[feat](trx-server): gzip-compress history replay blob

Add flate2 dependency and a new AUDIO_MSG_HISTORY_COMPRESSED (0x0a)
wire type. The server gzip-compresses the full history blob before
sending; JSON history compresses ~10-20x so both transfer size and
client wait time drop significantly. The client decompresses and
dispatches sub-messages from the embedded framed stream. MAX_PAYLOAD_SIZE
is kept at 1 MB for normal messages; a separate 16 MB limit is applied
only to the compressed history type.

Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
This commit is contained in:
2026-03-09 21:17:49 +01:00
parent 26fbd37b6d
commit 409b173f62
7 changed files with 76 additions and 15 deletions
+1
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@@ -29,6 +29,7 @@ members = [
resolver = "2"
[workspace.dependencies]
flate2 = "1"
tokio = "1"
tokio-serial = "5"
serde = "1"
+2 -1
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@@ -8,6 +8,7 @@ version = "0.1.0"
edition = "2021"
[dependencies]
flate2 = { workspace = true }
tokio = { workspace = true, features = ["full"] }
serde = { workspace = true, features = ["derive"] }
serde_json = { workspace = true }
@@ -27,4 +28,4 @@ trx-frontend-http-json = { path = "trx-frontend/trx-frontend-http-json" }
trx-frontend-rigctl = { path = "trx-frontend/trx-frontend-rigctl" }
[features]
default = []
default = []
+33 -2
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@@ -10,6 +10,8 @@ use std::sync::{Arc, Mutex};
use std::time::Duration;
use bytes::Bytes;
use flate2::read::GzDecoder;
use std::io::Read as _;
use tokio::io::BufReader;
use tokio::net::TcpStream;
use tokio::sync::{broadcast, mpsc, watch};
@@ -19,8 +21,9 @@ use trx_frontend::RemoteRigEntry;
use trx_core::audio::{
read_audio_msg, write_audio_msg, AudioStreamInfo, AUDIO_MSG_AIS_DECODE, AUDIO_MSG_APRS_DECODE,
AUDIO_MSG_CW_DECODE, AUDIO_MSG_FT8_DECODE, AUDIO_MSG_HF_APRS_DECODE, AUDIO_MSG_RX_FRAME,
AUDIO_MSG_STREAM_INFO, AUDIO_MSG_TX_FRAME, AUDIO_MSG_VDES_DECODE, AUDIO_MSG_WSPR_DECODE,
AUDIO_MSG_CW_DECODE, AUDIO_MSG_FT8_DECODE, AUDIO_MSG_HF_APRS_DECODE,
AUDIO_MSG_HISTORY_COMPRESSED, AUDIO_MSG_RX_FRAME, AUDIO_MSG_STREAM_INFO, AUDIO_MSG_TX_FRAME,
AUDIO_MSG_VDES_DECODE, AUDIO_MSG_WSPR_DECODE,
};
use trx_core::decode::DecodedMessage;
@@ -147,6 +150,34 @@ async fn handle_audio_connection(
Ok((AUDIO_MSG_RX_FRAME, payload)) => {
let _ = rx_tx.send(Bytes::from(payload));
}
Ok((AUDIO_MSG_HISTORY_COMPRESSED, payload)) => {
// Decompress gzip blob, then iterate the embedded framed messages.
let mut decompressed = Vec::new();
if GzDecoder::new(payload.as_slice())
.read_to_end(&mut decompressed)
.is_ok()
{
let mut pos = 0;
while pos + 5 <= decompressed.len() {
let _msg_type = decompressed[pos];
let len = u32::from_be_bytes([
decompressed[pos + 1],
decompressed[pos + 2],
decompressed[pos + 3],
decompressed[pos + 4],
]) as usize;
pos += 5;
if pos + len > decompressed.len() {
break;
}
let json = &decompressed[pos..pos + len];
if let Ok(msg) = serde_json::from_slice::<DecodedMessage>(json) {
let _ = decode_tx.send(msg);
}
pos += len;
}
}
}
Ok((
AUDIO_MSG_VDES_DECODE
| AUDIO_MSG_AIS_DECODE
+1
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@@ -12,3 +12,4 @@ tokio = { workspace = true, features = ["full"] }
serde = { workspace = true, features = ["derive"] }
serde_json = { workspace = true }
tracing = { workspace = true }
flate2 = { workspace = true }
+14 -3
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@@ -18,9 +18,15 @@ pub const AUDIO_MSG_WSPR_DECODE: u8 = 0x06;
pub const AUDIO_MSG_AIS_DECODE: u8 = 0x07;
pub const AUDIO_MSG_VDES_DECODE: u8 = 0x08;
pub const AUDIO_MSG_HF_APRS_DECODE: u8 = 0x09;
/// Compressed history blob: payload is a gzip-compressed sequence of normal
/// framed messages (each: `[1 byte type][4 bytes BE length][payload]`).
pub const AUDIO_MSG_HISTORY_COMPRESSED: u8 = 0x0a;
/// Maximum payload size (1 MB) to reject bogus frames early.
