feat: Android VoIP client — Phase 1 (audio quality, network adaptation, crate skeleton)

- New wzp-android crate with Oboe C++ backend, lock-free SPSC ring buffers,
  engine orchestrator, codec pipeline, and Android Gradle project structure
- AEC (NLMS adaptive filter), AGC (two-stage with fast attack/slow release),
  windowed-sinc FIR resampler replacing linear interpolation (wzp-codec)
- Opus encoder tuning: complexity 7 default, set_expected_loss support
- Mobile jitter buffer: asymmetric EMA (fast up/slow down), handoff spike
  detection with 2s cooldown, configurable safety margin
- Network-aware quality control: cellular-specific thresholds, faster
  downgrade on cellular, proactive tier drop on WiFi→cellular handoff,
  FEC ratio boost during network transitions
- Handoff detection in PathMonitor via RTT jitter spike analysis

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
This commit is contained in:
Claude
2026-04-04 18:07:55 +00:00
parent aa09275015
commit 26e9c55f1f
31 changed files with 2775 additions and 245 deletions

202
Cargo.lock generated
View File

@@ -291,12 +291,6 @@ dependencies = [
"tower-service",
]
[[package]]
name = "base16ct"
version = "0.2.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "4c7f02d4ea65f2c1853089ffd8d2787bdbc63de2f0d29dedbcf8ccdfa0ccd4cf"
[[package]]
name = "base64"
version = "0.22.1"
@@ -467,7 +461,6 @@ dependencies = [
"iana-time-zone",
"js-sys",
"num-traits",
"serde",
"wasm-bindgen",
"windows-link",
]
@@ -628,24 +621,6 @@ dependencies = [
"libc",
]
[[package]]
name = "crunchy"
version = "0.2.4"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "460fbee9c2c2f33933d720630a6a0bac33ba7053db5344fac858d4b8952d77d5"
[[package]]
name = "crypto-bigint"
version = "0.5.5"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "0dc92fb57ca44df6db8059111ab3af99a63d5d0f8375d9972e319a379c6bab76"
dependencies = [
"generic-array",
"rand_core 0.6.4",
"subtle",
"zeroize",
]
[[package]]
name = "crypto-common"
version = "0.1.7"
@@ -669,7 +644,6 @@ dependencies = [
"digest",
"fiat-crypto",
"rustc_version",
"serde",
"subtle",
"zeroize",
]
@@ -836,7 +810,6 @@ source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "9ed9a281f7bc9b7576e61468ba615a66a5c8cfdff42420a70aa82701a3b1e292"
dependencies = [
"block-buffer",
"const-oid",
"crypto-common",
"subtle",
]
@@ -871,21 +844,6 @@ dependencies = [
"rustfft",
]
[[package]]
name = "ecdsa"
version = "0.16.9"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "ee27f32b5c5292967d2d4a9d7f1e0b0aed2c15daded5a60300e4abb9d8020bca"
dependencies = [
"der",
"digest",
"elliptic-curve",
"rfc6979",
"serdect",
"signature",
"spki",
]
[[package]]
name = "ed25519"
version = "2.2.3"
@@ -893,7 +851,6 @@ source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "115531babc129696a58c64a4fef0a8bf9e9698629fb97e9e40767d235cfbcd53"
dependencies = [
"pkcs8",
"serde",
"signature",
]
@@ -918,26 +875,6 @@ version = "1.15.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "48c757948c5ede0e46177b7add2e67155f70e33c07fea8284df6576da70b3719"
[[package]]
name = "elliptic-curve"
version = "0.13.8"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "b5e6043086bf7973472e0c7dff2142ea0b680d30e18d9cc40f267efbf222bd47"
dependencies = [
"base16ct",
"crypto-bigint",
"digest",
"ff",
"generic-array",
"group",
"pkcs8",
"rand_core 0.6.4",
"sec1",
"serdect",
"subtle",
"zeroize",
]
[[package]]
name = "encoding_rs"
version = "0.8.35"
@@ -981,16 +918,6 @@ version = "2.3.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "37909eebbb50d72f9059c3b6d82c0463f2ff062c9e95845c43a6c9c0355411be"
[[package]]
name = "ff"
version = "0.13.1"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "c0b50bfb653653f9ca9095b427bed08ab8d75a137839d9ad64eb11810d5b6393"
dependencies = [
"rand_core 0.6.4",
"subtle",
]
[[package]]
name = "fiat-crypto"
version = "0.2.9"
@@ -1151,7 +1078,6 @@ checksum = "85649ca51fd72272d7821adaf274ad91c288277713d9c18820d8499a7ff69e9a"
dependencies = [
"typenum",
"version_check",
"zeroize",
]
[[package]]
@@ -1211,17 +1137,6 @@ version = "0.3.3"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "0cc23270f6e1808e30a928bdc84dea0b9b4136a8bc82338574f23baf47bbd280"
[[package]]
name = "group"
version = "0.13.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "f0f9ef7462f7c099f518d754361858f86d8a07af53ba9af0fe635bbccb151a63"
dependencies = [
"ff",
"rand_core 0.6.4",
"subtle",
]
[[package]]
name = "h2"
version = "0.4.13"
@@ -1705,21 +1620,6 @@ dependencies = [
"wasm-bindgen",
]
[[package]]
name = "k256"
version = "0.13.4"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "f6e3919bbaa2945715f0bb6d3934a173d1e9a59ac23767fbaaef277265a7411b"
dependencies = [
"cfg-if",
"ecdsa",
"elliptic-curve",
"once_cell",
"serdect",
"sha2",
"signature",
]
[[package]]
name = "lazy_static"
version = "1.5.0"
@@ -2483,16 +2383,6 @@ dependencies = [
"web-sys",
]
[[package]]
name = "rfc6979"
version = "0.4.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "f8dd2a808d456c4a54e300a23e9f5a67e122c3024119acbfd73e3bf664491cb2"
dependencies = [
"hmac",
"subtle",
]
[[package]]
name = "ring"
version = "0.17.14"
@@ -2671,21 +2561,6 @@ version = "1.2.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "94143f37725109f92c262ed2cf5e59bce7498c01bcc1502d7b9afe439a4e9f49"
[[package]]
name = "sec1"
version = "0.7.3"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "d3e97a565f76233a6003f9f5c54be1d9c5bdfa3eccfb189469f11ec4901c47dc"
dependencies = [
"base16ct",
"der",
"generic-array",
"pkcs8",
"serdect",
"subtle",
"zeroize",
]
[[package]]
name = "security-framework"
version = "3.7.0"
@@ -2790,16 +2665,6 @@ dependencies = [
"serde",
]
[[package]]
name = "serdect"
version = "0.2.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "a84f14a19e9a014bb9f4512488d9829a68e04ecabffb0f9904cd1ace94598177"
dependencies = [
"base16ct",
"serde",
]
[[package]]
name = "sha1"
version = "0.10.6"
@@ -2853,7 +2718,6 @@ version = "2.2.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "77549399552de45a898a580c1b41d445bf730df867cc44e6c0233bbc4b8329de"
dependencies = [
"digest",
"rand_core 0.6.4",
]
@@ -3067,15 +2931,6 @@ version = "0.1.8"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "7694e1cfe791f8d31026952abf09c69ca6f6fa4e1a1229e18988f06a04a12dca"
[[package]]
name = "tiny-keccak"
version = "2.0.2"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "2c9d3793400a45f954c52e73d068316d76b6f4e36977e3fcebb13a2721e80237"
dependencies = [
"crunchy",
]
[[package]]
name = "tinystr"
version = "0.8.2"
@@ -3495,18 +3350,6 @@ version = "1.0.4"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "b6c140620e7ffbb22c2dee59cafe6084a59b5ffc27a8859a5f0d494b5d52b6be"
[[package]]
name = "uuid"
version = "1.23.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "5ac8b6f42ead25368cf5b098aeb3dc8a1a2c05a3eee8a9a1a68c640edbfc79d9"
dependencies = [
"getrandom 0.4.2",
"js-sys",
"serde_core",
"wasm-bindgen",
]
[[package]]
name = "valuable"
version = "0.1.1"
@@ -3546,28 +3389,7 @@ dependencies = [
[[package]]
name = "warzone-protocol"
version = "0.0.38"
dependencies = [
"base64",
"bincode",
"bip39",
"chacha20poly1305",
"chrono",
"curve25519-dalek",
"ed25519-dalek",
"hex",
"hkdf",
"k256",
"rand 0.8.5",
"serde",
"serde_json",
"sha2",
"thiserror 2.0.18",
"tiny-keccak",
"uuid",
"x25519-dalek",
"zeroize",
]
version = "0.1.0"
[[package]]
name = "wasi"
@@ -4179,6 +4001,28 @@ version = "0.6.2"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "9edde0db4769d2dc68579893f2306b26c6ecfbe0ef499b013d731b7b9247e0b9"
[[package]]
name = "wzp-android"
version = "0.1.0"
dependencies = [
"anyhow",
"async-trait",
"bytes",
"cc",
"libc",
"serde",
"serde_json",
"thiserror 2.0.18",
"tokio",
"tracing",
"tracing-subscriber",
"wzp-codec",
"wzp-crypto",
"wzp-fec",
"wzp-proto",
"wzp-transport",
]
[[package]]
name = "wzp-client"
version = "0.1.0"

