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rtp: Add AC-3 RTP payloader/depayloader

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
This commit is contained in:
Tim-Philipp Müller 2023-12-19 13:16:29 +00:00 committed by GStreamer Marge Bot
parent 802ff6a67c
commit 6b628485c5
9 changed files with 1437 additions and 0 deletions
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100
docs/plugins/gst_plugins_cache.json
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@ -7295,6 +7295,86 @@
"rsrtp": {
"description": "GStreamer Rust RTP Plugin",
"elements": {
"rtpac3depay2": {
"author": "Tim-Philipp Müller <tim centricular com>",
"description": "Depayload an AC-3 Audio Stream from RTP packets (RFC 4184)",
"hierarchy": [
"GstRtpAc3Depay",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: audio\n encoding-name: AC3\n clock-rate: { (int)48000, (int)44100, (int)32000 }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "audio/x-ac3:\n channels: [ 1, 6 ]\n rate: { (int)48000, (int)44100, (int)32000 }\n framed: true\n alignment: frame\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
"rtpac3pay2": {
"author": "Tim-Philipp Müller <tim centricular com>",
"description": "Payload an AC-3 Audio Elementary Stream into RTP packets (RFC 4184)",
"hierarchy": [
"GstRtpAc3Pay",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "audio/x-ac3:\n rate: { (int)48000, (int)44100, (int)32000 }\n channels: [ 1, 6 ]\n framed: true\n alignment: frame\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n media: audio\n encoding-name: AC3\n clock-rate: { (int)48000, (int)44100, (int)32000 }\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"aggregate-mode": {
"blurb": "Whether to send out audio frames immediately or aggregate them until a packet is full.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "auto (-1)",
"mutable": "null",
"readable": true,
"type": "GstRtpAc3PayAggregateMode",
"writable": true
},
"max-ptime": {
"blurb": "Maximum duration of the packet data in ns (-1 = unlimited up to MTU)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "9223372036854775807",
"min": "-1",
"mutable": "playing",
"readable": true,
"type": "gint64",
"writable": true
}
},
"rank": "marginal"
},
"rtpav1depay": {
"author": "Vivienne Watermeier <vwatermeier@igalia.com>",
"description": "Depayload AV1 from RTP packets",
@ -8410,6 +8490,26 @@
}
}
},
"GstRtpAc3PayAggregateMode": {
"kind": "enum",
"values": [
{
"desc": "Automatic: zero-latency if upstream is live, otherwise aggregate frames until packet is full.",
"name": "auto",
"value": "-1"
},
{
"desc": "Zero Latency: always send out frames right away, do not wait for more frames to fill a packet.",
"name": "zero-latency",
"value": "0"
},
{
"desc": "Aggregate: collect audio frames until we have a full packet or the max-ptime limit is hit (if set).",
"name": "aggregate",
"value": "1"
}
]
},
"GstRtpBaseAudioPay2": {
"hierarchy": [
"GstRtpBaseAudioPay2",

122
net/rtp/src/ac3/ac3_audio_utils.rs Normal file
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@ -0,0 +1,122 @@
// GStreamer RTP AC-3 Audio Utility Functions
//
// Copyright (C) 2023 Tim-Philipp Müller <tim centricular com>
//
// This Source Code Form is subject to the terms of the Mozilla Public License, v2.0.
// If a copy of the MPL was not distributed with this file, You can obtain one at
// <https://mozilla.org/MPL/2.0/>.
//
// SPDX-License-Identifier: MPL-2.0
use gst::prelude::MulDiv;
const AC3_SAMPLES_PER_FRAME: u16 = 1536;
#[derive(Debug, Clone)]
pub(crate) struct FrameHeader {
pub channels: u16,
pub sample_rate: u16,
pub frame_len: usize,
}
impl PartialEq for FrameHeader {
fn eq(&self, other: &Self) -> bool {
self.sample_rate == other.sample_rate && self.channels == other.channels
}
}
impl FrameHeader {
pub(crate) fn duration(&self) -> u64 {
let samples = AC3_SAMPLES_PER_FRAME as u64;
let sample_rate = self.sample_rate as u64;
samples
.mul_div_ceil(*gst::ClockTime::SECOND, sample_rate)
.unwrap()
}
}
const FRAME_LENS_32000: [u16; 38] = [
96u16, 96, 120, 120, 144, 144, 168, 168, 192, 192, 240, 240, 288, 288, 336, 336, 384, 384, 480,
480, 576, 576, 672, 672, 768, 768, 960, 960, 1152, 1152, 1344, 1344, 1536, 1536, 1728, 1728,
1920, 1920,
];
const FRAME_LENS_44100: [u16; 38] = [
69u16, 70, 87, 88, 104, 105, 121, 122, 139, 140, 174, 175, 208, 209, 243, 244, 278, 279, 348,
349, 417, 418, 487, 488, 557, 558, 696, 697, 835, 836, 975, 976, 1114, 1115, 1253, 1254, 1393,
1394,
];
const FRAME_LENS_48000: [u16; 38] = [
64u16, 64, 80, 80, 96, 96, 112, 112, 128, 128, 160, 160, 192, 192, 224, 224, 256, 256, 320,
320, 384, 384, 448, 448, 512, 512, 640, 640, 768, 768, 896, 896, 1024, 1024, 1152, 1152, 1280,
1280,
];
pub(crate) fn peek_frame_header(data: &[u8]) -> Result<FrameHeader, ()> {
// Need sync info and start of bit stream info (bsi)
if data.len() < 5 + 3 {
return Err(());
}
let sync_hdr = u16::from_be_bytes([data[0], data[1]]);
if sync_hdr != 0x0b77 {
return Err(());
}
// skipping 2 bytes of CRC
