use anyhow::{Context as _, Result};use collections::HashMap;use gpui::{App, BackgroundExecutor, BorrowAppContext, Global};use log::info; #[cfg(not(any(all(target_os = "windows", target_env = "gnu"), target_os = "freebsd")))]mod non_windows_and_freebsd_deps {    pub(super) use gpui::AsyncApp;    pub(super) use libwebrtc::native::apm;    pub(super) use parking_lot::Mutex;    pub(super) use rodio::cpal::Sample;    pub(super) use rodio::source::LimitSettings;    pub(super) use std::sync::Arc;} #[cfg(not(any(all(target_os = "windows", target_env = "gnu"), target_os = "freebsd")))]use non_windows_and_freebsd_deps::*; use rodio::{    Decoder, OutputStream, OutputStreamBuilder, Source, mixer::Mixer, nz, source::Buffered,};use settings::Settings;use std::{io::Cursor, num::NonZero, path::PathBuf, sync::atomic::Ordering, time::Duration};use util::ResultExt; mod audio_settings;mod replays;mod rodio_ext;pub use audio_settings::AudioSettings;pub use rodio_ext::RodioExt; use crate::audio_settings::LIVE_SETTINGS; // We are migrating to 16kHz sample rate from 48kHz. In the future// once we are reasonably sure most users have upgraded we will// remove the LEGACY parameters.//// We migrate to 16kHz because it is sufficient for speech and required// by the denoiser and future Speech to Text layers.pub const SAMPLE_RATE: NonZero<u32> = nz!(16000);pub const CHANNEL_COUNT: NonZero<u16> = nz!(1);pub const BUFFER_SIZE: usize = // echo canceller and livekit want 10ms of audio    (SAMPLE_RATE.get() as usize / 100) * CHANNEL_COUNT.get() as usize; pub const LEGACY_SAMPLE_RATE: NonZero<u32> = nz!(48000);pub const LEGACY_CHANNEL_COUNT: NonZero<u16> = nz!(2); pub const REPLAY_DURATION: Duration = Duration::from_secs(30); pub fn init(cx: &mut App) {    LIVE_SETTINGS.initialize(cx);} #[derive(Debug, Copy, Clone, Eq, Hash, PartialEq)]pub enum Sound {    Joined,    GuestJoined,    Leave,    Mute,    Unmute,    StartScreenshare,    StopScreenshare,    AgentDone,} impl Sound {    fn file(&self) -> &'static str {        match self {            Self::Joined => "joined_call",            Self::GuestJoined => "guest_joined_call",            Self::Leave => "leave_call",            Self::Mute => "mute",            Self::Unmute => "unmute",            Self::StartScreenshare => "start_screenshare",            Self::StopScreenshare => "stop_screenshare",            Self::AgentDone => "agent_done",        }    }} pub struct Audio {    output_handle: Option<OutputStream>,    output_mixer: Option<Mixer>,    #[cfg(not(any(all(target_os = "windows", target_env = "gnu"), target_os = "freebsd")))]    pub echo_canceller: Arc<Mutex<apm::AudioProcessingModule>>,    source_cache: HashMap<Sound, Buffered<Decoder<Cursor<Vec<u8>>>>>,    replays: replays::Replays,} impl Default for Audio {    fn default() -> Self {        Self {            output_handle: Default::default(),            output_mixer: Default::default(),            #[cfg(not(any(                all(target_os = "windows", target_env = "gnu"),                target_os = "freebsd"            )))]            echo_canceller: Arc::new(Mutex::new(apm::AudioProcessingModule::new(                true, false, false, false,            ))),            source_cache: Default::default(),            replays: Default::default(),        }    }} impl Global for Audio {} impl Audio {    fn ensure_output_exists(&mut self) -> Result<&Mixer> {        #[cfg(debug_assertions)]        log::warn!(            "Audio does not sound correct without optimizations. Use a release build to debug audio issues"        );         if self.output_handle.is_none() {            let output_handle = OutputStreamBuilder::open_default_stream()                .context("Could not open default output stream")?;            info!("Output stream: {:?}", output_handle);            self.output_handle = Some(output_handle);            if let Some(output_handle) = &self.output_handle {                let (mixer, source) = rodio::mixer::mixer(CHANNEL_COUNT, SAMPLE_RATE);                // or the mixer will end immediately as its empty.                mixer.add(rodio::source::Zero::new(CHANNEL_COUNT, SAMPLE_RATE));                self.output_mixer = Some(mixer);                 // The webrtc apm is not yet compiling for windows & freebsd                #[cfg(not(any(                    any(all(target_os = "windows", target_env = "gnu")),                    target_os = "freebsd"                )))]                let echo_canceller = Arc::clone(&self.echo_canceller);                #[cfg(not(any(                    any(all(target_os = "windows", target_env = "gnu")),                    target_os = "freebsd"                )))]                let source = source.inspect_buffer::<BUFFER_SIZE, _>(move |buffer| {                    let mut buf: [i16; _] = buffer.map(|s| s.to_sample());                    echo_canceller                        .lock()                        .process_reverse_stream(                            &mut buf,                            SAMPLE_RATE.get() as i32,                            CHANNEL_COUNT.get().into(),                        )                        .expect("Audio input and output threads should not panic");                });                output_handle.mixer().add(source);            }        }         Ok(self            .output_mixer            .as_ref()            .expect("we only get here if opening the outputstream succeeded"))    }     pub fn save_replays(        &self,        executor: BackgroundExecutor,    ) -> gpui::Task<anyhow::Result<(PathBuf, Duration)>> {        self.replays.replays_to_tar(executor)    }     #[cfg(not(any(all(target_os = "windows", target_env = "gnu"), target_os = "freebsd")))]    pub fn open_microphone(voip_parts: VoipParts) -> anyhow::Result<impl Source> {        let stream = rodio::microphone::MicrophoneBuilder::new()            .default_device()?            .default_config()?            .prefer_sample_rates([                SAMPLE_RATE, // sample rates trivially resamplable to `SAMPLE_RATE`                SAMPLE_RATE.saturating_mul(nz!(2)),                SAMPLE_RATE.saturating_mul(nz!(3)),                SAMPLE_RATE.saturating_mul(nz!(4)),            ])            .prefer_channel_counts([nz!(1), nz!(2), nz!(3), nz!(4)])            .prefer_buffer_sizes(512..)            .open_stream()?;        info!("Opened microphone: {:?}", stream.config());         let stream = stream            .possibly_disconnected_channels_to_mono()            .constant_samplerate(SAMPLE_RATE)            .limit(LimitSettings::live_performance())            .process_buffer::<BUFFER_SIZE, _>(move |buffer| {                let mut int_buffer: [i16; _] = buffer.map(|s| s.to_sample());                if voip_parts                    .echo_canceller                    .lock()                    .process_stream(                        &mut int_buffer,                        SAMPLE_RATE.get() as i32,                        CHANNEL_COUNT.get() as i32,                    )                    .context("livekit audio processor error")                    .log_err()                    .is_some()                {                    for (sample, processed) in buffer.iter_mut().zip(&int_buffer) {                        *sample = (*processed).to_sample();                    }                }            })            .denoise()            .context("Could not set up denoiser")?            .automatic_gain_control(0.90, 1.0, 0.0, 5.0)            .periodic_access(Duration::from_millis(100), move |agc_source| {                agc_source                    .set_enabled(LIVE_SETTINGS.auto_microphone_volume.load(Ordering::Relaxed));                let denoise = agc_source.inner_mut();                denoise.set_enabled(LIVE_SETTINGS.denoise.load(Ordering::Relaxed));            });         let stream = if voip_parts.legacy_audio_compatible {            stream.constant_params(LEGACY_CHANNEL_COUNT, LEGACY_SAMPLE_RATE)        } else {            stream.constant_params(CHANNEL_COUNT, SAMPLE_RATE)        };         let (replay, stream) = stream.replayable(REPLAY_DURATION)?;        voip_parts            .replays            .add_voip_stream("local microphone".to_string(), replay);         Ok(stream)    }     pub fn play_voip_stream(        source: impl rodio::Source + Send + 'static,        speaker_name: String,        is_staff: bool,        cx: &mut App,    ) -> anyhow::Result<()> {        let (replay_source, source) = source            .constant_params(CHANNEL_COUNT, SAMPLE_RATE)            .automatic_gain_control(0.90, 1.0, 0.0, 5.0)            .periodic_access(Duration::from_millis(100), move |agc_source| {                agc_source.set_enabled(LIVE_SETTINGS.auto_speaker_volume.load(Ordering::Relaxed));            })            .replayable(REPLAY_DURATION)            .expect("REPLAY_DURATION is longer than 100ms");         cx.update_default_global(|this: &mut Self, _cx| {            let output_mixer = this                .ensure_output_exists()                .context("Could not get output mixer")?;            output_mixer.add(source);            if is_staff {                this.replays.add_voip_stream(speaker_name, replay_source);            }            Ok(())        })    }     pub fn play_sound(sound: Sound, cx: &mut App) {        cx.update_default_global(|this: &mut Self, cx| {            let source = this.sound_source(sound, cx).log_err()?;            let output_mixer = this                .ensure_output_exists()                .context("Could not get output mixer")                .log_err()?;             output_mixer.add(source);            Some(())        });    }     pub fn end_call(cx: &mut App) {        cx.update_default_global(|this: &mut Self, _cx| {            this.output_handle.take();        });    }     fn sound_source(&mut self, sound: Sound, cx: &App) -> Result<impl Source + use<>> {        if let Some(wav) = self.source_cache.get(&sound) {            return Ok(wav.clone());        }         let path = format!("sounds/{}.wav", sound.file());        let bytes = cx            .asset_source()            .load(&path)?            .map(anyhow::Ok)            .with_context(|| format!("No asset available for path {path}"))??            .into_owned();        let cursor = Cursor::new(bytes);        let source = Decoder::new(cursor)?.buffered();         self.source_cache.insert(sound, source.clone());         Ok(source)    }} #[cfg(not(any(all(target_os = "windows", target_env = "gnu"), target_os = "freebsd")))]pub struct VoipParts {    echo_canceller: Arc<Mutex<apm::AudioProcessingModule>>,    replays: replays::Replays,    legacy_audio_compatible: bool,} #[cfg(not(any(all(target_os = "windows", target_env = "gnu"), target_os = "freebsd")))]impl VoipParts {    pub fn new(cx: &AsyncApp) -> anyhow::Result<Self> {        let (apm, replays) = cx.try_read_default_global::<Audio, _>(|audio, _| {            (Arc::clone(&audio.echo_canceller), audio.replays.clone())        })?;        let legacy_audio_compatible =            AudioSettings::try_read_global(cx, |settings| settings.legacy_audio_compatible)                .unwrap_or(true);         Ok(Self {            legacy_audio_compatible,            echo_canceller: apm,            replays,        })    }}