View File

@@ -9,6 +9,7 @@ members = [
"crates/wzp-relay",
"crates/wzp-client",
"crates/wzp-web",
"crates/wzp-android",
]
[workspace.package]

View File

@@ -0,0 +1,64 @@
plugins {
id("com.android.application")
id("org.jetbrains.kotlin.android")
}
android {
namespace = "com.wzp.phone"
compileSdk = 34
defaultConfig {
applicationId = "com.wzp.phone"
minSdk = 26 // AAudio requires API 26
targetSdk = 34
versionCode = 1
versionName = "0.1.0"
ndk { abiFilters += listOf("arm64-v8a", "armeabi-v7a") }
}
buildTypes {
release {
isMinifyEnabled = true
proguardFiles(
getDefaultProguardFile("proguard-android-optimize.txt"),
"proguard-rules.pro"
)
}
}
compileOptions {
sourceCompatibility = JavaVersion.VERSION_1_8
targetCompatibility = JavaVersion.VERSION_1_8
}
kotlinOptions {
jvmTarget = "1.8"
}
buildFeatures { compose = true }
composeOptions { kotlinCompilerExtensionVersion = "1.5.8" }
ndkVersion = "26.1.10909125"
}
// cargo-ndk integration: build the Rust native library for Android targets
tasks.register<Exec>("cargoNdkBuild") {
workingDir = file("${project.rootDir}/..")
commandLine(
"cargo", "ndk",
"-t", "arm64-v8a", "-t", "armeabi-v7a",
"-o", "${project.projectDir}/src/main/jniLibs",
"build", "--release", "-p", "wzp-android"
)
}
tasks.named("preBuild") { dependsOn("cargoNdkBuild") }
dependencies {
implementation("androidx.core:core-ktx:1.12.0")
implementation("androidx.lifecycle:lifecycle-runtime-ktx:2.7.0")
implementation("androidx.activity:activity-compose:1.8.2")
implementation(platform("androidx.compose:compose-bom:2024.01.00"))
implementation("androidx.compose.ui:ui")
implementation("androidx.compose.material3:material3")
}

9
android/app/proguard-rules.pro vendored Normal file
View File

@@ -0,0 +1,9 @@
# WZPhone ProGuard rules
# Keep JNI native methods
-keepclasseswithmembernames class * {
native <methods>;
}
# Keep the WZP engine bridge class
-keep class com.wzp.phone.engine.** { *; }

View File

@@ -0,0 +1,33 @@
<?xml version="1.0" encoding="utf-8"?>
<manifest xmlns:android="http://schemas.android.com/apk/res/android">
<uses-permission android:name="android.permission.INTERNET" />
<uses-permission android:name="android.permission.RECORD_AUDIO" />
<uses-permission android:name="android.permission.FOREGROUND_SERVICE" />
<uses-permission android:name="android.permission.FOREGROUND_SERVICE_PHONE_CALL" />
<uses-permission android:name="android.permission.WAKE_LOCK" />
<uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" />
<uses-permission android:name="android.permission.BLUETOOTH_CONNECT" />
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
<application
android:name=".WzpApplication"
android:label="WZ Phone"
android:supportsRtl="true"
android:theme="@style/Theme.Material3.DayNight">
<activity
android:name=".ui.call.CallActivity"
android:exported="true"
android:launchMode="singleTask">
<intent-filter>
<action android:name="android.intent.action.MAIN" />
<category android:name="android.intent.category.LAUNCHER" />
</intent-filter>
</activity>
<service
android:name=".service.CallService"
android:foregroundServiceType="phoneCall"
android:exported="false" />
</application>
</manifest>

4
android/build.gradle.kts Normal file
View File

@@ -0,0 +1,4 @@
plugins {
id("com.android.application") version "8.2.0" apply false
id("org.jetbrains.kotlin.android") version "1.9.22" apply false
}

View File

@@ -0,0 +1,4 @@
org.gradle.jvmargs=-Xmx2048m -Dfile.encoding=UTF-8
android.useAndroidX=true
kotlin.code.style=official
android.nonTransitiveRClass=true

View File

@@ -0,0 +1,17 @@
pluginManagement {
repositories {
google()
mavenCentral()
gradlePluginPortal()
}
}
dependencyResolution {
repositories {
google()
mavenCentral()
}
}
rootProject.name = "WZPhone"
include(":app")

View File

@@ -0,0 +1,30 @@
[package]
name = "wzp-android"
version.workspace = true
edition.workspace = true
license.workspace = true
rust-version.workspace = true
description = "WarzonePhone Android native VoIP engine — Oboe audio, JNI bridge, call pipeline"
[lib]
crate-type = ["cdylib", "rlib"]
[dependencies]
wzp-proto = { workspace = true }
wzp-codec = { workspace = true }
wzp-fec = { workspace = true }
wzp-crypto = { workspace = true }
wzp-transport = { workspace = true }
tokio = { workspace = true }
tracing = { workspace = true }
tracing-subscriber = { workspace = true }
bytes = { workspace = true }
serde = { workspace = true }
serde_json = "1"
thiserror = { workspace = true }
async-trait = { workspace = true }
anyhow = "1"
libc = "0.2"
[build-dependencies]
cc = "1"

View File

@@ -0,0 +1,20 @@
fn main() {
let target = std::env::var("TARGET").unwrap_or_default();
if target.contains("android") {
// Real Oboe build for Android targets
cc::Build::new()
.cpp(true)
.std("c++17")
.file("cpp/oboe_bridge.cpp")
.include("cpp")
.compile("oboe_bridge");
} else {
// Stub for host builds / testing
cc::Build::new()
.cpp(true)
.std("c++17")
.file("cpp/oboe_stub.cpp")
.include("cpp")
.compile("oboe_bridge");
}
}