let (sample_rate, len_table) = {
let fscod = (data[4] >> 6) & 0b11;
match fscod {
0b00 => (48000, &FRAME_LENS_48000),
0b01 => (44100, &FRAME_LENS_44100),
0b10 => (32000, &FRAME_LENS_32000),
_ => return Err(()),
}
};
let frame_len = {
let frmsizcod = data[4] & 0b00111111;
let len_words = len_table.get(frmsizcod as usize).ok_or(())?;
len_words * 2
};
let bsi = &data[5..];
let _bsid = bsi[0] >> 3;
let _bsmod = bsi[0] & 0b00000111;
let channels = {
let bits = u16::from_be_bytes([bsi[1], bsi[2]]);
let acmod = (bits >> 13) & 0b111;
let (nfchans, skip_bits) = match acmod {
0b000 => (2, 0), // 1+1, dual mono
0b001 => (1, 0), // 1/0, center/mono
0b010 => (2, 2), // 2/0, stereo
0b011 => (3, 2), // 3/0, L C R
0b100 => (3, 2), // 2/1, L R S
0b101 => (4, 4), // 3/1, L C R S
0b110 => (4, 2), // 2/2, L R Sl Sr
0b111 => (5, 4), // 3/2, L C R Sl Sr
_ => unreachable!(),
};
let lfe_on = ((bits << (3 + skip_bits)) & 0x8000) >> 15;
nfchans + lfe_on
};
Ok(FrameHeader {
channels,
sample_rate: sample_rate as u16,
frame_len: frame_len as usize,
})
}

406
net/rtp/src/ac3/depay/imp.rs Normal file
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@ -0,0 +1,406 @@
// GStreamer RTP AC-3 Audio Depayloader
//
// Copyright (C) 2023 Tim-Philipp Müller <tim centricular com>
//
// This Source Code Form is subject to the terms of the Mozilla Public License, v2.0.
// If a copy of the MPL was not distributed with this file, You can obtain one at
// <https://mozilla.org/MPL/2.0/>.
//
// SPDX-License-Identifier: MPL-2.0
/**
* SECTION:element-rtpac3depay2
* @see_also: rtpac3pay2, rtpac3depay, rtpac3pay, avdec_ac3, avenc_ac3
*
* Depayload an AC-3 Audio Stream from RTP packets as per [RFC 4184][rfc-4184].
*
* [rfc-4184]: https://www.rfc-editor.org/rfc/rfc4184.html
*
* ## Example pipeline
*
* |[
* gst-launch-1.0 udpsrc caps='application/x-rtp,media=audio,clock-rate=48000,encoding-name=AC3,payload=96' ! rtpjitterbuffer latency=250 ! rtpac3depay2 ! decodebin3 ! audioconvert ! audioresample ! autoaudiosink
* ]| This will depayload an incoming RTP AC-3 audio stream and decode it and play it.
* You can use the `rtpac3pay2` or `rtpac3pay` elements with `avenc_ac3` to create such an RTP stream.
*
* Since: plugins-rs-0.13.0
*/
use atomic_refcell::AtomicRefCell;
use gst::{glib, subclass::prelude::*};
use once_cell::sync::Lazy;
use crate::basedepay::{
Packet, PacketToBufferRelation, RtpBaseDepay2Ext, RtpBaseDepay2Impl, TimestampOffset,
};
use crate::ac3::ac3_audio_utils;
#[derive(Default)]
pub struct RtpAc3Depay {
state: AtomicRefCell<State>,
}
#[derive(Debug, PartialEq)]
enum FragType {
NotFragmented,
Start,
Continuation,
End,
}
#[derive(Debug)]
struct FragmentedFrame {
data: Vec<u8>,
ext_seqnum: u64,
ext_timestamp: u64,
}
#[derive(Default)]
struct State {
last_frame_header: Option<ac3_audio_utils::FrameHeader>,
partial_frame: Option<FragmentedFrame>,
clock_rate: Option<i32>,
}
static CAT: Lazy<gst::DebugCategory> = Lazy::new(|| {
gst::DebugCategory::new(
"rtpac3depay2",
gst::DebugColorFlags::empty(),
Some("RTP AC-3 Audio Depayloader"),
)
});
#[glib::object_subclass]
impl ObjectSubclass for RtpAc3Depay {
const NAME: &'static str = "GstRtpAc3Depay";
type Type = super::RtpAc3Depay;
type ParentType = crate::basedepay::RtpBaseDepay2;
}
impl ObjectImpl for RtpAc3Depay {}
impl GstObjectImpl for RtpAc3Depay {}
impl ElementImpl for RtpAc3Depay {
fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
gst::subclass::ElementMetadata::new(
"RTP AC-3 Audio Depayloader",
"Codec/Depayloader/Network/RTP",
"Depayload an AC-3 Audio Stream from RTP packets (RFC 4184)",
"Tim-Philipp Müller <tim centricular com>",
)
});
Some(&*ELEMENT_METADATA)
}
fn pad_templates() -> &'static [gst::PadTemplate] {
static PAD_TEMPLATES: Lazy<Vec<gst::PadTemplate>> = Lazy::new(|| {
let sink_pad_template = gst::PadTemplate::new(
"sink",
gst::PadDirection::Sink,
gst::PadPresence::Always,
&gst::Caps::builder("application/x-rtp")
.field("media", "audio")
.field("encoding-name", "AC3")
.field("clock-rate", gst::List::new([48000i32, 44100, 32000]))
.build(),
)
.unwrap();
let src_pad_template = gst::PadTemplate::new(
"src",
gst::PadDirection::Src,
gst::PadPresence::Always,
&gst::Caps::builder("audio/x-ac3")
.field("channels", gst::IntRange::new(1, 6))
.field("rate", gst::List::new([48000i32, 44100, 32000]))
.field("framed", true)
.field("alignment", "frame")
.build(),
)
.unwrap();
vec![src_pad_template, sink_pad_template]
});
PAD_TEMPLATES.as_ref()
}
}
impl RtpBaseDepay2Impl for RtpAc3Depay {
const ALLOWED_META_TAGS: &'static [&'static str] = &["audio"];
fn set_sink_caps(&self, caps: &gst::Caps) -> bool {
let s = caps.structure(0).unwrap();
let mut state = self.state.borrow_mut();
state.clock_rate = s.get::<i32>("clock-rate").ok();
// We'll set output caps later based on the frame header
true
}
// Encapsulation of AC-3 Audio Streams:
// https://www.rfc-editor.org/rfc/rfc4184.html#section-4
//
// We either get 1-N whole AC-3 audio frames in an RTP packet,
// or a single AC-3 audio frame split over multiple RTP packets.
