From 17ca0a7918d84b77bedd25e99f990ae16439d081 Mon Sep 17 00:00:00 2001 From: Chiu-Hsiang Hsu Date: Mon, 25 Oct 2021 12:42:31 +0800 Subject: [PATCH] update to use re-export --- Cargo.toml | 7 +------ examples/broadcast/broadcast.rs | 4 ++-- .../data-channels-close.rs | 2 +- .../data-channels-create.rs | 2 +- .../data-channels-detach-create.rs | 6 +++--- .../data-channels-detach.rs | 6 +++--- .../data-channels-flow-control.rs | 2 +- examples/data-channels/data-channels.rs | 2 +- examples/ice-restart/ice-restart.rs | 2 +- .../insertable-streams/insertable-streams.rs | 6 +++--- examples/offer-answer/answer.rs | 2 +- examples/offer-answer/offer.rs | 2 +- .../play-from-disk-h264.rs | 8 ++++---- .../play-from-disk-renegotiation.rs | 6 +++--- .../play-from-disk-vp8/play-from-disk-vp8.rs | 8 ++++---- .../play-from-disk-vp9/play-from-disk-vp9.rs | 8 ++++---- examples/reflect/reflect.rs | 4 ++-- examples/rtp-forwarder/rtp-forwarder.rs | 6 +++--- examples/rtp-to-webrtc/rtp-to-webrtc.rs | 2 +- .../save-to-disk-h264/save-to-disk-h264.rs | 18 ++++++++--------- examples/save-to-disk-vpx/save-to-disk-vpx.rs | 20 +++++++++---------- examples/simulcast/simulcast.rs | 4 ++-- examples/swap-tracks/swap-tracks.rs | 7 ++++--- 23 files changed, 65 insertions(+), 69 deletions(-) diff --git a/Cargo.toml b/Cargo.toml index 01cf718..7c9822f 100644 --- a/Cargo.toml +++ b/Cargo.toml @@ -16,12 +16,7 @@ repository = "https://github.com/webrtc-rs/examples" [dev-dependencies] util = { package = "webrtc-util", version = "0.5.0" } -rtp = "0.6.0" -rtcp = "0.5.0" -data = { package = "webrtc-data", version = "0.3.0" } -interceptor = "0.4.0" -media = { package = "webrtc-media", version = "0.4.0" } -webrtc = "0.2.1" #{ path = "..", version = "0.2.1" } # +webrtc = "0.2.2" tokio = { version = "1.12.0", features = ["full"] } env_logger = "0.9.0" clap = "2" diff --git a/examples/broadcast/broadcast.rs b/examples/broadcast/broadcast.rs index 2cb3df4..36ccd10 100644 --- a/examples/broadcast/broadcast.rs +++ b/examples/broadcast/broadcast.rs @@ -1,13 +1,12 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; -use rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; use std::io::Write; use std::sync::Arc; use tokio::time::Duration; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::MediaEngine; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::RTPCodecType; use webrtc::media::rtp::rtp_receiver::RTCRtpReceiver; use webrtc::media::track::track_local::track_local_static_rtp::TrackLocalStaticRTP; @@ -17,6 +16,7 @@ use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; use webrtc::Error; #[tokio::main] diff --git a/examples/data-channels-close/data-channels-close.rs b/examples/data-channels-close/data-channels-close.rs index e3b4637..906395f 100644 --- a/examples/data-channels-close/data-channels-close.rs +++ b/examples/data-channels-close/data-channels-close.rs @@ -1,6 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; use std::io::Write; use std::sync::atomic::{AtomicI32, Ordering}; use std::sync::Arc; @@ -11,6 +10,7 @@ use webrtc::api::media_engine::MediaEngine; use webrtc::api::APIBuilder; use webrtc::data::data_channel::data_channel_message::DataChannelMessage; use webrtc::data::data_channel::RTCDataChannel; +use webrtc::interceptor::registry::Registry; use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; diff --git a/examples/data-channels-create/data-channels-create.rs b/examples/data-channels-create/data-channels-create.rs index 721aa3c..e3634e4 100644 --- a/examples/data-channels-create/data-channels-create.rs +++ b/examples/data-channels-create/data-channels-create.rs @@ -1,6 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; use