| /webrtc/rtcp/src/payload_feedbacks/receiver_estimated_maximum_bitrate/ |
| H A D | receiver_estimated_maximum_bitrate_test.rs | 37 assert_eq!(packet, expected); in test_receiver_estimated_maximum_bitrate_unmarshal()
55 assert_eq!(packet.bitrate, 8927168.0); in test_receiver_estimated_maximum_bitrate_truncate()
58 let output = packet.marshal().unwrap(); in test_receiver_estimated_maximum_bitrate_truncate()
62 packet.bitrate -= 1.0; in test_receiver_estimated_maximum_bitrate_truncate()
68 let mut output = packet.marshal().unwrap(); in test_receiver_estimated_maximum_bitrate_truncate()
81 assert_eq!(8927104.0, packet.bitrate); in test_receiver_estimated_maximum_bitrate_truncate()
87 let packet = ReceiverEstimatedMaximumBitrate { in test_receiver_estimated_maximum_bitrate_overflow() localVariable
99 let output = packet.marshal().unwrap(); in test_receiver_estimated_maximum_bitrate_overflow()
108 assert_eq!(packet.bitrate, f32::from_bits(0x67FFFFC0)); in test_receiver_estimated_maximum_bitrate_overflow()
111 let output = packet.marshal().unwrap(); in test_receiver_estimated_maximum_bitrate_overflow()
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| /webrtc/util/src/buffer/ |
| H A D | buffer_test.rs | 11 let mut packet: Vec<u8> = vec![0; 4]; in test_buffer() localVariable
20 assert_eq!(&packet[..n], &[0, 1]); in test_buffer()
37 assert_eq!(&packet[..n], &[2, 3, 4]); in test_buffer()
41 assert_eq!(&packet[..n], &[5, 6, 7]); in test_buffer()
57 assert_eq!(&packet[..n], &[3]); in test_buffer()
190 assert_eq!(&packet[..n], &[0, 1]); in test_buffer_limit_count()
209 assert_eq!(&packet[..n], &[2, 3]); in test_buffer_limit_count()
214 assert_eq!(&packet[..n], &[6, 7]); in test_buffer_limit_count()
253 assert_eq!(&packet[..n], &[0, 1]); in test_buffer_limit_size()
272 assert_eq!(&packet[..n], &[2, 3]); in test_buffer_limit_size()
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| H A D | mod.rs | 126 pub async fn write(&self, packet: &[u8]) -> Result<usize> { in write()
127 if packet.len() >= 0x10000 { in write()
144 while !b.available(packet.len()) { in write()
150 b.data[tail] = (packet.len() >> 8) as u8; in write()
157 b.data[tail] = packet.len() as u8; in write()
167 b.data[tail..end].copy_from_slice(&packet[..n]); in write()
171 let m = packet.len() - n; in write()
172 b.data[..m].copy_from_slice(&packet[n..]); in write()
183 Ok(packet.len()) in write()
212 if copied > packet.len() { in read()
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| /webrtc/interceptor/src/nack/responder/ |
| H A D | responder_stream.rs | 10 packets: Vec<Option<rtp::packet::Packet>>,
26 fn add(&mut self, packet: &rtp::packet::Packet) { in add()
27 let seq = packet.header.sequence_number; in add()
50 fn get(&self, seq: u16) -> Option<&rtp::packet::Packet> { in get()
77 async fn add(&self, pkt: &rtp::packet::Packet) { in add()
115 sb.add(&rtp::packet::Packet { in test_responder_stream()
128 if let Some(packet) = sb.get(seq) { in test_responder_stream()
130 packet.header.sequence_number, seq, in test_responder_stream()
132 seq, packet.header.sequence_number in test_responder_stream()
143 if let Some(packet) = sb.get(seq) { in test_responder_stream()
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| /webrtc/srtp/src/session/ |
| H A D | session_rtp_test.rs | 87 let packet = rtp::packet::Packet { in test_session_srtp_accept() localVariable
94 sa.write_rtp(&packet).await?; in test_session_srtp_accept()
126 let packet = rtp::packet::Packet { in test_session_srtp_listen() localVariable
136 sa.write_rtp(&packet).await?; in test_session_srtp_listen()