fn handle_packet(&self, packet: &Packet) -> Result<gst::FlowSuccess, gst::FlowError> {
let mut state = self.state.borrow_mut();
if packet.discont() {
state.partial_frame = None;
}
let payload = packet.payload();
if payload.len() < 2 + 6 {
gst::warning!(
CAT,
imp: self,
"Payload too small: {} bytes, but need at least 8 bytes",
payload.len(),
);
state.partial_frame = None;
self.obj().drop_packet(packet);
return Ok(gst::FlowSuccess::Ok);
}
gst::log!(
CAT,
imp: self,
"Have payload of {} bytes, header {:02x?} {:02x?}",
payload.len(),
payload[0],
payload[1],
);
// AC-3 specific header
let frag_type = match payload[0] & 0x03 {
0 => FragType::NotFragmented,
1 | 2 => FragType::Start,
3 => {
if packet.marker_bit() {
FragType::End
} else {
FragType::Continuation
}
}
_ => unreachable!(),
};
let num_frames_or_frags = payload[1] as usize;
// Clear out unfinished pending partial frame if needed
if frag_type == FragType::Start || frag_type == FragType::NotFragmented {
if let Some(partial_frame) = state.partial_frame.as_ref() {
gst::warning!(CAT, imp: self, "Dropping unfinished partial frame");
self.obj()
.drop_packets(partial_frame.ext_seqnum..=packet.ext_seqnum() - 1);
state.partial_frame = None;
}
}
// Skip to AC-3 payload data
let payload = &payload[2..];
match frag_type {
FragType::Start => {
let mut data = Vec::with_capacity(num_frames_or_frags * payload.len());
data.extend_from_slice(payload);
state.partial_frame = Some(FragmentedFrame {
data,
ext_seqnum: packet.ext_seqnum(),
ext_timestamp: packet.ext_timestamp(),
});
gst::trace!(CAT, imp: self, "Partial frame {:?}", state.partial_frame);
return Ok(gst::FlowSuccess::Ok);
}
FragType::Continuation | FragType::End => {
let Some(partial_frame) = state.partial_frame.as_mut() else {
gst::debug!(
CAT,
imp: self,
"{frag_type:?} packet but no partial frame (most likely indicates packet loss)",
);
self.obj().drop_packet(packet);
state.partial_frame = None;
return Ok(gst::FlowSuccess::Ok);
};
if partial_frame.ext_timestamp != packet.ext_timestamp() {
gst::warning!(
CAT,
imp: self,
"{frag_type:?} packet timestamp {} doesn't match existing partial fragment timestamp {}",
packet.ext_timestamp(),
partial_frame.ext_timestamp,
);
state.partial_frame = None;
self.obj().drop_packet(packet);
return Ok(gst::FlowSuccess::Ok);
}
partial_frame.data.extend_from_slice(payload);
gst::log!(
CAT,
imp: self,
"Added {frag_type:?} packet payload, assembled {} bytes now",
partial_frame.data.len()
);
if frag_type == FragType::End {
let partial_frame = state.partial_frame.take().unwrap();
let Ok(hdr) = ac3_audio_utils::peek_frame_header(&partial_frame.data) else {
gst::warning!(CAT, imp: self, "Could not parse frame header, dropping frame");
self.obj()
.drop_packets(partial_frame.ext_seqnum..=packet.ext_seqnum());
return Ok(gst::FlowSuccess::Ok);
};
gst::trace!(CAT, imp: self, "{hdr:?}");
if partial_frame.data.len() != hdr.frame_len {
gst::warning!(
CAT,
imp: self,
"Partial frame finished, but have {} bytes, and expected {} bytes!",
partial_frame.data.len(),
hdr.frame_len,
);
}
self.ensure_output_caps(&mut state, &hdr);
let mut outbuf = gst::Buffer::from_mut_slice(partial_frame.data);
let outbuf_ref = outbuf.get_mut().unwrap();
outbuf_ref.set_duration(gst::ClockTime::from_nseconds(hdr.duration()));
gst::trace!(CAT, imp: self, "Finishing buffer {outbuf:?}");
return self.obj().queue_buffer(
PacketToBufferRelation::Seqnums(
partial_frame.ext_seqnum..=packet.ext_seqnum(),
),
outbuf,
);
}
// Wait for more frame fragments
return Ok(gst::FlowSuccess::Ok);
}
FragType::NotFragmented => {
let mut offset = 0;
let mut ts_offset = 0;
while offset < payload.len() {
let Ok(hdr) = ac3_audio_utils::peek_frame_header(&payload[offset..]) else {
gst::warning!(CAT, imp: self, "Could not parse frame header at offset {offset}");
break;
};
gst::trace!(CAT, imp: self, "{hdr:?} at offset {offset}");
let frame_len = if offset + hdr.frame_len <= payload.len() {
hdr.frame_len
} else {
gst::warning!(
CAT,
imp: self,
"Frame at offset {offset} is {} bytes, but we have only {} bytes left!",
hdr.frame_len,
payload.len() - offset,
);
// We'll still push out what we have, there might be decodable blocks
payload.len() - offset
};
self.ensure_output_caps(&mut state, &hdr);
gst::trace!(CAT, imp: self, "Getting frame @ {offset}+{frame_len}");