std::io::Write; use std::sync::Arc; use tokio::time::Duration; @@ -8,6 +7,7 @@ use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::MediaEngine; use webrtc::api::APIBuilder; use webrtc::data::data_channel::data_channel_message::DataChannelMessage; +use webrtc::interceptor::registry::Registry; use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; diff --git a/examples/data-channels-detach-create/data-channels-detach-create.rs b/examples/data-channels-detach-create/data-channels-detach-create.rs index 4a8899a..ad07c99 100644 --- a/examples/data-channels-detach-create/data-channels-detach-create.rs +++ b/examples/data-channels-detach-create/data-channels-detach-create.rs @@ -1,7 +1,6 @@ use anyhow::Result; use bytes::Bytes; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; use std::io::Write; use std::sync::Arc; use tokio::time::Duration; @@ -9,6 +8,7 @@ use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::MediaEngine; use webrtc::api::setting_engine::SettingEngine; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; @@ -197,7 +197,7 @@ async fn main() -> Result<()> { } // read_loop shows how to read from the datachannel directly -async fn read_loop(d: Arc) -> Result<()> { +async fn read_loop(d: Arc) -> Result<()> { let mut buffer = vec![0u8; MESSAGE_SIZE]; loop { let n = match d.read(&mut buffer).await { @@ -216,7 +216,7 @@ async fn read_loop(d: Arc) -> Result<()> { } // write_loop shows how to write to the datachannel directly -async fn write_loop(d: Arc) -> Result<()> { +async fn write_loop(d: Arc) -> Result<()> { let mut result = Result::::Ok(0); while result.is_ok() { let timeout = tokio::time::sleep(Duration::from_secs(5)); diff --git a/examples/data-channels-detach/data-channels-detach.rs b/examples/data-channels-detach/data-channels-detach.rs index 11f8596..1cce9b5 100644 --- a/examples/data-channels-detach/data-channels-detach.rs +++ b/examples/data-channels-detach/data-channels-detach.rs @@ -1,7 +1,6 @@ use anyhow::Result; use bytes::Bytes; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; use std::io::Write; use std::sync::Arc; use tokio::time::Duration; @@ -10,6 +9,7 @@ use webrtc::api::media_engine::MediaEngine; use webrtc::api::setting_engine::SettingEngine; use webrtc::api::APIBuilder; use webrtc::data::data_channel::RTCDataChannel; +use webrtc::interceptor::registry::Registry; use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; @@ -206,7 +206,7 @@ async fn main() -> Result<()> { } // read_loop shows how to read from the datachannel directly -async fn read_loop(d: Arc) -> Result<()> { +async fn read_loop(d: Arc) -> Result<()> { let mut buffer = vec![0u8; MESSAGE_SIZE]; loop { let n = match d.read(&mut buffer).await { @@ -225,7 +225,7 @@ async fn read_loop(d: Arc) -> Result<()> { } // write_loop shows how to write to the datachannel directly -async fn write_loop(d: Arc) -> Result<()> { +async fn write_loop(d: Arc) -> Result<()> { let mut result = Result::::Ok(0); while result.is_ok() { let timeout = tokio::time::sleep(Duration::from_secs(5)); diff --git a/examples/data-channels-flow-control/data-channels-flow-control.rs b/examples/data-channels-flow-control/data-channels-flow-control.rs index 06c0998..060cb6c 100644 --- a/examples/data-channels-flow-control/data-channels-flow-control.rs +++ b/examples/data-channels-flow-control/data-channels-flow-control.rs @@ -1,7 +1,6 @@ use anyhow::Result; use bytes::Bytes; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; use std::io::Write; use std::sync::atomic::{AtomicUsize, Ordering}; use std::sync::Arc; @@ -13,6 +12,7 @@ use webrtc::api::APIBuilder; use webrtc::data::data_channel::data_channel_init::RTCDataChannelInit; use