View File

@@ -0,0 +1,278 @@
// Full Oboe implementation for Android
// This file is compiled only when targeting Android
#include "oboe_bridge.h"
#ifdef __ANDROID__
#include <oboe/Oboe.h>
#include <android/log.h>
#include <cstring>
#include <atomic>
#define LOG_TAG "wzp-oboe"
#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
#define LOGW(...) __android_log_print(ANDROID_LOG_WARN, LOG_TAG, __VA_ARGS__)
#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
// ---------------------------------------------------------------------------
// Ring buffer helpers (SPSC, lock-free)
// ---------------------------------------------------------------------------
static inline int32_t ring_available_read(const wzp_atomic_int* write_idx,
const wzp_atomic_int* read_idx,
int32_t capacity) {
int32_t w = std::atomic_load_explicit(write_idx, std::memory_order_acquire);
int32_t r = std::atomic_load_explicit(read_idx, std::memory_order_relaxed);
int32_t avail = w - r;
if (avail < 0) avail += capacity;
return avail;
}
static inline int32_t ring_available_write(const wzp_atomic_int* write_idx,
const wzp_atomic_int* read_idx,
int32_t capacity) {
return capacity - 1 - ring_available_read(write_idx, read_idx, capacity);
}
static inline void ring_write(int16_t* buf, int32_t capacity,
wzp_atomic_int* write_idx, const wzp_atomic_int* read_idx,
const int16_t* src, int32_t count) {
int32_t w = std::atomic_load_explicit(write_idx, std::memory_order_relaxed);
for (int32_t i = 0; i < count; i++) {
buf[w] = src[i];
w++;
if (w >= capacity) w = 0;
}
std::atomic_store_explicit(write_idx, w, std::memory_order_release);
}
static inline void ring_read(int16_t* buf, int32_t capacity,
const wzp_atomic_int* write_idx, wzp_atomic_int* read_idx,
int16_t* dst, int32_t count) {
int32_t r = std::atomic_load_explicit(read_idx, std::memory_order_relaxed);
for (int32_t i = 0; i < count; i++) {
dst[i] = buf[r];
r++;
if (r >= capacity) r = 0;
}
std::atomic_store_explicit(read_idx, r, std::memory_order_release);
}
// ---------------------------------------------------------------------------
// Global state
// ---------------------------------------------------------------------------
static std::shared_ptr<oboe::AudioStream> g_capture_stream;
static std::shared_ptr<oboe::AudioStream> g_playout_stream;
static const WzpOboeRings* g_rings = nullptr;
static std::atomic<bool> g_running{false};
static std::atomic<float> g_capture_latency_ms{0.0f};
static std::atomic<float> g_playout_latency_ms{0.0f};
// ---------------------------------------------------------------------------
// Capture callback
// ---------------------------------------------------------------------------
class CaptureCallback : public oboe::AudioStreamDataCallback {
public:
oboe::DataCallbackResult onAudioReady(
oboe::AudioStream* stream,
void* audioData,
int32_t numFrames) override {
if (!g_running.load(std::std::memory_order_relaxed) || !g_rings) {
return oboe::DataCallbackResult::Stop;
}
const int16_t* src = static_cast<const int16_t*>(audioData);
int32_t avail = ring_available_write(g_rings->capture_write_idx,
g_rings->capture_read_idx,
g_rings->capture_capacity);
int32_t to_write = (numFrames < avail) ? numFrames : avail;
if (to_write > 0) {
ring_write(g_rings->capture_buf, g_rings->capture_capacity,
g_rings->capture_write_idx, g_rings->capture_read_idx,
src, to_write);
}
// Update latency estimate
auto result = stream->calculateLatencyMillis();
if (result) {
g_capture_latency_ms.store(static_cast<float>(result.value()),
std::std::memory_order_relaxed);
}
return oboe::DataCallbackResult::Continue;
}
};
// ---------------------------------------------------------------------------
// Playout callback
// ---------------------------------------------------------------------------
class PlayoutCallback : public oboe::AudioStreamDataCallback {
public:
oboe::DataCallbackResult onAudioReady(
oboe::AudioStream* stream,
void* audioData,
int32_t numFrames) override {
if (!g_running.load(std::std::memory_order_relaxed) || !g_rings) {
memset(audioData, 0, numFrames * sizeof(int16_t));
return oboe::DataCallbackResult::Stop;
}
int16_t* dst = static_cast<int16_t*>(audioData);
int32_t avail = ring_available_read(g_rings->playout_write_idx,
g_rings->playout_read_idx,
g_rings->playout_capacity);
int32_t to_read = (numFrames < avail) ? numFrames : avail;
if (to_read > 0) {
ring_read(g_rings->playout_buf, g_rings->playout_capacity,
g_rings->playout_write_idx, g_rings->playout_read_idx,
dst, to_read);
}
// Fill remainder with silence on underrun
if (to_read < numFrames) {
memset(dst + to_read, 0, (numFrames - to_read) * sizeof(int16_t));
}
// Update latency estimate
auto result = stream->calculateLatencyMillis();
if (result) {
g_playout_latency_ms.store(static_cast<float>(result.value()),
std::std::memory_order_relaxed);
}
return oboe::DataCallbackResult::Continue;
}
};
static CaptureCallback g_capture_cb;
static PlayoutCallback g_playout_cb;
// ---------------------------------------------------------------------------
// Public C API
// ---------------------------------------------------------------------------
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings) {
if (g_running.load(std::std::memory_order_relaxed)) {
LOGW("wzp_oboe_start: already running");
return -1;
}
g_rings = rings;
// Build capture stream
oboe::AudioStreamBuilder captureBuilder;
captureBuilder.setDirection(oboe::Direction::Input)
->setPerformanceMode(oboe::PerformanceMode::LowLatency)
->setSharingMode(oboe::SharingMode::Exclusive)
->setFormat(oboe::AudioFormat::I16)
->setChannelCount(config->channel_count)
->setSampleRate(config->sample_rate)
->setFramesPerDataCallback(config->frames_per_burst)
->setInputPreset(oboe::InputPreset::VoiceCommunication)
->setDataCallback(&g_capture_cb);
oboe::Result result = captureBuilder.openStream(g_capture_stream);
if (result != oboe::Result::OK) {
LOGE("Failed to open capture stream: %s", oboe::convertToText(result));
return -2;
}
// Build playout stream
oboe::AudioStreamBuilder playoutBuilder;
playoutBuilder.setDirection(oboe::Direction::Output)
->setPerformanceMode(oboe::PerformanceMode::LowLatency)
->setSharingMode(oboe::SharingMode::Exclusive)
->setFormat(oboe::AudioFormat::I16)
->setChannelCount(config->channel_count)
->setSampleRate(config->sample_rate)
->setFramesPerDataCallback(config->frames_per_burst)
->setUsage(oboe::Usage::VoiceCommunication)
->setDataCallback(&g_playout_cb);
result = playoutBuilder.openStream(g_playout_stream);
if (result != oboe::Result::OK) {
LOGE("Failed to open playout stream: %s", oboe::convertToText(result));
g_capture_stream->close();
g_capture_stream.reset();
return -3;
}
g_running.store(true, std::std::memory_order_release);
// Start both streams
result = g_capture_stream->requestStart();
if (result != oboe::Result::OK) {
LOGE("Failed to start capture: %s", oboe::convertToText(result));
g_running.store(false, std::std::memory_order_release);
g_capture_stream->close();
g_playout_stream->close();
g_capture_stream.reset();
g_playout_stream.reset();
return -4;
}
result = g_playout_stream->requestStart();
if (result != oboe::Result::OK) {
LOGE("Failed to start playout: %s", oboe::convertToText(result));
g_running.store(false, std::std::memory_order_release);
g_capture_stream->requestStop();
g_capture_stream->close();
g_playout_stream->close();
g_capture_stream.reset();
g_playout_stream.reset();
return -5;
}
LOGI("Oboe started: sr=%d burst=%d ch=%d",
config->sample_rate, config->frames_per_burst, config->channel_count);
return 0;
}
void wzp_oboe_stop(void) {
g_running.store(false, std::std::memory_order_release);
if (g_capture_stream) {
g_capture_stream->requestStop();
g_capture_stream->close();
g_capture_stream.reset();
}
if (g_playout_stream) {
g_playout_stream->requestStop();
g_playout_stream->close();
g_playout_stream.reset();
}
g_rings = nullptr;
LOGI("Oboe stopped");
}
float wzp_oboe_capture_latency_ms(void) {
return g_capture_latency_ms.load(std::std::memory_order_relaxed);
}
float wzp_oboe_playout_latency_ms(void) {
return g_playout_latency_ms.load(std::std::memory_order_relaxed);
}
int wzp_oboe_is_running(void) {
return g_running.load(std::std::memory_order_relaxed) ? 1 : 0;
}
#else
// Non-Android fallback — should not be reached; oboe_stub.cpp is used instead.
// Provide empty implementations just in case.
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings) {
(void)config; (void)rings;
return -99;
}
void wzp_oboe_stop(void) {}
float wzp_oboe_capture_latency_ms(void) { return 0.0f; }
float wzp_oboe_playout_latency_ms(void) { return 0.0f; }
int wzp_oboe_is_running(void) { return 0; }
#endif // __ANDROID__

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#ifndef WZP_OBOE_BRIDGE_H
#define WZP_OBOE_BRIDGE_H
#include <stdint.h>
#ifdef __cplusplus
#include <atomic>
typedef std::atomic<int32_t> wzp_atomic_int;
extern "C" {
#else
#include <stdatomic.h>
typedef atomic_int wzp_atomic_int;
#endif
typedef struct {
int32_t sample_rate;
int32_t frames_per_burst;
int32_t channel_count;
} WzpOboeConfig;
typedef struct {
int16_t* capture_buf;
int32_t capture_capacity;
wzp_atomic_int* capture_write_idx;
wzp_atomic_int* capture_read_idx;
int16_t* playout_buf;
int32_t playout_capacity;
wzp_atomic_int* playout_write_idx;
wzp_atomic_int* playout_read_idx;
} WzpOboeRings;
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings);
void wzp_oboe_stop(void);
float wzp_oboe_capture_latency_ms(void);
float wzp_oboe_playout_latency_ms(void);
int wzp_oboe_is_running(void);
#ifdef __cplusplus
}
#endif
#endif // WZP_OBOE_BRIDGE_H

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// Stub implementation for non-Android host builds (testing, cargo check, etc.)
#include "oboe_bridge.h"
#include <stdio.h>
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings) {
(void)config;
(void)rings;
fprintf(stderr, "wzp_oboe_start: stub (not on Android)\n");
return 0;
}
void wzp_oboe_stop(void) {
fprintf(stderr, "wzp_oboe_stop: stub (not on Android)\n");
}
float wzp_oboe_capture_latency_ms(void) {
return 0.0f;
}
float wzp_oboe_playout_latency_ms(void) {
return 0.0f;
}
int wzp_oboe_is_running(void) {
return 0;
}

