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/webrtc/media/
H A DREADME.md6 <a href="https://github.com/webrtc-rs/media/actions">
7 <img src="https://github.com/webrtc-rs/media/workflows/cargo/badge.svg">
9 <a href="https://codecov.io/gh/webrtc-rs/media">
10 <img src="https://codecov.io/gh/webrtc-rs/media/branch/main/graph/badge.svg">
12 <a href="https://deps.rs/repo/github/webrtc-rs/media">
13 <img src="https://deps.rs/repo/github/webrtc-rs/media/status.svg">
15 <a href="https://crates.io/crates/webrtc-media">
16 <img src="https://img.shields.io/crates/v/webrtc-media.svg">
18 <a href="https://docs.rs/webrtc-media">
19 <img src="https://docs.rs/webrtc-media/badge.svg">
H A DCargo.toml2 name = "webrtc-media"
8 documentation = "https://docs.rs/webrtc-media"
10 repository = "https://github.com/webrtc-rs/media"
H A DCHANGELOG.md1 # webrtc-media changelog
7 …his `SampleBuilder` would sometimes, incorrectly, drop packets that carry media when they appeared…
13 …entiate inconsequential padding packets being dropped from those carrying media. Contributed by [@…
23 …s, but not always, available in the [GitHub Releases](https://github.com/webrtc-rs/media/releases).
/webrtc/webrtc/src/peer_connection/sdp/
H A Dmod.rs93 let codec_type = RTPCodecType::from(media.media_name.media.as_str()); in track_details_from_sdp()
344 media = media.with_fingerprint(f.algorithm.clone(), f.value.to_uppercase()); in add_data_media_section()
444 media = media.with_codec( in add_transceiver_sdp()
453 media = media.with_value_attribute( in add_transceiver_sdp()
508 media = media.with_extmap(sdp::extmap::ExtMap { in add_transceiver_sdp()
519 media = in add_transceiver_sdp()
524 media = media.with_value_attribute( in add_transceiver_sdp()
533 media = media.with_media_source( in add_transceiver_sdp()
612 media = media.with_property_attribute(direction.to_string()); in add_transceiver_sdp()
615 media = media.with_fingerprint( in add_transceiver_sdp()
[all …]
H A Dsdp_test.rs287 media: "audio".to_owned(), in test_track_details_from_sdp()
308 media: "video".to_owned(), in test_track_details_from_sdp()
483 media: "audio".to_owned(), in test_have_application_media_section()
526 media: &[MediaSection], in fingerprint_test()
547 media, in fingerprint_test()
595 let media = vec![ in test_media_description_fingerprints() localVariable
954 if desc.media_name.media != "audio" { in test_populate_sdp_reject()
986 media: "video".to_owned(), in test_get_rids()
1015 media: "audio".to_owned(), in test_codecs_from_media_description()
1046 media: "audio".to_owned(), in test_codecs_from_media_description()
[all …]
/webrtc/examples/examples/save-to-disk-vpx/
H A Dsave-to-disk-vpx.rs14 use webrtc::media::io::ivf_reader::IVFFileHeader;
15 use webrtc::media::io::ivf_writer::IVFWriter;
16 use webrtc::media::io::ogg_writer::OggWriter;
26 writer: Arc<Mutex<dyn webrtc::media::io::Writer + Send + Sync>>, in save_to_disk()
129 let ivf_writer: Arc<Mutex<dyn webrtc::media::io::Writer + Send + Sync>> = in main()
145 let ogg_writer: Arc<Mutex<dyn webrtc::media::io::Writer + Send + Sync>> = Arc::new(Mutex::new( in main()
/webrtc/webrtc/
H A DCargo.toml22 media = { version = "0.6.0", path = "../media", package = "webrtc-media" }
/webrtc/sdp/src/description/
H A Dmod.rs5 pub mod media; module
H A Dmedia.rs77 media: codec_type, in new_jsep_media_description()
225 pub media: String, field
234 self.media.clone(), in fmt()
H A Ddescription_test.rs2 use super::media::*;
125 media: "audio".to_string(), in test_marshal()
153 media: "video".to_string(), in test_marshal()
/webrtc/sdp/src/
H A Dlib.rs12 pub use description::{media::MediaDescription, session::SessionDescription};
/webrtc/examples/examples/save-to-disk-h264/
H A Dsave-to-disk-h264.rs14 use webrtc::media::io::h264_writer::H264Writer;
