#![cfg_attr(all(target_os = "windows", target_env = "gnu"), allow(unused))] mod remote_video_track_view;#[cfg(any(    test,    feature = "test-support",    all(target_os = "windows", target_env = "gnu")))]pub mod test; use anyhow::{anyhow, Context as _, Result};use cpal::traits::{DeviceTrait, HostTrait, StreamTrait as _};use futures::{io, Stream, StreamExt as _};use gpui::{    BackgroundExecutor, ScreenCaptureFrame, ScreenCaptureSource, ScreenCaptureStream, Task,};use parking_lot::Mutex;use std::{borrow::Cow, collections::VecDeque, future::Future, pin::Pin, sync::Arc, thread};use util::{debug_panic, ResultExt as _};#[cfg(not(all(target_os = "windows", target_env = "gnu")))]use webrtc::{    audio_frame::AudioFrame,    audio_source::{native::NativeAudioSource, AudioSourceOptions, RtcAudioSource},    audio_stream::native::NativeAudioStream,    video_frame::{VideoBuffer, VideoFrame, VideoRotation},    video_source::{native::NativeVideoSource, RtcVideoSource, VideoResolution},    video_stream::native::NativeVideoStream,}; #[cfg(all(    not(any(test, feature = "test-support")),    not(all(target_os = "windows", target_env = "gnu"))))]use livekit::track::RemoteAudioTrack;#[cfg(all(    not(any(test, feature = "test-support")),    not(all(target_os = "windows", target_env = "gnu"))))]pub use livekit::*;#[cfg(any(    test,    feature = "test-support",    all(target_os = "windows", target_env = "gnu")))]use test::track::RemoteAudioTrack;#[cfg(any(    test,    feature = "test-support",    all(target_os = "windows", target_env = "gnu")))]pub use test::*; pub use remote_video_track_view::{RemoteVideoTrackView, RemoteVideoTrackViewEvent}; pub enum AudioStream {    Input {        _thread_handle: std::sync::mpsc::Sender<()>,        _transmit_task: Task<()>,    },    Output {        _task: Task<()>,    },} struct Dispatcher(Arc<dyn gpui::PlatformDispatcher>); #[cfg(not(all(target_os = "windows", target_env = "gnu")))]impl livekit::dispatcher::Dispatcher for Dispatcher {    fn dispatch(&self, runnable: livekit::dispatcher::Runnable) {        self.0.dispatch(runnable, None);    }     fn dispatch_after(        &self,        duration: std::time::Duration,        runnable: livekit::dispatcher::Runnable,    ) {        self.0.dispatch_after(duration, runnable);    }} struct HttpClientAdapter(Arc<dyn http_client::HttpClient>); fn http_2_status(status: http_client::http::StatusCode) -> http_2::StatusCode {    http_2::StatusCode::from_u16(status.as_u16())        .expect("valid status code to status code conversion")} #[cfg(not(all(target_os = "windows", target_env = "gnu")))]impl livekit::dispatcher::HttpClient for HttpClientAdapter {    fn get(        &self,        url: &str,    ) -> Pin<Box<dyn Future<Output = io::Result<livekit::dispatcher::Response>> + Send>> {        let http_client = self.0.clone();        let url = url.to_string();        Box::pin(async move {            let response = http_client                .get(&url, http_client::AsyncBody::empty(), false)                .await                .map_err(io::Error::other)?;            Ok(livekit::dispatcher::Response {                status: http_2_status(response.status()),                body: Box::pin(response.into_body()),            })        })    }     fn send_async(        &self,        request: http_2::Request<Vec<u8>>,    ) -> Pin<Box<dyn Future<Output = io::Result<livekit::dispatcher::Response>> + Send>> {        let http_client = self.0.clone();        let mut builder = http_client::http::Request::builder()            .method(request.method().as_str())            .uri(request.uri().to_string());         for (key, value) in request.headers().iter() {            builder = builder.header(key.as_str(), value.as_bytes());        }         if !request.extensions().is_empty() {            debug_panic!(                "Livekit sent an HTTP request with a protocol extension that Zed doesn't support!"            );        }         let request = builder            .body(http_client::AsyncBody::from_bytes(                request.into_body().into(),            ))            .unwrap();         Box::pin(async move {            let response = http_client.send(request).await.map_err(io::Error::other)?;            Ok(livekit::dispatcher::Response {                status: http_2_status(response.status()),                body: Box::pin(response.into_body()),            })        })    }} #[cfg(all(target_os = "windows", target_env = "gnu"))]pub fn init(    dispatcher: Arc<dyn gpui::PlatformDispatcher>,    http_client: Arc<dyn http_client::HttpClient>,) {} #[cfg(not(all(target_os = "windows", target_env = "gnu")))]pub fn init(    dispatcher: Arc<dyn gpui::PlatformDispatcher>,    http_client: Arc<dyn http_client::HttpClient>,) {    livekit::dispatcher::set_dispatcher(Dispatcher(dispatcher));    livekit::dispatcher::set_http_client(HttpClientAdapter(http_client));} #[cfg(not(all(target_os = "windows", target_env = "gnu")))]pub async fn capture_local_video_track(    capture_source: &dyn ScreenCaptureSource,) -> Result<(track::LocalVideoTrack, Box<dyn ScreenCaptureStream>)> {    let resolution = capture_source.resolution()?;    let track_source = NativeVideoSource::new(VideoResolution {        width: resolution.width.0 as u32,        height: resolution.height.0 as u32,    });     let capture_stream = capture_source        .stream({            let track_source = track_source.clone();            Box::new(move |frame| {                if let Some(buffer) = video_frame_buffer_to_webrtc(frame) {                    track_source.capture_frame(&VideoFrame {                        rotation: VideoRotation::VideoRotation0,                        timestamp_us: 0,                        buffer,                    });                }            })        })        .await??;     Ok((        track::LocalVideoTrack::create_video_track(            "screen share",            RtcVideoSource::Native(track_source),        ),        capture_stream,    ))} #[cfg(not(all(target_os = "windows", target_env = "gnu")))]pub fn capture_local_audio_track(    background_executor: &BackgroundExecutor,) -> Result<Task<(track::LocalAudioTrack, AudioStream)>> {    use util::maybe;     let (frame_tx, mut frame_rx) = futures::channel::mpsc::unbounded();    let (thread_handle, thread_kill_rx) = std::sync::mpsc::channel::<()>();    let sample_rate;    let channels;     if cfg!