webrtc::data::data_channel::data_channel_message::DataChannelMessage; use webrtc::data::data_channel::RTCDataChannel; +use webrtc::interceptor::registry::Registry; use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_candidate::RTCIceCandidate; use webrtc::peer::ice::ice_server::RTCIceServer; diff --git a/examples/data-channels/data-channels.rs b/examples/data-channels/data-channels.rs index 1ea0f0b..f4aa14b 100644 --- a/examples/data-channels/data-channels.rs +++ b/examples/data-channels/data-channels.rs @@ -1,6 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; use std::io::Write; use std::sync::Arc; use tokio::time::Duration; @@ -9,6 +8,7 @@ use webrtc::api::media_engine::MediaEngine; use webrtc::api::APIBuilder; use webrtc::data::data_channel::data_channel_message::DataChannelMessage; use webrtc::data::data_channel::RTCDataChannel; +use webrtc::interceptor::registry::Registry; use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; diff --git a/examples/ice-restart/ice-restart.rs b/examples/ice-restart/ice-restart.rs index a6e8dca..2351d44 100644 --- a/examples/ice-restart/ice-restart.rs +++ b/examples/ice-restart/ice-restart.rs @@ -2,7 +2,6 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; use hyper::service::{make_service_fn, service_fn}; use hyper::{Body, Method, Request, Response, Server, StatusCode}; -use interceptor::registry::Registry; use std::io::Write; use std::net::SocketAddr; use std::str::FromStr; @@ -14,6 +13,7 @@ use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::MediaEngine; use webrtc::api::APIBuilder; use webrtc::data::data_channel::RTCDataChannel; +use webrtc::interceptor::registry::Registry; use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_connection_state::RTCIceConnectionState; use webrtc::peer::peer_connection::RTCPeerConnection; diff --git a/examples/insertable-streams/insertable-streams.rs b/examples/insertable-streams/insertable-streams.rs index 7ded0aa..fb1a469 100644 --- a/examples/insertable-streams/insertable-streams.rs +++ b/examples/insertable-streams/insertable-streams.rs @@ -1,8 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; -use media::io::ivf_reader::IVFReader; -use media::Sample; use std::fs::File; use std::io::BufReader; use std::io::Write; @@ -13,6 +10,7 @@ use tokio::time::Duration; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_VP8}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::RTCRtpCodecCapability; use webrtc::media::track::track_local::track_local_static_sample::TrackLocalStaticSample; use webrtc::media::track::track_local::TrackLocal; @@ -21,6 +19,8 @@ use webrtc::peer::ice::ice_connection_state::RTCIceConnectionState; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::webrtc_media::io::ivf_reader::IVFReader; +use webrtc::webrtc_media::Sample; use webrtc::Error; const CIPHER_KEY: u8 = 0xAA; diff --git a/examples/offer-answer/answer.rs b/examples/offer-answer/answer.rs index bc69add..5c22729 100644 --- a/examples/offer-answer/answer.rs +++ b/examples/offer-answer/answer.rs @@ -2,7 +2,6 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; use hyper::service::{make_service_fn, service_fn}; use hyper::{Body, Client, Method, Request, Response, Server, StatusCode}; -use interceptor::registry::Registry; use std::io::Write; use std::net::SocketAddr; use std::str::FromStr; @@ -14,6 +13,7 @@ use webrtc::api::media_engine::MediaEngine; use webrtc::api::APIBuilder; use webrtc::data::data_channel::data_channel_message::DataChannelMessage; use webrtc::data::data_channel::RTCDataChannel; +use webrtc::interceptor::registry::Registry; use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_candidate::{RTCIceCandidate, RTCIceCandidateInit}; use