169 let packet = rtp::packet::Packet { in test_session_srtp_multi_ssrc() localVariable
176 sa.write_rtp(&packet).await?; in test_session_srtp_multi_ssrc()
243 let packet = rtp::packet::Packet { in test_session_srtp_replay_protection() localVariable
286 for packet in &packets { in test_session_srtp_replay_protection()
287 sa.udp_tx.send(packet).await?; in test_session_srtp_replay_protection()
290 sa.udp_tx.send(packet).await?; in test_session_srtp_replay_protection()
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| H A D | session_rtcp_test.rs | 146 pkt: &(dyn rtcp::packet::Packet + Send + Sync), in encrypt_srtcp()
182 let packet = picture_loss_indication::PictureLossIndication { in test_session_srtcp_replay_protection() localVariable
187 let encrypted = encrypt_srtcp(&mut local_context, &packet)?; in test_session_srtcp_replay_protection()
217 for packet in &packets { in test_session_srtcp_replay_protection()
218 sa.udp_tx.send(packet).await?; in test_session_srtcp_replay_protection()
221 sa.udp_tx.send(packet).await?; in test_session_srtcp_replay_protection()
223 for packet in &packets { in test_session_srtcp_replay_protection()
225 sa.udp_tx.send(packet).await?; in test_session_srtcp_replay_protection()
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| /webrtc/media/src/io/h264_writer/ |
| H A D | h264_writer_test.rs | 80 let packet = rtp::packet::Packet { in test_write_rtp() localVariable
85 h264writer.write_rtp(&packet)?; in test_write_rtp()
113 let packet = rtp::packet::Packet { in test_write_rtp_fu() localVariable
118 h264writer.write_rtp(&packet)?; in test_write_rtp_fu()
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| H A D | mod.rs | 50 fn write_rtp(&mut self, packet: &rtp::packet::Packet) -> Result<()> { in write_rtp()
51 if packet.payload.is_empty() { in write_rtp()
56 self.has_key_frame = is_key_frame(&packet.payload); in write_rtp()
68 let payload = cached_packet.depacketize(&packet.payload)?; in write_rtp()
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| /webrtc/interceptor/src/mock/ |
| H A D | mock_stream.rs | 10 type RTCPPackets = Vec<Box<dyn rtcp::packet::Packet + Send + Sync>>;
20 rtp_out_modified_tx: mpsc::Sender<rtp::packet::Packet>,
22 rtp_in_rx: Mutex<mpsc::Receiver<rtp::packet::Packet>>,
25 rtp_out_modified_rx: Mutex<mpsc::Receiver<rtp::packet::Packet>>,
27 rtp_in_tx: Mutex<Option<mpsc::Sender<rtp::packet::Packet>>>,
153 pkt: &[Box<dyn rtcp::packet::Packet + Send + Sync>], in write_rtcp()
184 pub async fn receive_rtp(&self, pkt: rtp::packet::Packet) { in receive_rtp()
251 pkts: &[Box<dyn rtcp::packet::Packet + Send + Sync>], in write()
268 let marshaled = rtcp::packet::marshal(&pkts)?; in read()
328 s.write_rtp(&rtp::packet::Packet::default()).await?; in test_mock_stream()
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| /webrtc/rtp/src/codecs/h264/ |
| H A D | mod.rs | 210 fn depacketize(&mut self, packet: &Bytes) -> Result<Bytes> { in depacketize()
211 if packet.len() <= 2 { in depacketize()
219 let b0 = packet[0]; in depacketize()
225 payload.put_u32(packet.len() as u32); in depacketize()
229 payload.put(&*packet.clone()); in depacketize()
234 while curr_offset < packet.len() { in depacketize()
236 ((packet[curr_offset] as usize) << 8) | packet[curr_offset + 1] as usize; in depacketize()
239 if packet.len() < curr_offset + nalu_size { in depacketize()
242 packet.len() - curr_offset, in depacketize()
258 if packet.len() < FUA_HEADER_SIZE { in depacketize()
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| /webrtc/media/src/io/ogg_writer/ |
