
&& command -v xz >/dev/null 2>&1 \
&& command -v unzip >/dev/null 2>&1 \
&& ldconfig -p 2>/dev/null | grep -q 'libgcc_s\.so\.1' \
&& ldconfig -p 2>/dev/null | grep -q 'libstdc\+\+\.so\.6' \
&& ldconfig -p 2>/dev/null | grep -q 'libc++\.so\.1' \
&& ldconfig -p 2>/dev/null | grep -q 'libc++abi\.so\.1'; then
log "Debian laptop media/audio/TTS dependencies already installed; skipping apt install"
return
fi
log "Installing Debian laptop dependencies (ffmpeg, ALSA tools, V4L2 tools, flite, espeak)..."
log "Installing Debian laptop rover dependencies (ffmpeg, ALSA tools, V4L2 tools, flite/espeak, Chrome TTS runtime deps)..."
apt-get update
apt-get install -y --no-install-recommends ffmpeg alsa-utils v4l-utils ca-certificates flite espeak
apt-get install -y --no-install-recommends \
ffmpeg alsa-utils v4l-utils ca-certificates flite espeak python3 curl xz-utils unzip libasound2-plugins libgcc-s1 libstdc++6 libc++1 libc++abi1 \
|| apt-get install -y --no-install-recommends \
ffmpeg alsa-utils v4l-utils ca-certificates flite espeak python3 curl xz-utils unzip libasound2-plugins libgcc-s1 libstdc++6 libc++1-14 libc++abi1-14
}
disable_debian_laptop_desktop_audio_stack() {
# This profile is for a dedicated rover laptop. PipeWire/PulseAudio are good
# desktop defaults, but they can grab the hardware device and make the rover's
# root/systemd ALSA services fail or route through a moving per-user graph.
# Mask them globally and kill already-running instances so ALSA owns the box,
# which is the closest behavior to the Pi rover appliance setup.
log "Disabling desktop audio daemons for dedicated laptop rover audio"
local -a user_units=(
pipewire.service
pipewire.socket
pipewire-pulse.service
pipewire-pulse.socket
wireplumber.service
pulseaudio.service
pulseaudio.socket
)
if command -v systemctl >/dev/null 2>&1; then
systemctl --global disable "${user_units[@]}" >/dev/null 2>&1 || true
systemctl --global mask "${user_units[@]}" >/dev/null 2>&1 || true
fi
pkill -x pipewire >/dev/null 2>&1 || true
pkill -x pipewire-pulse >/dev/null 2>&1 || true
pkill -x wireplumber >/dev/null 2>&1 || true
pkill -x pulseaudio >/dev/null 2>&1 || true
}
install_debian_laptop_audio_support() {
@@ -26,17 +66,92 @@ install_debian_laptop_audio_support() {
return
fi
# The laptop profile still uses roverd's existing audio contract: TTS plays
# to ALSA's default output, horn plays to the named "horn" device, and
# forwarded web audio plays to the named "forward" device. Installing one
# profile-specific asound.conf gives those paths independent softvol mixer
# controls without changing the TTS runtime code.
install -m 0644 pi/asound.debian-laptop.conf /etc/asound.conf
log "Installed Debian laptop ALSA config to /etc/asound.conf"
log "ALSA config updated; restarting audio clients or rebooting is recommended before testing laptop audio"
log "Installed dedicated Debian laptop ALSA config to /etc/asound.conf"
install -D -o root -g root -m 0755 pi/bin/chromegtts-daemon-laptop.py /usr/local/bin/chromegtts-daemon
log "Installed laptop chromegtts daemon"
install_google_tts_assets_laptop
log "ALSA config updated; reboot recommended before testing laptop rover audio"
}
install_google_tts_assets_laptop() {
local asset_dir="/opt/roverd/googletts"
local voice_dir="${asset_dir}/en-us-x-multi-r30"
local dist_url="https://storage.googleapis.com/chromeos-localmirror/distfiles/googletts-26.5.tar.xz"
local tmp_dir
local lib_member=""
local member
if [[ -f "${asset_dir}/libchrometts.so" && -f "${voice_dir}/pipeline.pb" ]]; then
log "Google Chrome TTS assets already installed; skipping download"
return
fi
tmp_dir="$(mktemp -d)"
log "Downloading Google Chrome TTS assets for Debian laptop profile..."
if ! curl -L -o "${tmp_dir}/googletts-26.5.tar.xz" "$dist_url"; then
rm -rf "$tmp_dir"
log "WARNING: failed to download Google Chrome TTS assets; chromegtts will be unavailable"
return
fi
local -a candidate_libs=()
case "$(uname -m)" in
aarch64|arm64)
candidate_libs=(libchrometts_arm64.so)
;;
armv7l|armhf)
candidate_libs=(libchrometts_armv7.so)
;;
x86_64|amd64)
candidate_libs=(libchrometts_x86_64.so libchrometts_amd64.so libchrometts_x64.so libchrometts.so)
;;
i386|i686)
candidate_libs=(libchrometts_x86.so libchrometts_i386.so libchrometts.so)
;;
*)
log "WARNING: unsupported Chrome TTS architecture $(uname -m); skipping Google TTS assets"
rm -rf "$tmp_dir"
return
;;
esac
for member in "${candidate_libs[@]}"; do
if tar -tf "${tmp_dir}/googletts-26.5.tar.xz" "$member" >/dev/null 2>&1; then
lib_member="$member"
break
fi
done
if [[ -z "$lib_member" ]]; then
log "WARNING: no libchrometts library matching $(uname -m) found in Google TTS archive; chromegtts will be unavailable"
rm -rf "$tmp_dir"
return
fi
if ! tar -xf "${tmp_dir}/googletts-26.5.tar.xz" -C "$tmp_dir" en-us-x-multi.zvoice "$lib_member"; then
rm -rf "$tmp_dir"
log "WARNING: failed to unpack Google Chrome TTS assets; chromegtts will be unavailable"
return
fi
install -d -o root -g root -m 0755 "$asset_dir"
install -o root -g root -m 0644 "${tmp_dir}/${lib_member}" "${asset_dir}/libchrometts.so"
rm -rf "$voice_dir"
install -d -o root -g root -m 0755 "$voice_dir"
unzip -q "${tmp_dir}/en-us-x-multi.zvoice" -d "$voice_dir"
chown -R root:root "$asset_dir"
find "$asset_dir" -type d -exec chmod 0755 {} +
find "$asset_dir" -type f -exec chmod 0644 {} +
rm -rf "$tmp_dir"
log "Installed Google Chrome TTS assets to $asset_dir using $lib_member"
}
install_debian_laptop_profile() {
install_debian_laptop_deps
disable_debian_laptop_desktop_audio_stack
install_debian_laptop_audio_support
}