webrtc::peer::ice::ice_server::RTCIceServer; diff --git a/examples/offer-answer/offer.rs b/examples/offer-answer/offer.rs index 7588ad9..57d2742 100644 --- a/examples/offer-answer/offer.rs +++ b/examples/offer-answer/offer.rs @@ -2,7 +2,6 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; use hyper::service::{make_service_fn, service_fn}; use hyper::{Body, Client, Method, Request, Response, Server, StatusCode}; -use interceptor::registry::Registry; use std::io::Write; use std::net::SocketAddr; use std::str::FromStr; @@ -13,6 +12,7 @@ use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::MediaEngine; use webrtc::api::APIBuilder; use webrtc::data::data_channel::data_channel_message::DataChannelMessage; +use webrtc::interceptor::registry::Registry; use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_candidate::{RTCIceCandidate, RTCIceCandidateInit}; use webrtc::peer::ice::ice_server::RTCIceServer; diff --git a/examples/play-from-disk-h264/play-from-disk-h264.rs b/examples/play-from-disk-h264/play-from-disk-h264.rs index d760707..74d1764 100644 --- a/examples/play-from-disk-h264/play-from-disk-h264.rs +++ b/examples/play-from-disk-h264/play-from-disk-h264.rs @@ -1,9 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; -use media::io::h264_reader::H264Reader; -use media::io::ogg_reader::OggReader; -use media::Sample; use std::fs::File; use std::io::BufReader; use std::io::Write; @@ -14,6 +10,7 @@ use tokio::time::Duration; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_H264, MIME_TYPE_OPUS}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::RTCRtpCodecCapability; use webrtc::media::track::track_local::track_local_static_sample::TrackLocalStaticSample; use webrtc::media::track::track_local::TrackLocal; @@ -22,6 +19,9 @@ use webrtc::peer::ice::ice_connection_state::RTCIceConnectionState; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::webrtc_media::io::h264_reader::H264Reader; +use webrtc::webrtc_media::io::ogg_reader::OggReader; +use webrtc::webrtc_media::Sample; use webrtc::Error; #[tokio::main] diff --git a/examples/play-from-disk-renegotiation/play-from-disk-renegotiation.rs b/examples/play-from-disk-renegotiation/play-from-disk-renegotiation.rs index 50226c8..fb7b907 100644 --- a/examples/play-from-disk-renegotiation/play-from-disk-renegotiation.rs +++ b/examples/play-from-disk-renegotiation/play-from-disk-renegotiation.rs @@ -2,9 +2,6 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; use hyper::service::{make_service_fn, service_fn}; use hyper::{Body, Method, Request, Response, Server, StatusCode}; -use interceptor::registry::Registry; -use media::io::ivf_reader::IVFReader; -use media::Sample; use std::fs::File; use std::io::BufReader; use std::io::Write; @@ -18,6 +15,7 @@ use tokio_util::codec::{BytesCodec, FramedRead}; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_VP8}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::RTCRtpCodecCapability; use webrtc::media::track::track_local::track_local_static_sample::TrackLocalStaticSample; use webrtc::media::track::track_local::TrackLocal; @@ -26,6 +24,8 @@ use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection::RTCPeerConnection; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::webrtc_media::io::ivf_reader::IVFReader; +use webrtc::webrtc_media::Sample; use webrtc::Error; #[macro_use] diff --git a/examples/play-from-disk-vp8/play-from-disk-vp8.rs b/examples/play-from-disk-vp8/play-from-disk-vp8.rs index 20f43d7..d491e02 100644 --- a/examples/play-from-disk-vp8/play-from-disk-vp8.rs +++ b/examples/play-from-disk-vp8/play-from-disk-vp8.rs @@ -1,9 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; -use