/// Maximum payload size for normal messages (1 MB).
const MAX_PAYLOAD_SIZE: u32 = 1_048_576;
/// Maximum payload size for the compressed history blob (16 MB).
/// A compressed 24-hour history on a busy channel can reach several MB.
const MAX_HISTORY_PAYLOAD_SIZE: u32 = 16_777_216;
#[derive(Debug, Clone, serde::Serialize, serde::Deserialize)]
pub struct AudioStreamInfo {
@@ -59,10 +65,15 @@ pub async fn read_audio_msg<R: AsyncRead + Unpin>(
) -> std::io::Result<(u8, Vec<u8>)> {
let msg_type = reader.read_u8().await?;
let len = reader.read_u32().await?;
if len > MAX_PAYLOAD_SIZE {
let limit = if msg_type == AUDIO_MSG_HISTORY_COMPRESSED {
MAX_HISTORY_PAYLOAD_SIZE
} else {
MAX_PAYLOAD_SIZE
};
if len > limit {
return Err(std::io::Error::new(
std::io::ErrorKind::InvalidData,
format!("audio frame too large: {} bytes", len),
format!("audio frame too large: {} bytes (type={:#04x})", len, msg_type),
));
}
let mut payload = vec![0u8; len as usize];
+2 -1
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@@ -13,6 +13,7 @@ default = ["soapysdr"]
soapysdr = ["trx-backend/soapysdr"]
[dependencies]
flate2 = { workspace = true }
tokio = { workspace = true, features = ["full"] }
tokio-serial = { workspace = true }
serde = { workspace = true, features = ["derive"] }
@@ -37,4 +38,4 @@ trx-cw = { path = "../decoders/trx-cw" }
trx-decode-log = { path = "../decoders/trx-decode-log" }
trx-ft8 = { path = "../decoders/trx-ft8" }
trx-wspr = { path = "../decoders/trx-wspr" }
trx-protocol = { path = "../trx-protocol" }
trx-protocol = { path = "../trx-protocol" }
+23 -8
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@@ -11,7 +11,10 @@ use std::sync::{Arc, Mutex};
use std::time::{Duration, Instant};
use bytes::Bytes;
use flate2::write::GzEncoder;
use flate2::Compression;
use num_complex::Complex;
use std::io::Write as _;
use tokio::io::AsyncWriteExt;
use tokio::net::{TcpListener, TcpStream};
use tokio::sync::{broadcast, mpsc, watch};
@@ -22,8 +25,8 @@ use trx_aprs::AprsDecoder;
use trx_core::audio::{
read_audio_msg, write_audio_msg, AudioStreamInfo,
AUDIO_MSG_AIS_DECODE, AUDIO_MSG_APRS_DECODE, AUDIO_MSG_CW_DECODE, AUDIO_MSG_FT8_DECODE,
AUDIO_MSG_HF_APRS_DECODE, AUDIO_MSG_RX_FRAME, AUDIO_MSG_STREAM_INFO, AUDIO_MSG_TX_FRAME,
AUDIO_MSG_VDES_DECODE, AUDIO_MSG_WSPR_DECODE,
AUDIO_MSG_HF_APRS_DECODE, AUDIO_MSG_HISTORY_COMPRESSED, AUDIO_MSG_RX_FRAME,
AUDIO_MSG_STREAM_INFO, AUDIO_MSG_TX_FRAME, AUDIO_MSG_VDES_DECODE, AUDIO_MSG_WSPR_DECODE,
};
use trx_core::decode::{
AisMessage, AprsPacket, CwEvent, DecodedMessage, Ft8Message, VdesMessage, WsprMessage,
@@ -1856,15 +1859,27 @@ async fn handle_audio_client(
};
let (blob, replayed_history_count) = history_blob;
if !blob.is_empty() {
writer.write_all(&blob).await?;
writer.flush().await?;
}
if replayed_history_count > 0 {
// Gzip-compress the blob before sending. JSON history compresses very
// well (~10-20x) so this dramatically reduces both transfer size and
// the time the client spends waiting for data.
let compressed = {
let mut enc = GzEncoder::new(
Vec::with_capacity(blob.len() / 8),
Compression::fast(),
);
enc.write_all(&blob)
.and_then(|_| enc.finish())
.unwrap_or(blob.clone())
};
write_audio_msg(&mut writer, AUDIO_MSG_HISTORY_COMPRESSED, &compressed).await?;
info!(
"Audio client {} replayed {} history messages in {:?}",
"Audio client {} replayed {} history messages in {:?} ({} → {} bytes, {:.1}x)",
peer,
replayed_history_count,
history_replay_started_at.elapsed()
history_replay_started_at.elapsed(),
blob.len(),
compressed.len(),
blob.len() as f64 / compressed.len().max(1) as f64,
);
}