let mut outbuf =
packet.payload_subbuffer_from_offset_with_length(offset, frame_len);
let outbuf_ref = outbuf.get_mut().unwrap();
outbuf_ref.set_duration(gst::ClockTime::from_nseconds(hdr.duration()));
gst::trace!(CAT, imp: self, "Finishing frame @ {offset}, buffer {outbuf:?}");
self.obj().queue_buffer(
PacketToBufferRelation::SeqnumsWithOffset {
seqnums: packet.ext_seqnum()..=packet.ext_seqnum(),
timestamp_offset: TimestampOffset::Pts(
gst::Signed::<gst::ClockTime>::from(ts_offset as i64),
),
},
outbuf,
)?;
offset += frame_len;
ts_offset += hdr.duration();
}
}
}
Ok(gst::FlowSuccess::Ok)
}
fn stop(&self) -> Result<(), gst::ErrorMessage> {
*self.state.borrow_mut() = State::default();
Ok(())
}
}
impl RtpAc3Depay {
fn ensure_output_caps(&self, state: &mut State, frame_header: &ac3_audio_utils::FrameHeader) {
let update_caps = state.last_frame_header.as_ref() != Some(frame_header);
if update_caps {
if state.clock_rate != Some(frame_header.sample_rate as i32) {
gst::warning!(
CAT,
imp: self,
"clock-rate {} does not match sample rate {}!",
state.clock_rate.unwrap(),
frame_header.sample_rate,
);
}
let src_caps = gst::Caps::builder("audio/x-ac3")
.field("rate", frame_header.sample_rate as i32)
.field("channels", frame_header.channels as i32)
.field("framed", true)
.field("alignment", "frame")
.build();
gst::info!(CAT, imp: self, "Setting output caps {src_caps}..");
// Ignore failure here and let the next buffer push yield an appropriate flow return
self.obj().set_src_caps(&src_caps);
state.last_frame_header = Some(frame_header.clone());
}
}
}

28
net/rtp/src/ac3/depay/mod.rs Normal file
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@ -0,0 +1,28 @@
// GStreamer RTP AC-3 Audio Depayloader
//
// Copyright (C) 2023 Tim-Philipp Müller <tim centricular com>
//
// This Source Code Form is subject to the terms of the Mozilla Public License, v2.0.
// If a copy of the MPL was not distributed with this file, You can obtain one at
// <https://mozilla.org/MPL/2.0/>.
//
// SPDX-License-Identifier: MPL-2.0
use gst::glib;
use gst::prelude::*;
pub mod imp;
glib::wrapper! {
pub struct RtpAc3Depay(ObjectSubclass<imp::RtpAc3Depay>)
@extends crate::basedepay::RtpBaseDepay2, gst::Element, gst::Object;
}
pub fn register(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
gst::Element::register(
Some(plugin),
"rtpac3depay2",
gst::Rank::MARGINAL,
RtpAc3Depay::static_type(),
)
}

6
net/rtp/src/ac3/mod.rs Normal file
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@ -0,0 +1,6 @@
// SPDX-License-Identifier: MPL-2.0
pub mod depay;
pub mod pay;
mod ac3_audio_utils;

711
net/rtp/src/ac3/pay/imp.rs Normal file
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@ -0,0 +1,711 @@
// GStreamer RTP AC-3 Audio Payloader
//
// Copyright (C) 2023 Tim-Philipp Müller <tim centricular com>
//
// This Source Code Form is subject to the terms of the Mozilla Public License, v2.0.
// If a copy of the MPL was not distributed with this file, You can obtain one at
// <https://mozilla.org/MPL/2.0/>.
//
// SPDX-License-Identifier: MPL-2.0
/**
* SECTION:element-rtpac3pay2
* @see_also: rtpac3depay2, rtpac3pay, rtpac3depay, avdec_ac3, a52dec, avenc_ac3
*
* Payload an AC-3 Audio Elementary Stream into RTP packets as per [RFC 4184][rfc-4184].
* Also see the [IANA media-type page for AC-3][iana-ac3].
*
* [rfc-4184]: https://www.rfc-editor.org/rfc/rfc4184.html#section-3.5
* [iana-ac3]: https://www.iana.org/assignments/media-types/audio/ac3
*
* ## Aggregation Modes
*
* The default aggregation mode is `auto`: If upstream is live, the payloader will send out all
* audio frames immediately, even if they don't completely fill a packet, in order to minimise
* latency. If upstream is not live, the payloader will by default aggregate audio frames until
* it has completely filled an RTP packet as per the configured MTU size or the `max-ptime`
* property if it is set (it is not set by default).
*
* The aggregation mode can be controlled via the `aggregate-mode` property.
*
* ## Example pipeline
*
* |[
* gst-launch audiotestsrc wave=ticks ! avenc_ac3 ! ac3parse ! rtpac3pay2 ! udpsink host=127.0.0.1 port=5004
* ]| This will encode an audio test signal to AC-3 and then payload the encoded audio
* into RTP packets and send them out via UDP to localhost (IPv4) port 5004.