media::io::ivf_reader::IVFReader; -use media::io::ogg_reader::OggReader; -use media::Sample; use std::fs::File; use std::io::BufReader; use std::io::Write; @@ -14,6 +10,7 @@ use tokio::time::Duration; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_OPUS, MIME_TYPE_VP8}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::RTCRtpCodecCapability; use webrtc::media::track::track_local::track_local_static_sample::TrackLocalStaticSample; use webrtc::media::track::track_local::TrackLocal; @@ -22,6 +19,9 @@ use webrtc::peer::ice::ice_connection_state::RTCIceConnectionState; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::webrtc_media::io::ivf_reader::IVFReader; +use webrtc::webrtc_media::io::ogg_reader::OggReader; +use webrtc::webrtc_media::Sample; use webrtc::Error; #[tokio::main] diff --git a/examples/play-from-disk-vp9/play-from-disk-vp9.rs b/examples/play-from-disk-vp9/play-from-disk-vp9.rs index 537349c..6bcce3e 100644 --- a/examples/play-from-disk-vp9/play-from-disk-vp9.rs +++ b/examples/play-from-disk-vp9/play-from-disk-vp9.rs @@ -1,9 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; -use media::io::ivf_reader::IVFReader; -use media::io::ogg_reader::OggReader; -use media::Sample; use std::fs::File; use std::io::BufReader; use std::io::Write; @@ -14,6 +10,7 @@ use tokio::time::Duration; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_OPUS, MIME_TYPE_VP9}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::RTCRtpCodecCapability; use webrtc::media::track::track_local::track_local_static_sample::TrackLocalStaticSample; use webrtc::media::track::track_local::TrackLocal; @@ -22,6 +19,9 @@ use webrtc::peer::ice::ice_connection_state::RTCIceConnectionState; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::webrtc_media::io::ivf_reader::IVFReader; +use webrtc::webrtc_media::io::ogg_reader::OggReader; +use webrtc::webrtc_media::Sample; use webrtc::Error; #[tokio::main] diff --git a/examples/reflect/reflect.rs b/examples/reflect/reflect.rs index b638897..03b8d7d 100644 --- a/examples/reflect/reflect.rs +++ b/examples/reflect/reflect.rs @@ -1,7 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; -use rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; use std::collections::HashMap; use std::io::Write; use std::sync::Arc; @@ -9,6 +7,7 @@ use tokio::time::Duration; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_OPUS, MIME_TYPE_VP8}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::{RTCRtpCodecCapability, RTCRtpCodecParameters, RTPCodecType}; use webrtc::media::rtp::rtp_receiver::RTCRtpReceiver; use webrtc::media::track::track_local::track_local_static_rtp::TrackLocalStaticRTP; @@ -18,6 +17,7 @@ use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; #[tokio::main] async fn main() -> Result<()> { diff --git a/examples/rtp-forwarder/rtp-forwarder.rs b/examples/rtp-forwarder/rtp-forwarder.rs index 371657f..7416091 100644 --- a/examples/rtp-forwarder/rtp-forwarder.rs +++ b/examples/rtp-forwarder/rtp-forwarder.rs @@ -1,7 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; -use rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; use std::collections::HashMap; use std::io::Write; use std::sync::Arc; @@ -11,6 +9,7 @@ use util::{Conn, Marshal, Unmarshal}; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_OPUS, MIME_TYPE_VP8}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::{RTCRtpCodecCapability, RTCRtpCodecParameters, RTPCodecType}; use webrtc::media::rtp::rtp_receiver::RTCRtpReceiver; use webrtc::media::track::track_remote::TrackRemote; @@ -19,6 +18,7 @@ use webrtc::peer::ice::ice_connection_state::RTCIceConnectionState; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; #[derive(Clone)] struct