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//! Lock-free SPSC ring buffer audio backend for Android (Oboe).
//!
//! The ring buffers are shared between Rust and C++: the Oboe callbacks
//! (running on a high-priority audio thread) read/write directly into
//! the buffers via atomic indices, while the Rust codec thread on the
//! other side does the same.
use std::sync::atomic::{AtomicI32, Ordering};
use tracing::info;
#[allow(unused_imports)]
use tracing::warn;
/// Number of samples per 20 ms frame at 48 kHz mono.
pub const FRAME_SAMPLES: usize = 960;
/// Default ring buffer capacity: 8 frames = 160 ms at 48 kHz.
const RING_CAPACITY: usize = 7680;
// ---------------------------------------------------------------------------
// FFI declarations matching oboe_bridge.h
// ---------------------------------------------------------------------------
#[repr(C)]
#[allow(non_snake_case)]
struct WzpOboeConfig {
sample_rate: i32,
frames_per_burst: i32,
channel_count: i32,
}
#[repr(C)]
#[allow(non_snake_case)]
struct WzpOboeRings {
capture_buf: *mut i16,
capture_capacity: i32,
capture_write_idx: *mut AtomicI32,
capture_read_idx: *mut AtomicI32,
playout_buf: *mut i16,
playout_capacity: i32,
playout_write_idx: *mut AtomicI32,
playout_read_idx: *mut AtomicI32,
}
unsafe impl Send for WzpOboeRings {}
unsafe impl Sync for WzpOboeRings {}
unsafe extern "C" {
fn wzp_oboe_start(config: *const WzpOboeConfig, rings: *const WzpOboeRings) -> i32;
fn wzp_oboe_stop();
fn wzp_oboe_capture_latency_ms() -> f32;
fn wzp_oboe_playout_latency_ms() -> f32;
fn wzp_oboe_is_running() -> i32;
}
// ---------------------------------------------------------------------------
// SPSC Ring Buffer
// ---------------------------------------------------------------------------
/// Single-producer single-consumer lock-free ring buffer.
///
/// The producer calls `write()` and the consumer calls `read()`.
/// Atomics use acquire/release ordering to ensure correct visibility
/// across the Oboe audio thread and the Rust codec thread.
pub struct RingBuffer {
buf: Vec<i16>,
capacity: usize,
write_idx: AtomicI32,
read_idx: AtomicI32,
}
impl RingBuffer {
/// Create a new ring buffer with the given capacity (in samples).
///
/// The actual usable capacity is `capacity - 1` to distinguish
/// full from empty.
pub fn new(capacity: usize) -> Self {
Self {
buf: vec![0i16; capacity],
capacity,
write_idx: AtomicI32::new(0),
read_idx: AtomicI32::new(0),
}
}
/// Number of samples available to read.
pub fn available_read(&self) -> usize {
let w = self.write_idx.load(Ordering::Acquire);
let r = self.read_idx.load(Ordering::Relaxed);
let avail = w - r;
if avail < 0 {
(avail + self.capacity as i32) as usize
} else {
avail as usize
}
}
/// Number of samples that can be written before the buffer is full.
pub fn available_write(&self) -> usize {
self.capacity - 1 - self.available_read()
}
/// Write samples into the ring buffer (producer side).
///
/// Returns the number of samples actually written (may be less than
/// `data.len()` if the buffer is nearly full).
pub fn write(&self, data: &[i16]) -> usize {
let avail = self.available_write();
let count = data.len().min(avail);
if count == 0 {
return 0;
}
let mut w = self.write_idx.load(Ordering::Relaxed) as usize;
let cap = self.capacity;
let buf_ptr = self.buf.as_ptr() as *mut i16;
for i in 0..count {
// SAFETY: w is always in [0, capacity) and we are the sole producer.
unsafe {
*buf_ptr.add(w) = data[i];
}
w += 1;
if w >= cap {
w = 0;
}
}
self.write_idx.store(w as i32, Ordering::Release);
count
}
/// Read samples from the ring buffer (consumer side).
///
/// Returns the number of samples actually read (may be less than
/// `out.len()` if the buffer doesn't have enough data).
pub fn read(&self, out: &mut [i16]) -> usize {
let avail = self.available_read();
let count = out.len().min(avail);
if count == 0 {
return 0;
}
let mut r = self.read_idx.load(Ordering::Relaxed) as usize;
let cap = self.capacity;
let buf_ptr = self.buf.as_ptr();
for i in 0..count {
// SAFETY: r is always in [0, capacity) and we are the sole consumer.
unsafe {
out[i] = *buf_ptr.add(r);
}
r += 1;
if r >= cap {
r = 0;
}
}
self.read_idx.store(r as i32, Ordering::Release);
count
}
/// Get a raw pointer to the buffer data (for FFI).
fn buf_ptr(&self) -> *mut i16 {
self.buf.as_ptr() as *mut i16
}
/// Get a raw pointer to the write index atomic (for FFI).
fn write_idx_ptr(&self) -> *mut AtomicI32 {
&self.write_idx as *const AtomicI32 as *mut AtomicI32
}
/// Get a raw pointer to the read index atomic (for FFI).
fn read_idx_ptr(&self) -> *mut AtomicI32 {
&self.read_idx as *const AtomicI32 as *mut AtomicI32
}
}
// SAFETY: The ring buffer is designed for SPSC use where producer and consumer
// are on different threads. The atomic indices provide the synchronization.
unsafe impl Send for RingBuffer {}
unsafe impl Sync for RingBuffer {}
// ---------------------------------------------------------------------------
// Oboe Backend
// ---------------------------------------------------------------------------
/// Oboe-based audio backend for Android.
///
/// Owns two SPSC ring buffers (capture and playout) that are shared with
/// the C++ Oboe callbacks via raw pointers. The Oboe callbacks run on
/// high-priority audio threads managed by the Android audio system.
pub struct OboeBackend {
capture_ring: RingBuffer,
playout_ring: RingBuffer,
started: bool,
}
impl OboeBackend {
/// Create a new backend with default ring buffer sizes (160 ms each).
pub fn new() -> Self {
Self {
capture_ring: RingBuffer::new(RING_CAPACITY),
playout_ring: RingBuffer::new(RING_CAPACITY),
started: false,
}
}
/// Start Oboe audio streams.
///
/// This sets up the ring buffer pointers and calls into the C++ layer
/// to open and start the capture and playout Oboe streams.
pub fn start(&mut self) -> Result<(), anyhow::Error> {
if self.started {
return Ok(());
}
let config = WzpOboeConfig {
sample_rate: 48_000,
frames_per_burst: FRAME_SAMPLES as i32,
channel_count: 1,
};
let rings = WzpOboeRings {
capture_buf: self.capture_ring.buf_ptr(),
capture_capacity: self.capture_ring.capacity as i32,
capture_write_idx: self.capture_ring.write_idx_ptr(),
capture_read_idx: self.capture_ring.read_idx_ptr(),
playout_buf: self.playout_ring.buf_ptr(),
playout_capacity: self.playout_ring.capacity as i32,
playout_write_idx: self.playout_ring.write_idx_ptr(),
playout_read_idx: self.playout_ring.read_idx_ptr(),
};
let ret = unsafe { wzp_oboe_start(&config, &rings) };
if ret != 0 {
return Err(anyhow::anyhow!("wzp_oboe_start failed with code {}", ret));
}
self.started = true;
info!("Oboe backend started");
Ok(())
}
/// Stop Oboe audio streams.
pub fn stop(&mut self) {
if !self.started {
return;
}
unsafe { wzp_oboe_stop() };
self.started = false;
info!("Oboe backend stopped");
}
/// Read captured audio samples from the capture ring buffer.
///
/// Returns the number of samples actually read. The caller should
/// provide a buffer of at least `FRAME_SAMPLES` (960) samples.
pub fn read_capture(&self, out: &mut [i16]) -> usize {
self.capture_ring.read(out)
}
/// Write audio samples to the playout ring buffer.
///
/// Returns the number of samples actually written.
pub fn write_playout(&self, samples: &[i16]) -> usize {
self.playout_ring.write(samples)
}
/// Get the current capture latency in milliseconds (from Oboe).
#[allow(unused)]
pub fn capture_latency_ms(&self) -> f32 {
unsafe { wzp_oboe_capture_latency_ms() }
}
/// Get the current playout latency in milliseconds (from Oboe).
#[allow(unused)]
pub fn playout_latency_ms(&self) -> f32 {
unsafe { wzp_oboe_playout_latency_ms() }
}
/// Check if the Oboe streams are currently running.
#[allow(unused)]
pub fn is_running(&self) -> bool {
unsafe { wzp_oboe_is_running() != 0 }
}
}
impl Drop for OboeBackend {
fn drop(&mut self) {
self.stop();
}
}
// ---------------------------------------------------------------------------
// Thread affinity / priority helpers
// ---------------------------------------------------------------------------
/// Pin the current thread to the highest-numbered CPU cores (big cores on
/// ARM big.LITTLE architectures). Falls back silently on failure.
#[allow(unused)]
pub fn pin_to_big_core() {
#[cfg(target_os = "android")]
{
unsafe {
let num_cpus = libc::sysconf(libc::_SC_NPROCESSORS_ONLN);
if num_cpus <= 0 {
warn!("pin_to_big_core: could not determine CPU count");
return;
}
let num_cpus = num_cpus as usize;
// Target the upper half of CPUs (big cores on most big.LITTLE SoCs)
let start = num_cpus / 2;
let mut set: libc::cpu_set_t = std::mem::zeroed();
libc::CPU_ZERO(&mut set);
for cpu in start..num_cpus {
libc::CPU_SET(cpu, &mut set);
}
let ret = libc::sched_setaffinity(
0, // current thread
std::mem::size_of::<libc::cpu_set_t>(),
&set,
);
if ret != 0 {
warn!("sched_setaffinity failed: {}", std::io::Error::last_os_error());
} else {
info!(start, num_cpus, "pinned to big cores");
}
}
}
#[cfg(not(target_os = "android"))]
{
// No-op on non-Android
}
}
/// Attempt to set SCHED_FIFO real-time priority for the current thread.
/// Falls back silently on failure (requires appropriate permissions on Android).
#[allow(unused)]
pub fn set_realtime_priority() {
#[cfg(target_os = "android")]
{
unsafe {
let param = libc::sched_param {
sched_priority: 2, // Low RT priority — enough for audio, safe
};
let ret = libc::sched_setscheduler(0, libc::SCHED_FIFO, &param);
if ret != 0 {
warn!(
"sched_setscheduler(SCHED_FIFO) failed: {}",
std::io::Error::last_os_error()
);
} else {
info!("set SCHED_FIFO priority 2");
}
}
}
#[cfg(not(target_os = "android"))]
{
// No-op on non-Android
}
}
// ---------------------------------------------------------------------------
// Tests
// ---------------------------------------------------------------------------
#[cfg(test)]
mod tests {
use super::*;
#[test]
fn ring_buffer_write_read() {
let ring = RingBuffer::new(16);
let data = [1i16, 2, 3, 4, 5];
assert_eq!(ring.write(&data), 5);
assert_eq!(ring.available_read(), 5);
let mut out = [0i16; 5];
assert_eq!(ring.read(&mut out), 5);
assert_eq!(out, [1, 2, 3, 4, 5]);
assert_eq!(ring.available_read(), 0);
}
#[test]
fn ring_buffer_wraparound() {
let ring = RingBuffer::new(8);
let data = [10i16, 20, 30, 40, 50, 60]; // 6 samples, capacity 8 (usable 7)
assert_eq!(ring.write(&data), 6);
let mut out = [0i16; 4];
assert_eq!(ring.read(&mut out), 4);
assert_eq!(out, [10, 20, 30, 40]);
// Now write more, which should wrap around
let data2 = [70i16, 80, 90, 100];
assert_eq!(ring.write(&data2), 4);
let mut out2 = [0i16; 6];
assert_eq!(ring.read(&mut out2), 6);
assert_eq!(out2, [50, 60, 70, 80, 90, 100]);
}
#[test]
fn ring_buffer_full() {
let ring = RingBuffer::new(4); // usable capacity = 3
let data = [1i16, 2, 3, 4, 5];
assert_eq!(ring.write(&data), 3); // Only 3 fit
assert_eq!(ring.available_write(), 0);
}
#[test]
fn oboe_backend_stub_start_stop() {
let mut backend = OboeBackend::new();
backend.start().expect("stub start should succeed");
assert!(backend.started);
backend.stop();
assert!(!backend.started);
}
}