* You can use the `rtpac3depay2` or `rtpac3depay` elements to depayload such a stream, and
* the `avdec_ac3` or `a52dec` elements to decode the depayloaded stream.
*
* Since: plugins-rs-0.13.0
*/
use atomic_refcell::AtomicRefCell;
use std::collections::VecDeque;
use std::sync::{Arc, Mutex};
use gst::{glib, prelude::*, subclass::prelude::*};
use once_cell::sync::Lazy;
use crate::basepay::{
PacketToBufferRelation, RtpBasePay2Ext, RtpBasePay2Impl, RtpBasePay2ImplExt, TimestampOffset,
};
use crate::ac3::ac3_audio_utils;
use super::RtpAc3PayAggregateMode;
// https://www.rfc-editor.org/rfc/rfc4184.html#section-4.1
const AC3_SPECIFIC_HEADER_LEN: usize = 2;
#[derive(Clone)]
struct Settings {
max_ptime: Option<gst::ClockTime>,
aggregate_mode: RtpAc3PayAggregateMode,
}
impl Default for Settings {
fn default() -> Self {
Settings {
aggregate_mode: RtpAc3PayAggregateMode::Auto,
max_ptime: None,
}
}
}
#[derive(Default)]
pub struct RtpAc3Pay {
state: AtomicRefCell<State>,
settings: Mutex<Settings>,
is_live: Mutex<Option<bool>>,
}
#[derive(Debug)]
struct QueuedFrame {
// Id of the input buffer this frame came from
id: u64,
// Time offset to the timestamp of the buffer this input buffer came from
// (will be non-zero if the input buffer contained multiple audio frames)
pts_offset: u64,
// Mapped buffer data and offset into the buffer data
buffer: Arc<gst::MappedBuffer<gst::buffer::Readable>>,
offset: usize,
// Audio frame header
header: ac3_audio_utils::FrameHeader,
}
impl QueuedFrame {
fn duration(&self) -> u64 {
self.header.duration()
}
fn len(&self) -> usize {
self.header.frame_len
}
fn data(&self) -> &[u8] {
let end_offset = self.offset + self.len();
&self.buffer[self.offset..end_offset]
}
}
#[derive(Default)]
struct State {
// Queued audio frames (we collect until min-ptime/max-ptime is hit or the packet is full)
queued_frames: VecDeque<QueuedFrame>,
// Desired "packet time", i.e. packet duration, from the downstream caps, if set
ptime: Option<gst::ClockTime>,
max_ptime: Option<gst::ClockTime>,
}
static CAT: Lazy<gst::DebugCategory> = Lazy::new(|| {
gst::DebugCategory::new(
"rtpac3pay2",
gst::DebugColorFlags::empty(),
Some("RTP AC-3 Audio Payloader"),
)
});
#[glib::object_subclass]
impl ObjectSubclass for RtpAc3Pay {
const NAME: &'static str = "GstRtpAc3Pay";
type Type = super::RtpAc3Pay;
type ParentType = crate::basepay::RtpBasePay2;
}
impl ObjectImpl for RtpAc3Pay {
fn properties() -> &'static [glib::ParamSpec] {
static PROPERTIES: Lazy<Vec<glib::ParamSpec>> = Lazy::new(|| {
vec![
glib::ParamSpecEnum::builder_with_default("aggregate-mode", Settings::default().aggregate_mode)
.nick("Aggregate Mode")
.blurb("Whether to send out audio frames immediately or aggregate them until a packet is full.")
.build(),
// Using same type/semantics as C payloaders
glib::ParamSpecInt64::builder("max-ptime")
.nick("Maximum Packet Time")
.blurb("Maximum duration of the packet data in ns (-1 = unlimited up to MTU)")
.default_value(
Settings::default()
.max_ptime
.map(gst::ClockTime::nseconds)
.map(|x| x as i64)
.unwrap_or(-1),
)
.minimum(-1)
.maximum(i64::MAX)
.mutable_playing()
.build(),
]
});
PROPERTIES.as_ref()
}
fn set_property(&self, _id: usize, value: &glib::Value, pspec: &glib::ParamSpec) {
let mut settings = self.settings.lock().unwrap();
match pspec.name() {
"aggregate-mode" => {
settings.aggregate_mode = value
.get::<RtpAc3PayAggregateMode>()
.expect("type checked upstream");
}
"max-ptime" => {
let new_max_ptime = match value.get::<i64>().unwrap() {
-1 => None,
v @ 0.. => Some(gst::ClockTime::from_nseconds(v as u64)),
_ => unreachable!(),
};
let changed = settings.max_ptime != new_max_ptime;
settings.max_ptime = new_max_ptime;
drop(settings);
if changed {
let _ = self
.obj()
.post_message(gst::message::Latency::builder().src(&*self.obj()).build());
}
}
_ => unimplemented!(),
};
}
fn property(&self, _id: usize, pspec: &glib::ParamSpec) -> glib::Value {
let settings = self.settings.lock().unwrap();
match pspec.name() {
"aggregate-mode" => settings.aggregate_mode.to_value(),
"max-ptime" => (settings
.max_ptime
.map(gst::ClockTime::nseconds)
.map(|x| x as i64)
.unwrap_or(-1))
.to_value(),
_ => unimplemented!(),
}
}
}
impl GstObjectImpl for RtpAc3Pay {}
impl ElementImpl for RtpAc3Pay {
fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
gst::subclass::ElementMetadata::new(
"RTP AC-3 Audio Payloader",
"Codec/Payloader/Network/RTP",
"Payload an AC-3 Audio Elementary Stream into RTP packets (RFC 4184)",
"Tim-Philipp Müller <tim centricular com>",
)
});
Some(&*ELEMENT_METADATA)
}
fn pad_templates() -> &'static [gst::PadTemplate] {
static PAD_TEMPLATES: Lazy<Vec<gst::PadTemplate>> = Lazy::new(|| {