| H A D | ogg_writer_test.rs | 12 let mut valid_packet = rtp::packet::Packet { in test_ogg_writer_add_packet_and_close()
39 rtp::packet::Packet::default(), in test_ogg_writer_add_packet_and_close()
50 for (msg1, _msg2, packet, err) in add_packet_test_case { in test_ogg_writer_add_packet_and_close()
52 let result = writer.write_rtp(&packet); in test_ogg_writer_add_packet_and_close()
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| H A D | mod.rs | 160 fn write_rtp(&mut self, packet: &rtp::packet::Packet) -> Result<()> { in write_rtp()
162 let payload = opus_packet.depacketize(&packet.payload)?; in write_rtp()
166 let increment = packet.header.timestamp - self.previous_timestamp; in write_rtp()
169 self.previous_timestamp = packet.header.timestamp; in write_rtp()
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| /webrtc/media/src/io/ivf_writer/ |
| H A D | ivf_writer_test.rs | 13 let mut valid_packet = rtp::packet::Packet { in test_ivf_writer_add_packet_and_close()
40 let mut mid_part_packet = rtp::packet::Packet { in test_ivf_writer_add_packet_and_close()
67 let mut keyframe_packet = rtp::packet::Packet { in test_ivf_writer_add_packet_and_close()
118 rtp::packet::Packet::default(), in test_ivf_writer_add_packet_and_close()
154 for (msg1, _msg2, packet, err, seen_key_frame, count) in add_packet_test_case { in test_ivf_writer_add_packet_and_close()
161 let result = writer.write_rtp(&packet); in test_ivf_writer_add_packet_and_close()
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| H A D | mod.rs | 58 fn write_rtp(&mut self, packet: &rtp::packet::Packet) -> Result<()> { in write_rtp()
65 let payload = depacketizer.depacketize(&packet.payload)?; in write_rtp()
70 || (self.current_frame.is_none() && !depacketizer.is_partition_head(&packet.payload)) in write_rtp()
87 if !packet.header.marker { in write_rtp()
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| /webrtc/rtp/src/codecs/opus/ |
| H A D | mod.rs | 33 fn depacketize(&mut self, packet: &Bytes) -> Result<Bytes> { in depacketize()
34 if packet.is_empty() { in depacketize()
37 Ok(packet.clone()) in depacketize()
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| /webrtc/media/src/io/sample_builder/ |
| H A D | mod.rs | 11 use rtp::{packet::Packet, packetizer::Depacketizer};
90 if let Some(ref packet) = self.buffer[i as usize] { in too_old()
91 found_head = Some(packet.header.timestamp); in too_old()
103 if let Some(ref packet) = self.buffer[i as usize] { in too_old()
104 found_tail = Some(packet.header.timestamp); in too_old()
243 while let Some(ref packet) = self.buffer[i as usize] { in build_sample()
247 let is_same_timestamp = head_timestamp.map(|t| packet.header.timestamp == t); in build_sample()
251 .is_partition_tail(packet.header.marker, &packet.payload); in build_sample()
291 if let Some(ref packet) = self.buffer[i as usize] { in build_sample()
292 after_timestamp = packet.header.timestamp; in build_sample()
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| /webrtc/rtcp/src/compound_packet/ |
| H A D | mod.rs | 5 error::Error, header::*, packet::*, receiver_report::*, sender_report::*,
54 for packet in &self.0 { in raw_size()
55 l += packet.marshal_size(); in raw_size()
89 for packet in &self.0 { in marshal_to()
90 let n = packet.marshal_to(buf)?; in marshal_to()
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| /webrtc/interceptor/src/ |
| H A D | lib.rs | 82 async fn write(&self, pkt: &rtp::packet::Packet, attributes: &Attributes) -> Result<usize>; in write()
87 &rtp::packet::Packet,
98 async fn write(&self, pkt: &rtp::packet::Packet, attributes: &Attributes) -> Result<usize> { in write()