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//! Engine commands sent from the JNI/UI thread to the engine.
use wzp_proto::QualityProfile;
/// Commands that can be sent to the running engine.
pub enum EngineCommand {
/// Mute or unmute the microphone.
SetMute(bool),
/// Enable or disable speaker (loudspeaker) mode.
SetSpeaker(bool),
/// Force a specific quality profile (overrides adaptive logic).
ForceProfile(QualityProfile),
/// Stop the call and shut down the engine.
Stop,
}

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//! Engine orchestrator — manages the call lifecycle.
//!
//! The engine owns:
//! - The Oboe audio backend (start/stop)
//! - A codec thread running the `Pipeline`
//! - A tokio runtime for async network I/O
//! - Command channel for control from the JNI/UI thread
use std::sync::atomic::{AtomicBool, Ordering};
use std::sync::{Arc, Mutex};
use std::time::Instant;
use tracing::{error, info, warn};
use wzp_proto::QualityProfile;
use crate::audio_android::{OboeBackend, FRAME_SAMPLES};
use crate::commands::EngineCommand;
use crate::pipeline::Pipeline;
use crate::stats::{CallState, CallStats};
/// Configuration to start a call.
pub struct CallStartConfig {
/// Initial quality profile.
pub profile: QualityProfile,
/// Relay server address (host:port).
pub relay_addr: String,
/// Authentication token for the relay.
pub auth_token: Vec<u8>,
/// 32-byte identity seed for key derivation.
pub identity_seed: [u8; 32],
}
impl Default for CallStartConfig {
fn default() -> Self {
Self {
profile: QualityProfile::GOOD,
relay_addr: String::new(),
auth_token: Vec::new(),
identity_seed: [0u8; 32],
}
}
}
/// Shared state between the engine owner and background threads.
struct EngineState {
running: AtomicBool,
muted: AtomicBool,
speaker: AtomicBool,
stats: Mutex<CallStats>,
command_tx: std::sync::mpsc::Sender<EngineCommand>,
command_rx: Mutex<Option<std::sync::mpsc::Receiver<EngineCommand>>>,
}
/// The WarzonePhone Android engine.
///
/// Manages the entire call pipeline: audio capture/playout via Oboe,
/// codec encode/decode, FEC, jitter buffer, and network transport.
///
/// Thread model:
/// - **UI/JNI thread**: calls `start_call`, `stop_call`, `set_mute`, etc.
/// - **Codec thread**: runs `Pipeline` encode/decode loop, reads/writes ring buffers
/// - **Tokio runtime** (2 worker threads): async network send/recv
pub struct WzpEngine {
state: Arc<EngineState>,
codec_thread: Option<std::thread::JoinHandle<()>>,
#[allow(unused)]
tokio_runtime: Option<tokio::runtime::Runtime>,
call_start: Option<Instant>,
}
impl WzpEngine {
/// Create a new idle engine.
pub fn new() -> Self {
let (tx, rx) = std::sync::mpsc::channel();
let state = Arc::new(EngineState {
running: AtomicBool::new(false),
muted: AtomicBool::new(false),
speaker: AtomicBool::new(false),
stats: Mutex::new(CallStats::default()),
command_tx: tx,
command_rx: Mutex::new(Some(rx)),
});
Self {
state,
codec_thread: None,
tokio_runtime: None,
call_start: None,
}
}
/// Start a call with the given configuration.
///
/// This creates the tokio runtime, starts the Oboe audio backend,
/// and spawns the codec thread.
pub fn start_call(&mut self, config: CallStartConfig) -> Result<(), anyhow::Error> {
if self.state.running.load(Ordering::Acquire) {
return Err(anyhow::anyhow!("call already active"));
}
// Update state
{
let mut stats = self.state.stats.lock().unwrap();
*stats = CallStats {
state: CallState::Connecting,
..Default::default()
};
}
// Create tokio runtime with 2 worker threads
let runtime = tokio::runtime::Builder::new_multi_thread()
.worker_threads(2)
.thread_name("wzp-net")
.enable_all()
.build()?;
// Create async channels for network send/recv
let (send_tx, mut _send_rx) = tokio::sync::mpsc::channel::<Vec<u8>>(64);
let (_recv_tx, mut recv_rx) = tokio::sync::mpsc::channel::<Vec<u8>>(64);
// Spawn network tasks (placeholder — will use wzp-transport)
let _relay_addr = config.relay_addr.clone();
runtime.spawn(async move {
// Network send task: reads from send_rx, sends via transport
// This will be implemented when wzp-transport Android support is added
while let Some(_packet) = _send_rx.recv().await {
// TODO: send via wzp-transport
}
});
let recv_tx_clone = _recv_tx.clone();
runtime.spawn(async move {
// Network recv task: reads from transport, writes to recv_rx
// This will be implemented when wzp-transport Android support is added
let _tx = recv_tx_clone;
// TODO: recv from wzp-transport and forward
});
// Take the command receiver (it can only be taken once)
let command_rx = self
.state
.command_rx
.lock()
.unwrap()
.take()
.ok_or_else(|| anyhow::anyhow!("command receiver already taken"))?;
// Start the codec thread
let state = self.state.clone();
let profile = config.profile;
let codec_thread = std::thread::Builder::new()
.name("wzp-codec".into())
.spawn(move || {
// Pin to big cores and set RT priority on Android
crate::audio_android::pin_to_big_core();
crate::audio_android::set_realtime_priority();
// Create audio backend
let mut audio = OboeBackend::new();
if let Err(e) = audio.start() {
error!("failed to start audio: {e}");
state.running.store(false, Ordering::Release);
return;
}
// Create pipeline
let mut pipeline = match Pipeline::new(profile) {
Ok(p) => p,
Err(e) => {
error!("failed to create pipeline: {e}");
audio.stop();
state.running.store(false, Ordering::Release);
return;
}
};
state.running.store(true, Ordering::Release);
{
let mut stats = state.stats.lock().unwrap();
stats.state = CallState::Active;
}
info!("codec thread started");
let mut capture_buf = vec![0i16; FRAME_SAMPLES];
#[allow(unused_assignments)]
let mut recv_buf: Vec<u8> = Vec::new();
// Main codec loop: 20ms per iteration
let frame_duration = std::time::Duration::from_millis(20);
while state.running.load(Ordering::Relaxed) {
let loop_start = Instant::now();
// Process commands (non-blocking)
while let Ok(cmd) = command_rx.try_recv() {
match cmd {
EngineCommand::SetMute(m) => {
state.muted.store(m, Ordering::Relaxed);
info!(muted = m, "mute toggled");
}
EngineCommand::SetSpeaker(s) => {
state.speaker.store(s, Ordering::Relaxed);
info!(speaker = s, "speaker toggled");
}
EngineCommand::ForceProfile(p) => {
pipeline.force_profile(p);
info!(?p, "profile forced");
}
EngineCommand::Stop => {
info!("stop command received");
state.running.store(false, Ordering::Release);
break;
}
}
}
if !state.running.load(Ordering::Relaxed) {
break;
}
// --- Capture → Encode → Send ---
let captured = audio.read_capture(&mut capture_buf);
if captured >= FRAME_SAMPLES {
let muted = state.muted.load(Ordering::Relaxed);
if let Some(encoded) = pipeline.encode_frame(&capture_buf, muted) {
// Send to network (best-effort)
let _ = send_tx.try_send(encoded);
}
}
// --- Recv → Decode → Playout ---
// Drain received packets from the network channel
while let Ok(data) = recv_rx.try_recv() {
recv_buf = data;
// Deserialize the packet and feed to pipeline
// For now, feed raw bytes — full MediaPacket deserialization
// will be added with the transport integration
let _ = &recv_buf; // suppress unused warning
}
// Decode from jitter buffer
if let Some(pcm) = pipeline.decode_frame() {
audio.write_playout(&pcm);
}
// --- Update stats ---
{
let pstats = pipeline.stats();
let mut stats = state.stats.lock().unwrap();
stats.frames_encoded = pstats.frames_encoded;
stats.frames_decoded = pstats.frames_decoded;
stats.underruns = pstats.underruns;
stats.jitter_buffer_depth = pstats.jitter_depth;
stats.quality_tier = pstats.quality_tier;
}
// Sleep for remainder of the 20ms frame period
let elapsed = loop_start.elapsed();
if elapsed < frame_duration {
std::thread::sleep(frame_duration - elapsed);
}
}
// Cleanup
audio.stop();
{
let mut stats = state.stats.lock().unwrap();
stats.state = CallState::Closed;
}
info!("codec thread exited");
})?;
self.codec_thread = Some(codec_thread);
self.tokio_runtime = Some(runtime);
self.call_start = Some(Instant::now());
info!("call started");
Ok(())
}
/// Stop the current call and clean up all resources.
pub fn stop_call(&mut self) {
if !self.state.running.load(Ordering::Acquire) {
return;
}
// Signal stop
self.state.running.store(false, Ordering::Release);
let _ = self.state.command_tx.send(EngineCommand::Stop);
// Join codec thread
if let Some(handle) = self.codec_thread.take() {
if let Err(e) = handle.join() {
warn!("codec thread panicked: {e:?}");
}
}
// Shut down tokio runtime
if let Some(rt) = self.tokio_runtime.take() {
rt.shutdown_timeout(std::time::Duration::from_secs(2));
}
self.call_start = None;
info!("call stopped");
}
/// Set microphone mute state.
pub fn set_mute(&self, muted: bool) {
let _ = self.state.command_tx.send(EngineCommand::SetMute(muted));
}
/// Set speaker (loudspeaker) mode.
#[allow(unused)]
pub fn set_speaker(&self, enabled: bool) {
let _ = self
.state
.command_tx
.send(EngineCommand::SetSpeaker(enabled));
}
/// Force a specific quality profile (overrides adaptive logic).
#[allow(unused)]
pub fn force_profile(&self, profile: QualityProfile) {
let _ = self
.state
.command_tx
.send(EngineCommand::ForceProfile(profile));
}
/// Get a snapshot of the current call statistics.
pub fn get_stats(&self) -> CallStats {
let mut stats = self.state.stats.lock().unwrap().clone();
// Update duration from wall clock
if let Some(start) = self.call_start {
stats.duration_secs = start.elapsed().as_secs_f64();
}
stats
}
/// Check if a call is currently active.
pub fn is_active(&self) -> bool {
self.state.running.load(Ordering::Acquire)
}
/// Destroy the engine, stopping any active call.
pub fn destroy(mut self) {
self.stop_call();
info!("engine destroyed");
}
}
impl Drop for WzpEngine {
fn drop(&mut self) {
self.stop_call();
}
}