let sink_pad_template = gst::PadTemplate::new(
"sink",
gst::PadDirection::Sink,
gst::PadPresence::Always,
&gst::Caps::builder("audio/x-ac3")
.field("rate", gst::List::new([48000i32, 44100, 32000]))
.field("channels", gst::IntRange::new(1, 6))
.field("framed", true)
.field("alignment", "frame")
.build(),
)
.unwrap();
let src_pad_template = gst::PadTemplate::new(
"src",
gst::PadDirection::Src,
gst::PadPresence::Always,
&gst::Caps::builder("application/x-rtp")
.field("media", "audio")
.field("encoding-name", "AC3")
.field("clock-rate", gst::List::new([48000i32, 44100, 32000]))
.build(),
)
.unwrap();
vec![sink_pad_template, src_pad_template]
});
PAD_TEMPLATES.as_ref()
}
}
impl RtpBasePay2Impl for RtpAc3Pay {
const ALLOWED_META_TAGS: &'static [&'static str] = &["audio"];
fn set_sink_caps(&self, caps: &gst::Caps) -> bool {
let s = caps.structure(0).unwrap();
let channels = s.get::<i32>("channels").unwrap();
let rate = s.get::<i32>("rate").unwrap();
let src_caps = gst::Caps::builder("application/x-rtp")
.field("media", "audio")
.field("encoding-name", "AC3")
.field("clock-rate", rate)
.field("channels", channels.to_string());
self.obj().set_src_caps(&src_caps.build());
true
}
fn negotiate(&self, mut src_caps: gst::Caps) {
// Fixate as a first step
src_caps.fixate();
let s = src_caps.structure(0).unwrap();
// Negotiate ptime/maxptime with downstream and use them in combination with the
// properties. See https://www.iana.org/assignments/media-types/audio/ac3
let ptime = s
.get::<u32>("ptime")
.ok()
.map(u64::from)
.map(gst::ClockTime::from_mseconds);
let max_ptime = s
.get::<u32>("maxptime")
.ok()
.map(u64::from)
.map(gst::ClockTime::from_mseconds);
self.parent_negotiate(src_caps);
let mut state = self.state.borrow_mut();
state.ptime = ptime;
state.max_ptime = max_ptime;
drop(state);
}
// Encapsulation of AC-3 Audio Elementary Streams:
// https://www.rfc-editor.org/rfc/rfc4184.html#section-4.2
//
// We either put 1-N whole AC-3 audio frames into a single RTP packet,
// or split a single AC-3 audio frame over multiple RTP packets.
//
fn handle_buffer(
&self,
buffer: &gst::Buffer,
id: u64,
) -> Result<gst::FlowSuccess, gst::FlowError> {
let mut state = self.state.borrow_mut();
let mut settings = self.settings.lock().unwrap();
if buffer.flags().contains(gst::BufferFlags::DISCONT) {
gst::debug!(CAT, imp: self, "Discont on {buffer:?}, pushing out any pending frames");
self.send_packets(&settings, &mut state, SendPacketMode::ForcePending)?;
}
let map = buffer.clone().into_mapped_buffer_readable().map_err(|_| {
gst::error!(CAT, imp: self, "Can't map buffer readable");
gst::FlowError::Error
})?;
// Arc so we can share the MappedBuffer amongst multiple frames.
// Todo: could probably do something more clever to avoid the heap
// allocation in the case where the input buffer contains a single
// audio frame only (which is probably the normal case).
let map = Arc::new(map);
let data = map.as_slice();
let mut pts_offset = 0;
let mut map_offset = 0;
loop {
let Ok(frame_hdr) = ac3_audio_utils::peek_frame_header(&data[map_offset..]) else {
gst::warning!(
CAT,
imp: self,
"Failed to parse AC-3 audio frame header for {buffer:?} at offset {map_offset}",
);
if map_offset > 0 {
break;
}
self.send_packets(&settings, &mut state, SendPacketMode::ForcePending)?;
self.obj().drop_buffers(..=id);
return Ok(gst::FlowSuccess::Ok);
};
let queued_frame = QueuedFrame {
id,
pts_offset,
buffer: map.clone(),
offset: map_offset,
header: frame_hdr,
};
let frame_len = queued_frame.len();
let frame_dur = queued_frame.duration();
if map_offset + frame_len > data.len() {
gst::warning!(CAT, imp: self, "Short audio frame for {buffer:?} at offset {map_offset}");
}
pts_offset += frame_dur;
map_offset += frame_len;
state.queued_frames.push_back(queued_frame);
if map_offset >= data.len() {
break;
}
}
// Make sure we have queried upstream liveness if needed
if settings.aggregate_mode == RtpAc3PayAggregateMode::Auto {
self.ensure_upstream_liveness(&mut settings);
}
self.send_packets(&settings, &mut state, SendPacketMode::WhenReady)
}
fn drain(&self) -> Result<gst::FlowSuccess, gst::FlowError> {
let settings = self.settings.lock().unwrap().clone();
let mut state = self.state.borrow_mut();
self.send_packets(&settings, &mut state, SendPacketMode::ForcePending)
}
fn flush(&self) {
let mut state = self.state.borrow_mut();
state.queued_frames.clear();
}
#[allow(clippy::single_match)]
fn src_query(&self, query: &mut gst::QueryRef) -> bool {
let res = self.parent_src_query(query);
if !res {
return false;
}
match query.view_mut() {
gst::QueryViewMut::Latency(query) => {
let settings = self.settings.lock().unwrap();
let (is_live, mut min, mut max) = query.result();
{