(any(test, feature = "test-support")) {        sample_rate = 2;        channels = 1;    } else {        let (device, config) = default_device(true)?;        sample_rate = config.sample_rate().0;        channels = config.channels() as u32;        thread::spawn(move || {            maybe!({                if let Some(name) = device.name().ok() {                    log::info!("Using microphone: {}", name)                } else {                    log::info!("Using microphone: <unknown>");                }                 let stream = device                    .build_input_stream_raw(                        &config.config(),                        cpal::SampleFormat::I16,                        move |data, _: &_| {                            frame_tx                                .unbounded_send(AudioFrame {                                    data: Cow::Owned(data.as_slice::<i16>().unwrap().to_vec()),                                    sample_rate,                                    num_channels: channels,                                    samples_per_channel: data.len() as u32 / channels,                                })                                .ok();                        },                        |err| log::error!("error capturing audio track: {:?}", err),                        None,                    )                    .context("failed to build input stream")?;                 stream.play()?;                // Keep the thread alive and holding onto the `stream`                thread_kill_rx.recv().ok();                anyhow::Ok(Some(()))            })            .log_err();        });    }     Ok(background_executor.spawn({        let background_executor = background_executor.clone();        async move {            let source = NativeAudioSource::new(                AudioSourceOptions {                    echo_cancellation: true,                    noise_suppression: true,                    auto_gain_control: true,                },                sample_rate,                channels,                100,            );            let transmit_task = background_executor.spawn({                let source = source.clone();                async move {                    while let Some(frame) = frame_rx.next().await {                        source.capture_frame(&frame).await.log_err();                    }                }            });             let track = track::LocalAudioTrack::create_audio_track(                "microphone",                RtcAudioSource::Native(source),            );             (                track,                AudioStream::Input {                    _thread_handle: thread_handle,                    _transmit_task: transmit_task,                },            )        }    }))} #[cfg(not(all(target_os = "windows", target_env = "gnu")))]pub fn play_remote_audio_track(    track: &RemoteAudioTrack,    background_executor: &BackgroundExecutor,) -> Result<AudioStream> {    let track = track.clone();    // We track device changes in our output because Livekit has a resampler built in,    // and it's easy to create a new native audio stream when the device changes.    if cfg!(any(test, feature = "test-support")) {        Ok(AudioStream::Output {            _task: background_executor.spawn(async {}),        })    } else {        let mut default_change_listener = DeviceChangeListener::new(false)?;        let (output_device, output_config) = default_device(false)?;         let _task = background_executor.spawn({            let background_executor = background_executor.clone();            async move {                let (mut _receive_task, mut _thread) =                    start_output_stream(output_config, output_device, &track, &background_executor);                 while let Some(_) = default_change_listener.next().await {                    let Some((output_device, output_config)) = get_default_output().log_err()                    else {                        continue;                    };                     if let Ok(name) = output_device.name() {                        log::info!("Using speaker: {}", name)                    } else {                        log::info!("Using speaker: <unknown>")                    }                     (_receive_task, _thread) = start_output_stream(                        output_config,                        output_device,                        &track,                        &background_executor,                    );                }                 futures::future::pending::<()>().await;            }        });         Ok(AudioStream::Output { _task })    }} fn default_device(input: bool) -> anyhow::Result<(cpal::Device, cpal::SupportedStreamConfig)> {    let device;    let config;    if input {        device = cpal::default_host()            .default_input_device()            .ok_or_else(|| anyhow!