15 use webrtc::media::io::ogg_writer::OggWriter;
25 writer: Arc<Mutex<dyn webrtc::media::io::Writer + Send + Sync>>, in save_to_disk()
122 let h264_writer: Arc<Mutex<dyn webrtc::media::io::Writer + Send + Sync>> = in main()
124 let ogg_writer: Arc<Mutex<dyn webrtc::media::io::Writer + Send + Sync>> = Arc::new(Mutex::new( in main()
/webrtc/webrtc/src/peer_connection/policy/
H A Dsdp_semantics.rs63 use sdp::description::media::MediaDescription;
89 .map(|md| md.media_name.media.clone()) in get_md_names()
/webrtc/
H A DCargo.toml10 "media",
/webrtc/webrtc/src/peer_connection/
H A Dmod.rs1227 for media in &parsed.media_descriptions { in set_local_description()
1228 if media.media_name.media == MEDIA_SECTION_APPLICATION { in set_local_description()
1232 let kind = RTPCodecType::from(media.media_name.media.as_str()); in set_local_description()
1233 let direction = get_peer_direction(media); in set_local_description()
1240 let mid_value = match get_mid_value(media) { in set_local_description()
1326 for media in &parsed.media_descriptions { in set_remote_description()
1338 if media.media_name.media == MEDIA_SECTION_APPLICATION { in set_remote_description()
1342 let kind = RTPCodecType::from(media.media_name.media.as_str()); in set_remote_description()
1419 for media in &parsed.media_descriptions { in set_remote_description()
1431 if media.media_name.media == MEDIA_SECTION_APPLICATION { in set_remote_description()
[all …]
H A Dpeer_connection_internal.rs733 for media in &parsed.media_descriptions { in generate_matched_sdp()
734 if let Some(mid_value) = get_mid_value(media) { in generate_matched_sdp()
739 if media.media_name.media == MEDIA_SECTION_APPLICATION { in generate_matched_sdp()
749 let kind = RTPCodecType::from(media.media_name.media.as_str()); in generate_matched_sdp()
750 let direction = get_peer_direction(media); in generate_matched_sdp()
768 rid_map: get_rids(media), in generate_matched_sdp()
875 if only_media_section.media_name.media == RTPCodecType::Audio.to_string() { in handle_undeclared_ssrc()
/webrtc/examples/examples/reflect/
H A Dreflect.rs148 let mut media = vec![]; in main() localVariable
150 media.push("audio"); in main()
153 media.push("video"); in main()
155 for s in media { in main()
/webrtc/webrtc/src/
H A Dlib.rs10 pub use media;
/webrtc/sdp/src/util/
H A Dutil_test.rs3 use crate::description::media::*;
11 media: "video".to_string(), in get_test_session_description()
/webrtc/webrtc/src/api/media_engine/
H A Dmod.rs598 for media in &desc.media_descriptions { in update_from_remote_description()
600 && media.media_name.media.to_lowercase() == "audio" in update_from_remote_description()
605 && media.media_name.media.to_lowercase() == "video" in update_from_remote_description()
613 let codecs = codecs_from_media_description(media)?; in update_from_remote_description()
639 let extensions = rtp_extensions_from_media_description(media)?; in update_from_remote_description()
/webrtc/webrtc/src/track/track_local/
H A Dtrack_local_static_sample.rs7 use media::Sample;
242 use media::Sample;
/webrtc/examples/examples/play-from-disk-h264/
H A Dplay-from-disk-h264.rs16 use webrtc::media::io::h264_reader::H264Reader;
17 use webrtc::media::io::ogg_reader::OggReader;
18 use webrtc::media::Sample;
/webrtc/examples/examples/play-from-disk-vpx/
H A Dplay-from-disk-vpx.rs16 use webrtc::media::io::ivf_reader::IVFReader;
17 use webrtc::media::io::ogg_reader::OggReader;
18 use webrtc::media::Sample;
/webrtc/examples/examples/insertable-streams/
H A Dinsertable-streams.rs16 use webrtc::media::io::ivf_reader::IVFReader;
17 use webrtc::media::Sample;
/webrtc/examples/examples/rtp-forwarder/
H A DREADME.md36 You can run any of these commands at anytime. The media is live/stateless, you can switch commands …

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