UdpConn { @@ -213,7 +213,7 @@ async fn main() -> Result<()> { while let Ok((n, _)) = track.read(&mut b).await { // Unmarshal the packet and update the PayloadType let mut buf = &b[..n]; - let mut rtp_packet = rtp::packet::Packet::unmarshal(&mut buf)?; + let mut rtp_packet = webrtc::rtp::packet::Packet::unmarshal(&mut buf)?; rtp_packet.header.payload_type = c.payload_type; // Marshal into original buffer with updated PayloadType diff --git a/examples/rtp-to-webrtc/rtp-to-webrtc.rs b/examples/rtp-to-webrtc/rtp-to-webrtc.rs index f651880..1a0fcab 100644 --- a/examples/rtp-to-webrtc/rtp-to-webrtc.rs +++ b/examples/rtp-to-webrtc/rtp-to-webrtc.rs @@ -1,12 +1,12 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; use std::io::Write; use std::sync::Arc; use tokio::net::UdpSocket; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_VP8}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::RTCRtpCodecCapability; use webrtc::media::track::track_local::track_local_static_rtp::TrackLocalStaticRTP; use webrtc::media::track::track_local::{TrackLocal, TrackLocalWriter}; diff --git a/examples/save-to-disk-h264/save-to-disk-h264.rs b/examples/save-to-disk-h264/save-to-disk-h264.rs index d36942d..b00635e 100644 --- a/examples/save-to-disk-h264/save-to-disk-h264.rs +++ b/examples/save-to-disk-h264/save-to-disk-h264.rs @@ -1,9 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; -use media::io::h264_writer::H264Writer; -use media::io::ogg_writer::OggWriter; -use rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; use std::fs::File; use std::io::Write; use std::sync::Arc; @@ -12,6 +8,7 @@ use tokio::time::Duration; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_H264, MIME_TYPE_OPUS}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::{RTCRtpCodecCapability, RTCRtpCodecParameters, RTPCodecType}; use webrtc::media::rtp::rtp_receiver::RTCRtpReceiver; use webrtc::media::track::track_remote::TrackRemote; @@ -19,9 +16,12 @@ use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_connection_state::RTCIceConnectionState; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; +use webrtc::webrtc_media::io::h264_writer::H264Writer; +use webrtc::webrtc_media::io::ogg_writer::OggWriter; async fn save_to_disk( - writer: Arc>, + writer: Arc>, track: Arc, notify: Arc, ) -> Result<()> { @@ -118,11 +118,11 @@ async fn main() -> Result<()> { let video_file = matches.value_of("video").unwrap(); let audio_file = matches.value_of("audio").unwrap(); - let h264_writer: Arc> = + let h264_writer: Arc> = Arc::new(Mutex::new(H264Writer::new(File::create(video_file)?))); - let ogg_writer: Arc> = Arc::new(Mutex::new( - OggWriter::new(File::create(audio_file)?, 48000, 2)?, - )); + let ogg_writer: Arc> = Arc::new( + Mutex::new(OggWriter::new(File::create(audio_file)?, 48000, 2)?), + ); // Everything below is the WebRTC-rs API! Thanks for using it ❤️. diff --git a/examples/save-to-disk-vpx/save-to-disk-vpx.rs b/examples/save-to-disk-vpx/save-to-disk-vpx.rs index 8d9bf89..117804f 100644 --- a/examples/save-to-disk-vpx/save-to-disk-vpx.rs +++ b/examples/save-to-disk-vpx/save-to-disk-vpx.rs @@ -1,10 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; -use media::io::ivf_reader::IVFFileHeader; -use media::io::ivf_writer::IVFWriter; -use media::io::ogg_writer::OggWriter; -use rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; use std::fs::File; use std::io::Write; use std::sync::Arc; @@ -13,6 +8,7 @@ use tokio::time::Duration; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_OPUS, MIME_TYPE_VP8, MIME_TYPE_VP9}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::{RTCRtpCodecCapability, RTCRtpCodecParameters, RTPCodecType}; use