let mut live_guard = self.is_live.lock().unwrap();
if Some(is_live) != *live_guard {
gst::info!(CAT, imp: self, "Upstream is live: {is_live}");
*live_guard = Some(is_live);
}
}
if self.effective_aggregate_mode(&settings) == RtpAc3PayAggregateMode::Aggregate {
if let Some(max_ptime) = settings.max_ptime {
min += max_ptime;
max.opt_add_assign(max_ptime);
} else if is_live {
gst::warning!(
CAT,
imp: self,
"Aggregating packets in live mode, but no max_ptime configured. \
Configured latency may be too low!",
);
}
query.set(is_live, min, max);
}
}
_ => (),
}
true
}
fn start(&self) -> Result<(), gst::ErrorMessage> {
*self.state.borrow_mut() = State::default();
*self.is_live.lock().unwrap() = None;
self.parent_start()
}
fn stop(&self) -> Result<(), gst::ErrorMessage> {
*self.state.borrow_mut() = State::default();
*self.is_live.lock().unwrap() = None;
self.parent_stop()
}
}
#[derive(Debug, PartialEq)]
enum SendPacketMode {
WhenReady,
ForcePending,
}
impl RtpAc3Pay {
fn send_packets(
&self,
settings: &Settings,
state: &mut State,
send_mode: SendPacketMode,
) -> Result<gst::FlowSuccess, gst::FlowError> {
let agg_mode = self.effective_aggregate_mode(settings);
let max_payload_size = self.obj().max_payload_size() as usize - AC3_SPECIFIC_HEADER_LEN;
// Send out packets if there's enough data for one (or more), or if forced.
while let Some(first) = state.queued_frames.front() {
// Big audio frame that needs to be split across multiple packets?
if first.header.frame_len > max_payload_size {
let first = state.queued_frames.pop_front().unwrap();
let mut data = first.buffer.as_slice();
let mut frag_offset = 0;
let id = first.id;
while frag_offset < first.header.frame_len {
let left = first.header.frame_len - frag_offset;
let is_last = left <= max_payload_size;
let bytes_in_this_packet = std::cmp::min(left, max_payload_size);
// 0 1
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | MBZ | FT| NF |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
// https://www.rfc-editor.org/rfc/rfc4184.html#section-4.1.1
//
let frame_type: u16 = {
if frag_offset == 0 {
if bytes_in_this_packet >= (first.header.frame_len * 5 / 8) {
// Initial fragment of frame which includes the first 5/8ths of the frame
1
} else {
// Initial fragment of frame, which does not include the first 5/8ths of the frame
2
}
} else {
// Fragment of frame other than initial fragment
3
}
};
// The number fragments (and therefore packets) that make up the current frame
let n = (first.header.frame_len + max_payload_size - 1) / max_payload_size;
let ac3_specific_header = ((frame_type << 8) | (n as u16)).to_be_bytes();
// https://www.rfc-editor.org/rfc/rfc4184.html#section-3
self.obj().queue_packet(
id.into(),
rtp_types::RtpPacketBuilder::new()
.payload(ac3_specific_header.as_slice())
.payload(&data[0..bytes_in_this_packet])
.marker_bit(is_last),
)?;
data = &data[bytes_in_this_packet..];
frag_offset += bytes_in_this_packet;
}
continue;
}
let n_frames = state.queued_frames.len();
let queue_size = state.queued_frames.iter().map(|f| f.len()).sum::<usize>();
let queue_duration = state
.queued_frames
.iter()
.map(|f| f.duration())
.sum::<u64>();
// We optimistically add average size/duration to send out packets as early as possible
// if we estimate that the next frame would likely overflow our accumulation limits.
let avg_size = queue_size / n_frames;
let avg_duration = queue_duration / n_frames as u64;
let max_ptime = self.calc_effective_max_ptime(settings, state);
let is_ready = send_mode == SendPacketMode::ForcePending
|| agg_mode != RtpAc3PayAggregateMode::Aggregate
|| queue_size + avg_size > max_payload_size
|| (max_ptime.is_some() && queue_duration + avg_duration > max_ptime.unwrap());
gst::log!(
CAT,
imp: self,
"Queued: size {queue_size}, duration ~{}ms, mode: {:?} + {:?} => ready: {}",
queue_duration / 1_000_000,
agg_mode,
send_mode,
is_ready);
if !is_ready {
gst::log!(CAT, imp: self, "Not ready yet, waiting for more data");
break;
}
gst::trace!(CAT, imp: self, "Creating packet..");
let pts_offset = gst::ClockTime::from_nseconds(first.pts_offset);
let id = first.id;
let mut end_id = first.id;
let mut acc_duration = 0;
let mut acc_size = 0;
let mut n = 0; // Number of frames in packet
// Figure out how many frames we're going to put into the packet, needed for the header
for frame in &state.queued_frames {
gst::trace!(
CAT,
imp: self,
"{frame:?}, accumulated size {acc_size} duration ~{}ms",
acc_duration / 1_000_000);
// If this frame would overflow the packet, bail out and send out what we have.
//
// Don't take into account the max_ptime for the first frame, since it could be
// lower than the frame duration in which case we would never payload anything.