View File

@@ -0,0 +1,17 @@
//! WarzonePhone Android native VoIP engine.
//!
//! Provides:
//! - Oboe audio backend with lock-free SPSC ring buffers
//! - Engine orchestrator managing call lifecycle
//! - Codec pipeline thread (encode/decode/FEC/jitter)
//! - Call statistics and command interface
//!
//! On non-Android targets, the Oboe C++ layer compiles as a stub,
//! allowing `cargo check` and unit tests on the host.
pub mod audio_android;
pub mod commands;
pub mod engine;
pub mod pipeline;
pub mod stats;
// pub mod jni_bridge; // Added later by Agent 4

View File

@@ -0,0 +1,224 @@
//! Codec pipeline — encode/decode with FEC and jitter buffer.
//!
//! Runs on a dedicated thread, processing 20 ms frames at 48 kHz.
//! The pipeline is NOT Send/Sync (Opus encoder state) — it is owned
//! exclusively by the codec thread.
use tracing::{debug, warn};
use wzp_codec::{AdaptiveDecoder, AdaptiveEncoder};
use wzp_fec::{RaptorQFecDecoder, RaptorQFecEncoder};
use wzp_proto::jitter::{JitterBuffer, PlayoutResult};
use wzp_proto::quality::AdaptiveQualityController;
use wzp_proto::traits::{AudioDecoder, AudioEncoder, FecDecoder, FecEncoder};
use wzp_proto::traits::QualityController;
use wzp_proto::{MediaPacket, QualityProfile};
use crate::audio_android::FRAME_SAMPLES;
/// Maximum encoded frame size (Opus worst case at highest bitrate).
const MAX_ENCODED_BYTES: usize = 1275;
/// Pipeline statistics snapshot.
#[derive(Clone, Debug, Default)]
pub struct PipelineStats {
pub frames_encoded: u64,
pub frames_decoded: u64,
pub underruns: u64,
pub jitter_depth: usize,
pub quality_tier: u8,
}
/// The codec pipeline: encode, FEC, jitter buffer, decode.
///
/// This struct is owned by the codec thread and not shared.
pub struct Pipeline {
encoder: AdaptiveEncoder,
decoder: AdaptiveDecoder,
fec_encoder: RaptorQFecEncoder,
fec_decoder: RaptorQFecDecoder,
jitter_buffer: JitterBuffer,
quality_ctrl: AdaptiveQualityController,
// Pre-allocated scratch buffers
capture_buf: Vec<i16>,
#[allow(dead_code)]
playout_buf: Vec<i16>,
encode_out: Vec<u8>,
// Stats counters
frames_encoded: u64,
frames_decoded: u64,
underruns: u64,
}
impl Pipeline {
/// Create a new pipeline configured for the given quality profile.
pub fn new(profile: QualityProfile) -> Result<Self, anyhow::Error> {
let encoder = AdaptiveEncoder::new(profile)
.map_err(|e| anyhow::anyhow!("encoder init: {e}"))?;
let decoder = AdaptiveDecoder::new(profile)
.map_err(|e| anyhow::anyhow!("decoder init: {e}"))?;
let fec_encoder =
RaptorQFecEncoder::with_defaults(profile.frames_per_block as usize);
let fec_decoder =
RaptorQFecDecoder::with_defaults(profile.frames_per_block as usize);
let jitter_buffer = JitterBuffer::new(10, 250, 3);
let quality_ctrl = AdaptiveQualityController::new();
Ok(Self {
encoder,
decoder,
fec_encoder,
fec_decoder,
jitter_buffer,
quality_ctrl,
capture_buf: vec![0i16; FRAME_SAMPLES],
playout_buf: vec![0i16; FRAME_SAMPLES],
encode_out: vec![0u8; MAX_ENCODED_BYTES],
frames_encoded: 0,
frames_decoded: 0,
underruns: 0,
})
}
/// Encode a PCM frame into a compressed packet.
///
/// If `muted` is true, a silence frame is encoded (all zeros).
/// Returns the encoded bytes, or `None` on encoder error.
pub fn encode_frame(&mut self, pcm: &[i16], muted: bool) -> Option<Vec<u8>> {
let input = if muted {
// Zero the capture buffer for silence
for s in self.capture_buf.iter_mut() {
*s = 0;
}
&self.capture_buf[..]
} else {
pcm
};
match self.encoder.encode(input, &mut self.encode_out) {
Ok(n) => {
self.frames_encoded += 1;
let encoded = self.encode_out[..n].to_vec();
// Feed into FEC encoder
if let Err(e) = self.fec_encoder.add_source_symbol(&encoded) {
warn!("FEC encode error: {e}");
}
Some(encoded)
}
Err(e) => {
warn!("encode error: {e}");
None
}
}
}
/// Feed a received media packet into the jitter buffer.
pub fn feed_packet(&mut self, packet: MediaPacket) {
// Feed FEC symbols if present
let header = &packet.header;
if header.fec_block != 0 || header.fec_symbol != 0 {
let is_repair = header.is_repair;
if let Err(e) = self.fec_decoder.add_symbol(
header.fec_block,
header.fec_symbol,
is_repair,
&packet.payload,
) {
debug!("FEC symbol feed error: {e}");
}
}
self.jitter_buffer.push(packet);
}
/// Decode the next frame from the jitter buffer.
///
/// Returns decoded PCM samples, or `None` if the buffer is not ready.
pub fn decode_frame(&mut self) -> Option<Vec<i16>> {
match self.jitter_buffer.pop() {
PlayoutResult::Packet(pkt) => {
let mut pcm = vec![0i16; FRAME_SAMPLES];
match self.decoder.decode(&pkt.payload, &mut pcm) {
Ok(n) => {
self.frames_decoded += 1;
pcm.truncate(n);
Some(pcm)
}
Err(e) => {
warn!("decode error: {e}");
// Attempt PLC
self.generate_plc()
}
}
}
PlayoutResult::Missing { seq } => {
debug!(seq, "jitter buffer: missing packet, generating PLC");
self.generate_plc()
}
PlayoutResult::NotReady => {
self.underruns += 1;
None
}
}
}
/// Generate packet loss concealment output.
fn generate_plc(&mut self) -> Option<Vec<i16>> {
let mut pcm = vec![0i16; FRAME_SAMPLES];
match self.decoder.decode_lost(&mut pcm) {
Ok(n) => {
self.frames_decoded += 1;
pcm.truncate(n);
Some(pcm)
}
Err(e) => {
warn!("PLC error: {e}");
None
}
}
}
/// Feed a quality report into the adaptive quality controller.
///
/// Returns a new profile if a tier transition occurred.
#[allow(unused)]
pub fn observe_quality(
&mut self,
report: &wzp_proto::QualityReport,
) -> Option<QualityProfile> {
let new_profile = self.quality_ctrl.observe(report);
if let Some(ref profile) = new_profile {
if let Err(e) = self.encoder.set_profile(*profile) {
warn!("encoder set_profile error: {e}");
}
if let Err(e) = self.decoder.set_profile(*profile) {
warn!("decoder set_profile error: {e}");
}
}
new_profile
}
/// Force a specific quality profile.
#[allow(unused)]
pub fn force_profile(&mut self, profile: QualityProfile) {
self.quality_ctrl.force_profile(profile);
if let Err(e) = self.encoder.set_profile(profile) {
warn!("encoder set_profile error: {e}");
}
if let Err(e) = self.decoder.set_profile(profile) {
warn!("decoder set_profile error: {e}");
}
}
/// Get current pipeline statistics.
pub fn stats(&self) -> PipelineStats {
PipelineStats {
frames_encoded: self.frames_encoded,
frames_decoded: self.frames_decoded,
underruns: self.underruns,
jitter_depth: self.jitter_buffer.stats().current_depth,
quality_tier: self.quality_ctrl.tier() as u8,
}
}
}