134 pkts: &[Box<dyn rtcp::packet::Packet + Send + Sync>], in write()
141 &[Box<dyn rtcp::packet::Packet + Send + Sync>],
155 pkts: &[Box<dyn rtcp::packet::Packet + Send + Sync>], in write()
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| /webrtc/turn/src/proto/ |
| H A D | proto_test.rs | 27 for packet in data { in test_chrome_alloc_request()
29 m.write(&packet)?; in test_chrome_alloc_request()
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| /webrtc/rtp/src/codecs/vp8/ |
| H A D | mod.rs | 151 fn depacketize(&mut self, packet: &Bytes) -> Result<Bytes> { in depacketize()
152 let payload_len = packet.len(); in depacketize()
171 let reader = &mut packet.clone(); in depacketize()
229 if payload_index >= packet.len() { in depacketize()
233 Ok(packet.slice(payload_index..)) in depacketize()
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| /webrtc/interceptor/src/report/receiver/ |
| H A D | receiver_test.rs | 233 .receive_rtp(rtp::packet::Packet { in test_receiver_interceptor_overflow()
243 .receive_rtp(rtp::packet::Packet { in test_receiver_interceptor_overflow()
307 .receive_rtp(rtp::packet::Packet { in test_receiver_interceptor_overflow_five_pkts()
317 .receive_rtp(rtp::packet::Packet { in test_receiver_interceptor_overflow_five_pkts()
327 .receive_rtp(rtp::packet::Packet { in test_receiver_interceptor_overflow_five_pkts()
337 .receive_rtp(rtp::packet::Packet { in test_receiver_interceptor_overflow_five_pkts()
347 .receive_rtp(rtp::packet::Packet { in test_receiver_interceptor_overflow_five_pkts()
409 .receive_rtp(rtp::packet::Packet { in test_receiver_interceptor_packet_loss()
419 .receive_rtp(rtp::packet::Packet { in test_receiver_interceptor_packet_loss()
520 .receive_rtp(rtp::packet::Packet { in test_receiver_interceptor_overflow_and_packet_loss()
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| /webrtc/sctp/fuzz/ |
| H A D | Cargo.toml | 23 name = "packet"
24 path = "fuzz_targets/packet.rs"
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| /webrtc/rtp/src/packet/ |
| H A D | packet_test.rs | 42 let packet = Packet::unmarshal(buf)?; in test_basic() localVariable
44 packet, parsed_packet, in test_basic()
48 packet.header.marshal_size(), in test_basic()
53 packet.marshal_size(), in test_basic()
58 let raw = packet.marshal()?; in test_basic()
100 let packet = Packet { in test_extension() localVariable
134 let packet = Packet::unmarshal(buf)?; in test_padding() localVariable
137 let raw = packet.marshal()?; in test_padding()
296 let ext1 = packet in test_rfc_8285_one_byte_multiple_extensions_with_padding()
304 let ext2 = packet in test_rfc_8285_one_byte_multiple_extensions_with_padding()
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| /webrtc/util/benches/ |
| H A D | bench.rs | 8 let mut packet: Vec<u8> = vec![0; 4]; in buffer_write_then_read() localVariable
11 buffer.read(&mut packet, None).await.unwrap(); in buffer_write_then_read()
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| /webrtc/interceptor/src/report/sender/ |
| H A D | sender_test.rs | 85 .write_rtp(&rtp::packet::Packet { in test_sender_interceptor_after_rtp_packets()
148 .write_rtp(&rtp::packet::Packet { in test_sender_interceptor_after_rtp_packets_overflow()
158 .write_rtp(&rtp::packet::Packet { in test_sender_interceptor_after_rtp_packets_overflow()
168 .write_rtp(&rtp::packet::Packet { in test_sender_interceptor_after_rtp_packets_overflow()
178 .write_rtp(&rtp::packet::Packet { in test_sender_interceptor_after_rtp_packets_overflow()
188 .write_rtp(&rtp::packet::Packet { in test_sender_interceptor_after_rtp_packets_overflow()
|