("no audio input device available"))?;        config = device            .default_input_config()            .context("failed to get default input config")?;    } else {        device = cpal::default_host()            .default_output_device()            .ok_or_else(|| anyhow!("no audio output device available"))?;        config = device            .default_output_config()            .context("failed to get default output config")?;    }    Ok((device, config))} #[cfg(not(all(target_os = "windows", target_env = "gnu")))]fn get_default_output() -> anyhow::Result<(cpal::Device, cpal::SupportedStreamConfig)> {    let host = cpal::default_host();    let output_device = host        .default_output_device()        .context("failed to read default output device")?;    let output_config = output_device.default_output_config()?;    Ok((output_device, output_config))} #[cfg(not(all(target_os = "windows", target_env = "gnu")))]fn start_output_stream(    output_config: cpal::SupportedStreamConfig,    output_device: cpal::Device,    track: &track::RemoteAudioTrack,    background_executor: &BackgroundExecutor,) -> (Task<()>, std::sync::mpsc::Sender<()>) {    let buffer = Arc::new(Mutex::new(VecDeque::<i16>::new()));    let sample_rate = output_config.sample_rate();     let mut stream = NativeAudioStream::new(        track.rtc_track(),        sample_rate.0 as i32,        output_config.channels() as i32,    );     let receive_task = background_executor.spawn({        let buffer = buffer.clone();        async move {            const MS_OF_BUFFER: u32 = 100;            const MS_IN_SEC: u32 = 1000;            while let Some(frame) = stream.next().await {                let frame_size = frame.samples_per_channel * frame.num_channels;                debug_assert!(frame.data.len() == frame_size as usize);                 let buffer_size =                    ((frame.sample_rate * frame.num_channels) / MS_IN_SEC * MS_OF_BUFFER) as usize;                 let mut buffer = buffer.lock();                let new_size = buffer.len() + frame.data.len();                if new_size > buffer_size {                    let overflow = new_size - buffer_size;                    buffer.drain(0..overflow);                }                 buffer.extend(frame.data.iter());            }        }    });     // The _output_stream needs to be on it's own thread because it's !Send    // and we experienced a deadlock when it's created on the main thread.    let (thread, end_on_drop_rx) = std::sync::mpsc::channel::<()>();    thread::spawn(move || {        if cfg!(any(test, feature = "test-support")) {            // Can't play audio in tests            return;        }         let output_stream = output_device.build_output_stream(            &output_config.config(),            {                let buffer = buffer.clone();                move |data, _info| {                    let mut buffer = buffer.lock();                    if buffer.len() < data.len() {                        // Instead of partially filling a buffer, output silence. If a partial                        // buffer was outputted then this could lead to a perpetual state of                        // outputting partial buffers as it never gets filled enough for a full                        // frame.                        data.fill(0);                    } else {                        // SAFETY: We know that buffer has at least data.len() values in it.                        // because we just checked                        let mut drain = buffer.drain(..data.len());                        data.fill_with(|| unsafe { drain.next().unwrap_unchecked() });                    }                }            },            |error| log::error!("error playing audio track: {:?}", error),            None,        );         let Some(output_stream) = output_stream.log_err() else {            return;        };         output_stream.play().log_err();        // Block forever to keep the output stream alive        end_on_drop_rx.recv().ok();    });     (receive_task, thread)} #[cfg(all(target_os = "windows", target_env = "gnu"))]pub fn play_remote_video_track(    track: &track::RemoteVideoTrack,) -> impl Stream<Item = RemoteVideoFrame> {    futures::stream::empty()} #[cfg(not(all(target_os = "windows", target_env = "gnu")))]pub fn play_remote_video_track(    track: &track::RemoteVideoTrack,) -> impl Stream<Item = RemoteVideoFrame> {    NativeVideoStream::new(track.rtc_track())        .filter_map(|frame| async move { video_frame_buffer_from_webrtc(frame.buffer) })} #[cfg(target_os = "macos")]pub type RemoteVideoFrame = media::core_video::CVImageBuffer; #[cfg(target_os = "macos")]fn video_frame_buffer_from_webrtc(buffer: Box<dyn VideoBuffer>) -> Option<RemoteVideoFrame> {    use core_foundation::base::TCFType as _;    use media::core_video::CVImageBuffer;     let buffer = buffer.as_native()?;    let pixel_buffer = buffer.get_cv_pixel_buffer();    if pixel_buffer.is_null() {        return None;    }     unsafe { Some(CVImageBuffer::wrap_under_get_rule(pixel_buffer as _)) }} #[cfg(not(target_os = "macos"))]pub type RemoteVideoFrame = Arc<gpui::RenderImage>; #[cfg(not(any(target_os = "macos", all(target_os = "windows", target_env = "gnu"))))]fn video_frame_buffer_from_webrtc(buffer: Box<dyn VideoBuffer>) -> Option<RemoteVideoFrame> {    use gpui::RenderImage;    use image::{Frame, RgbaImage};    use livekit::webrtc::prelude::VideoFormatType;    use smallvec::SmallVec;    use std::alloc::{alloc, Layout};     let width = buffer.width();    let height = buffer.height();    let stride = width * 4;    let byte_len = (stride * height) as usize;    let argb_image = unsafe {        // Motivation for this unsafe code is to avoid initializing the frame data, since to_argb

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