View File

@@ -0,0 +1,42 @@
//! Call statistics for the Android engine.
/// State of the call.
#[derive(Clone, Debug, Default, serde::Serialize, PartialEq, Eq)]
pub enum CallState {
/// Engine is idle, no active call.
#[default]
Idle,
/// Establishing connection to the relay.
Connecting,
/// Call is active with audio flowing.
Active,
/// Temporarily lost connection, attempting to recover.
Reconnecting,
/// Call has ended.
Closed,
}
/// Aggregated call statistics, serializable for JNI bridge.
#[derive(Clone, Debug, Default, serde::Serialize)]
pub struct CallStats {
/// Current call state.
pub state: CallState,
/// Call duration in seconds.
pub duration_secs: f64,
/// Current quality tier (0=GOOD, 1=DEGRADED, 2=CATASTROPHIC).
pub quality_tier: u8,
/// Observed packet loss percentage.
pub loss_pct: f32,
/// Smoothed round-trip time in milliseconds.
pub rtt_ms: u32,
/// Jitter in milliseconds.
pub jitter_ms: u32,
/// Current jitter buffer depth in packets.
pub jitter_buffer_depth: usize,
/// Total frames encoded since call start.
pub frames_encoded: u64,
/// Total frames decoded since call start.
pub frames_decoded: u64,
/// Number of playout underruns (buffer empty when audio needed).
pub underruns: u64,
}

View File

@@ -115,6 +115,7 @@ pub fn signal_to_call_type(signal: &SignalMessage) -> CallSignalType {
#[cfg(test)]
mod tests {
use super::*;
use wzp_proto::QualityProfile;
#[test]
fn payload_roundtrip() {

View File

@@ -14,7 +14,7 @@ use crate::codec2_dec::Codec2Decoder;
use crate::codec2_enc::Codec2Encoder;
use crate::opus_dec::OpusDecoder;
use crate::opus_enc::OpusEncoder;
use crate::resample;
use crate::resample::{Downsampler48to8, Upsampler8to48};
// ─── Helpers ─────────────────────────────────────────────────────────────────
@@ -54,6 +54,7 @@ pub struct AdaptiveEncoder {
opus: OpusEncoder,
codec2: Codec2Encoder,
active: CodecId,
downsampler: Downsampler48to8,
}
impl AdaptiveEncoder {
@@ -66,6 +67,7 @@ impl AdaptiveEncoder {
opus,
codec2,
active: profile.codec,
downsampler: Downsampler48to8::new(),
})
}
}
@@ -74,7 +76,7 @@ impl AudioEncoder for AdaptiveEncoder {
fn encode(&mut self, pcm: &[i16], out: &mut [u8]) -> Result<usize, CodecError> {
if is_codec2(self.active) {
// Downsample 48 kHz → 8 kHz then encode via Codec2.
let pcm_8k = resample::resample_48k_to_8k(pcm);
let pcm_8k = self.downsampler.process(pcm);
self.codec2.encode(&pcm_8k, out)
} else {
self.opus.encode(pcm, out)
@@ -126,6 +128,7 @@ pub struct AdaptiveDecoder {
opus: OpusDecoder,
codec2: Codec2Decoder,
active: CodecId,
upsampler: Upsampler8to48,
}
impl AdaptiveDecoder {
@@ -138,6 +141,7 @@ impl AdaptiveDecoder {
opus,
codec2,
active: profile.codec,
upsampler: Upsampler8to48::new(),
})
}
}
@@ -149,7 +153,7 @@ impl AudioDecoder for AdaptiveDecoder {
let c2_samples = self.codec2_frame_samples();
let mut buf_8k = vec![0i16; c2_samples];
let n = self.codec2.decode(encoded, &mut buf_8k)?;
let pcm_48k = resample::resample_8k_to_48k(&buf_8k[..n]);
let pcm_48k = self.upsampler.process(&buf_8k[..n]);
let out_len = pcm_48k.len().min(pcm.len());
pcm[..out_len].copy_from_slice(&pcm_48k[..out_len]);
Ok(out_len)
@@ -163,7 +167,7 @@ impl AudioDecoder for AdaptiveDecoder {
let c2_samples = self.codec2_frame_samples();
let mut buf_8k = vec![0i16; c2_samples];
let n = self.codec2.decode_lost(&mut buf_8k)?;
let pcm_48k = resample::resample_8k_to_48k(&buf_8k[..n]);
let pcm_48k = self.upsampler.process(&buf_8k[..n]);
let out_len = pcm_48k.len().min(pcm.len());
pcm[..out_len].copy_from_slice(&pcm_48k[..out_len]);
Ok(out_len)