webrtc::media::rtp::rtp_receiver::RTCRtpReceiver; use webrtc::media::track::track_remote::TrackRemote; @@ -20,9 +16,13 @@ use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_connection_state::RTCIceConnectionState; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; +use webrtc::webrtc_media::io::ivf_reader::IVFFileHeader; +use webrtc::webrtc_media::io::ivf_writer::IVFWriter; +use webrtc::webrtc_media::io::ogg_writer::OggWriter; async fn save_to_disk( - writer: Arc>, + writer: Arc>, track: Arc, notify: Arc, ) -> Result<()> { @@ -125,7 +125,7 @@ async fn main() -> Result<()> { let video_file = matches.value_of("video").unwrap(); let audio_file = matches.value_of("audio").unwrap(); - let ivf_writer: Arc> = + let ivf_writer: Arc> = Arc::new(Mutex::new(IVFWriter::new( File::create(video_file)?, &IVFFileHeader { @@ -141,9 +141,9 @@ async fn main() -> Result<()> { unused: 0, // 28-31 }, )?)); - let ogg_writer: Arc> = Arc::new(Mutex::new( - OggWriter::new(File::create(audio_file)?, 48000, 2)?, - )); + let ogg_writer: Arc> = Arc::new( + Mutex::new(OggWriter::new(File::create(audio_file)?, 48000, 2)?), + ); // Everything below is the WebRTC-rs API! Thanks for using it ❤️. diff --git a/examples/simulcast/simulcast.rs b/examples/simulcast/simulcast.rs index 4f3b875..8bf4aa3 100644 --- a/examples/simulcast/simulcast.rs +++ b/examples/simulcast/simulcast.rs @@ -1,7 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; -use rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; use std::collections::HashMap; use std::io::Write; use std::sync::Arc; @@ -9,6 +7,7 @@ use tokio::time::Duration; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_VP8}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::{ RTCRtpCodecCapability, RTCRtpHeaderExtensionCapability, RTPCodecType, }; @@ -20,6 +19,7 @@ use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; use webrtc::Error; #[tokio::main] diff --git a/examples/swap-tracks/swap-tracks.rs b/examples/swap-tracks/swap-tracks.rs index 2fca6e1..d1000ee 100644 --- a/examples/swap-tracks/swap-tracks.rs +++ b/examples/swap-tracks/swap-tracks.rs @@ -1,7 +1,5 @@ use anyhow::Result; use clap::{App, AppSettings, Arg}; -use interceptor::registry::Registry; -use rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; use std::io::Write; use std::sync::atomic::{AtomicBool, AtomicUsize, Ordering}; use std::sync::Arc; @@ -9,6 +7,7 @@ use tokio::time::Duration; use webrtc::api::interceptor_registry::register_default_interceptors; use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_VP8}; use webrtc::api::APIBuilder; +use webrtc::interceptor::registry::Registry; use webrtc::media::rtp::rtp_codec::RTCRtpCodecCapability; use webrtc::media::rtp::rtp_receiver::RTCRtpReceiver; use webrtc::media::track::track_local::track_local_static_rtp::TrackLocalStaticRTP; @@ -18,6 +17,7 @@ use webrtc::peer::configuration::RTCConfiguration; use webrtc::peer::ice::ice_server::RTCIceServer; use webrtc::peer::peer_connection_state::RTCPeerConnectionState; use webrtc::peer::sdp::session_description::RTCSessionDescription; +use webrtc::rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication; use webrtc::Error; #[tokio::main] @@ -136,7 +136,8 @@ async fn main() -> Result<()> { // The total number of tracks let track_count = Arc::new(AtomicUsize::new(0)); // The channel of packets with a bit of buffer - let (packets_tx, mut packets_rx) = tokio::sync::mpsc::channel::(60); + let (packets_tx, mut packets_rx) = + tokio::sync::mpsc::channel::(60); let packets_tx = Arc::new(packets_tx); // Set a handler for when a new remote track starts, this handler copies inbound RTP packets,