//
// For the size check in bytes we know that the first frame will fit the mtu,
// because we already checked for the "audio frame bigger than mtu" scenario above.
if acc_size + frame.len() > max_payload_size
|| (max_ptime.is_some()
&& acc_duration > 0
&& acc_duration + frame.duration() > max_ptime.unwrap())
{
break;
}
// ... otherwise add frame to the packet (we'll do the actual payloading later)
n += 1;
acc_size += frame.len();
acc_duration += frame.duration();
end_id = frame.id;
// .. and check if there are more frames we can add to the packet
}
// Packet creation and payloading
let frame_type: u16 = 0; // One or more complete frames
let ac3_specific_header = ((frame_type << 8) | (n as u16)).to_be_bytes();
let mut packet = rtp_types::RtpPacketBuilder::new()
.marker_bit(true)
.payload(ac3_specific_header.as_slice());
// Add frames to the packet
for frame in state.queued_frames.iter().take(n) {
packet = packet.payload(frame.data());
}
self.obj().queue_packet(
PacketToBufferRelation::IdsWithOffset {
ids: (id..=end_id),
timestamp_offset: TimestampOffset::Pts(pts_offset),
},
packet,
)?;
// Now pop off all the frames we used (now that the packet has been written out)
for _ in 0..n {
let _ = state.queued_frames.pop_front();
}
}
gst::log!(CAT, imp: self, "All done for now, {} frames queued", state.queued_frames.len());
if send_mode == SendPacketMode::ForcePending {
self.obj().finish_pending_packets()?;
}
Ok(gst::FlowSuccess::Ok)
}
fn effective_aggregate_mode(&self, settings: &Settings) -> RtpAc3PayAggregateMode {
match settings.aggregate_mode {
RtpAc3PayAggregateMode::Auto => match self.is_live() {
Some(true) => RtpAc3PayAggregateMode::ZeroLatency,
Some(false) => RtpAc3PayAggregateMode::Aggregate,
None => RtpAc3PayAggregateMode::ZeroLatency,
},
mode => mode,
}
}
fn is_live(&self) -> Option<bool> {
*self.is_live.lock().unwrap()
}
// Query upstream live-ness if needed, in case of aggregate-mode=auto
fn ensure_upstream_liveness(&self, settings: &mut Settings) {
if settings.aggregate_mode != RtpAc3PayAggregateMode::Auto || self.is_live().is_some() {
return;
}
let mut q = gst::query::Latency::new();
let is_live = if self.obj().sink_pad().peer_query(&mut q) {
let (is_live, _, _) = q.result();
is_live
} else {
false
};
*self.is_live.lock().unwrap() = Some(is_live);
gst::info!(CAT, imp: self, "Upstream is live: {is_live}");
}
// We can get max ptime or ptime recommendations/restrictions from multiple places, e.g. the
// "max-ptime" property, but also from "maxptime" or "ptime" values from downstream / an SDP.
//
// Here we look at the various values and decide on an effective max ptime value.
//
// We'll just return the lowest of any set values.
//
fn calc_effective_max_ptime(&self, settings: &Settings, state: &State) -> Option<u64> {
[settings.max_ptime, state.max_ptime, state.ptime]
.into_iter()
.filter(|v| v.is_some())
.min()
.map(|t| t.unwrap().nseconds())
}
}

57
net/rtp/src/ac3/pay/mod.rs Normal file
View file

@ -0,0 +1,57 @@
// GStreamer RTP AC-3 Audio Payloader
//
// Copyright (C) 2023 Tim-Philipp Müller <tim centricular com>
//
// This Source Code Form is subject to the terms of the Mozilla Public License, v2.0.
// If a copy of the MPL was not distributed with this file, You can obtain one at
// <https://mozilla.org/MPL/2.0/>.
//
// SPDX-License-Identifier: MPL-2.0
use gst::glib;
use gst::prelude::*;
pub mod imp;
#[derive(Debug, Eq, PartialEq, Ord, PartialOrd, Hash, Clone, Copy, glib::Enum)]
#[repr(i32)]
#[enum_type(name = "GstRtpAc3PayAggregateMode")]
#[non_exhaustive]
pub(crate) enum RtpAc3PayAggregateMode {
#[enum_value(
name = "Automatic: zero-latency if upstream is live, otherwise aggregate frames until packet is full.",
nick = "auto"
)]
Auto = -1,
#[enum_value(
name = "Zero Latency: always send out frames right away, do not wait for more frames to fill a packet.",
nick = "zero-latency"
)]
ZeroLatency = 0,
#[enum_value(
name = "Aggregate: collect audio frames until we have a full packet or the max-ptime limit is hit (if set).",
nick = "aggregate"
)]
Aggregate = 1,
}
glib::wrapper! {
pub struct RtpAc3Pay(ObjectSubclass<imp::RtpAc3Pay>)
@extends crate::basepay::RtpBasePay2, gst::Element, gst::Object;
}
pub fn register(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
#[cfg(feature = "doc")]
{
RtpAc3PayAggregateMode::static_type().mark_as_plugin_api(gst::PluginAPIFlags::empty());
}
gst::Element::register(
Some(plugin),
"rtpac3pay2",
gst::Rank::MARGINAL,
RtpAc3Pay::static_type(),
)
}

4
net/rtp/src/lib.rs
View file

@ -31,6 +31,7 @@ mod baseaudiopay;
mod basedepay;
mod basepay;
mod ac3;
mod av1;
mod jpeg;
mod klv;
@ -61,6 +62,9 @@ fn plugin_init(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
.mark_as_plugin_api(gst::PluginAPIFlags::empty());
}
ac3::depay::register(plugin)?;
ac3::pay::register(plugin)?;
av1::depay::register(plugin)?;
av1::pay::register(plugin)?;

3
typos.toml
View file

@ -9,6 +9,9 @@ seeked = "seeked"
fiel = "fiel"
trun = "trun"
# net/rtp/src/ac3 - "5/8ths" - not sure how to allow this without also letting through all typos of 'this'
ths = "ths"
[files]
extend-exclude = [
"*.mcc",