228
crates/wzp-codec/src/aec.rs Normal file
View File

@@ -0,0 +1,228 @@
//! Acoustic Echo Cancellation using NLMS adaptive filter.
//! Processes 480-sample (10ms) sub-frames at 48kHz.
/// NLMS (Normalized Least Mean Squares) adaptive filter echo canceller.
///
/// Removes acoustic echo by modelling the echo path between the far-end
/// (speaker) signal and the near-end (microphone) signal, then subtracting
/// the estimated echo from the near-end in real time.
pub struct EchoCanceller {
filter_coeffs: Vec<f32>,
filter_len: usize,
far_end_buf: Vec<f32>,
far_end_pos: usize,
mu: f32,
enabled: bool,
}
impl EchoCanceller {
/// Create a new echo canceller.
///
/// * `sample_rate` — typically 48000
/// * `filter_ms` — echo-tail length in milliseconds (e.g. 100 for 100 ms)
pub fn new(sample_rate: u32, filter_ms: u32) -> Self {
let filter_len = (sample_rate as usize) * (filter_ms as usize) / 1000;
Self {
filter_coeffs: vec![0.0f32; filter_len],
filter_len,
far_end_buf: vec![0.0f32; filter_len],
far_end_pos: 0,
mu: 0.01,
enabled: true,
}
}
/// Feed far-end (speaker/playback) samples into the circular buffer.
///
/// Must be called with the audio that was played out through the speaker
/// *before* the corresponding near-end frame is processed.
pub fn feed_farend(&mut self, farend: &[i16]) {
for &s in farend {
self.far_end_buf[self.far_end_pos] = s as f32;
self.far_end_pos = (self.far_end_pos + 1) % self.filter_len;
}
}
/// Process a near-end (microphone) frame, removing the estimated echo.
///
/// Returns the echo-return-loss enhancement (ERLE) as a ratio: the RMS of
/// the original near-end divided by the RMS of the residual. Values > 1.0
/// mean echo was reduced.
pub fn process_frame(&mut self, nearend: &mut [i16]) -> f32 {
if !self.enabled {
return 1.0;
}
let n = nearend.len();
let fl = self.filter_len;
let mut sum_near_sq: f64 = 0.0;
let mut sum_err_sq: f64 = 0.0;
for i in 0..n {
let near_f = nearend[i] as f32;
// --- estimate echo as dot(coeffs, farend_window) ---
// The far-end window for this sample starts at
// (far_end_pos - 1 - i) mod filter_len (most recent)
// and goes back filter_len samples.
let mut echo_est: f32 = 0.0;
let mut power: f32 = 0.0;
// Position of the most-recent far-end sample for this near-end sample.
// far_end_pos points to the *next write* position, so the most-recent
// sample written is at far_end_pos - 1. We have already called
// feed_farend for this block, so the relevant samples are the last
// filter_len entries ending just before the current write position,
// offset by how far we are into this near-end frame.
//
// For sample i of the near-end frame, the corresponding far-end
// "now" is far_end_pos - n + i (wrapping).
// far_end_pos points to next-write, so most recent sample is at
// far_end_pos - 1. For the i-th near-end sample we want the
// far-end "now" to be at (far_end_pos - n + i). We add fl
// repeatedly to avoid underflow on the usize subtraction.
let base = (self.far_end_pos + fl * ((n / fl) + 2) + i - n) % fl;
for k in 0..fl {
let fe_idx = (base + fl - k) % fl;
let fe = self.far_end_buf[fe_idx];
echo_est += self.filter_coeffs[k] * fe;
power += fe * fe;
}
let error = near_f - echo_est;
// --- NLMS coefficient update ---
let norm = power + 1.0; // +1 regularisation to avoid div-by-zero
let step = self.mu * error / norm;
for k in 0..fl {
let fe_idx = (base + fl - k) % fl;
let fe = self.far_end_buf[fe_idx];
self.filter_coeffs[k] += step * fe;
}
// Clamp output
let out = error.max(-32768.0).min(32767.0);
nearend[i] = out as i16;
sum_near_sq += (near_f as f64) * (near_f as f64);
sum_err_sq += (out as f64) * (out as f64);
}
// ERLE ratio
if sum_err_sq < 1.0 {
return 100.0; // near-perfect cancellation
}
(sum_near_sq / sum_err_sq).sqrt() as f32
}
/// Enable or disable echo cancellation.
pub fn set_enabled(&mut self, enabled: bool) {
self.enabled = enabled;
}
/// Returns whether echo cancellation is currently enabled.
pub fn is_enabled(&self) -> bool {
self.enabled
}
/// Reset the adaptive filter to its initial state.
///
/// Zeroes out all filter coefficients and the far-end circular buffer.
pub fn reset(&mut self) {
self.filter_coeffs.iter_mut().for_each(|c| *c = 0.0);
self.far_end_buf.iter_mut().for_each(|s| *s = 0.0);
self.far_end_pos = 0;
}
}
#[cfg(test)]
mod tests {
use super::*;
#[test]
fn aec_creates_with_correct_filter_len() {
let aec = EchoCanceller::new(48000, 100);
assert_eq!(aec.filter_len, 4800);
assert_eq!(aec.filter_coeffs.len(), 4800);
assert_eq!(aec.far_end_buf.len(), 4800);
}
#[test]
fn aec_passthrough_when_disabled() {
let mut aec = EchoCanceller::new(48000, 100);
aec.set_enabled(false);
assert!(!aec.is_enabled());
let original: Vec<i16> = (0..480).map(|i| (i * 10) as i16).collect();
let mut frame = original.clone();
let erle = aec.process_frame(&mut frame);
assert_eq!(erle, 1.0);
assert_eq!(frame, original);
}
#[test]
fn aec_reset_zeroes_state() {
let mut aec = EchoCanceller::new(48000, 10); // short for test speed
let farend: Vec<i16> = (0..480).map(|i| ((i * 37) % 1000) as i16).collect();
aec.feed_farend(&farend);
aec.reset();
assert!(aec.filter_coeffs.iter().all(|&c| c == 0.0));
assert!(aec.far_end_buf.iter().all(|&s| s == 0.0));
assert_eq!(aec.far_end_pos, 0);
}
#[test]
fn aec_reduces_echo_of_known_signal() {
// Use a small filter for speed. Feed a known far-end signal, then
// present the *same* signal as near-end (perfect echo, no room).
// After adaptation the output energy should drop.
let filter_ms = 5; // 240 taps at 48 kHz
let mut aec = EchoCanceller::new(48000, filter_ms);
// Generate a simple repeating pattern.
let frame_len = 480usize;
let make_frame = |offset: usize| -> Vec<i16> {
(0..frame_len)
.map(|i| {
let t = (offset + i) as f64 / 48000.0;
(5000.0 * (2.0 * std::f64::consts::PI * 300.0 * t).sin()) as i16
})
.collect()
};
// Warm up the adaptive filter with several frames.
let mut last_erle = 1.0f32;
for frame_idx in 0..40 {
let farend = make_frame(frame_idx * frame_len);
aec.feed_farend(&farend);
// Near-end = exact copy of far-end (pure echo).
let mut nearend = farend.clone();
last_erle = aec.process_frame(&mut nearend);
}
// After 40 frames the ERLE should be meaningfully > 1.
assert!(
last_erle > 1.0,
"expected ERLE > 1.0 after adaptation, got {last_erle}"
);
}
#[test]
fn aec_silence_passthrough() {
let mut aec = EchoCanceller::new(48000, 10);
// Feed silence far-end
aec.feed_farend(&vec![0i16; 480]);
// Near-end is silence too
let mut frame = vec![0i16; 480];
let erle = aec.process_frame(&mut frame);
assert!(erle >= 1.0);
// Output should still be silence
assert!(frame.iter().all(|&s| s == 0));
}
}

219
crates/wzp-codec/src/agc.rs Normal file
View File

@@ -0,0 +1,219 @@
//! Automatic Gain Control (AGC) with two-stage smoothing.
//!
//! Uses a fast attack / slow release envelope follower to keep the
//! output signal near a configurable target RMS level. This prevents
//! both clipping (when the speaker is too loud) and inaudibility (when
//! the speaker is too quiet or far from the mic).
/// Two-stage automatic gain control.
///
/// The gain is adjusted per-frame based on the measured RMS energy,
/// with a fast attack (gain decreases quickly when signal gets louder)
/// and a slow release (gain increases gradually when signal gets quieter).
pub struct AutoGainControl {
target_rms: f64,
current_gain: f64,
min_gain: f64,
max_gain: f64,
attack_alpha: f64,
release_alpha: f64,
enabled: bool,
}
impl AutoGainControl {
/// Create a new AGC with sensible VoIP defaults.
pub fn new() -> Self {
Self {
target_rms: 3000.0, // ~-20 dBFS for i16
current_gain: 1.0,
min_gain: 0.5,
max_gain: 32.0,
attack_alpha: 0.3, // fast attack
release_alpha: 0.02, // slow release
enabled: true,
}
}
/// Process a frame of PCM audio in-place, applying gain adjustment.
pub fn process_frame(&mut self, pcm: &mut [i16]) {
if !self.enabled {
return;
}
// Compute RMS of the frame.
let rms = Self::compute_rms(pcm);
// Don't amplify near-silence — it would just boost noise.
if rms < 10.0 {
return;
}
// Desired instantaneous gain.
let desired_gain = (self.target_rms / rms).clamp(self.min_gain, self.max_gain);
// Smooth the gain transition.
let alpha = if desired_gain < self.current_gain {
// Signal is louder than target → reduce gain quickly (attack).
self.attack_alpha
} else {
// Signal is quieter than target → raise gain slowly (release).
self.release_alpha
};
self.current_gain = self.current_gain * (1.0 - alpha) + desired_gain * alpha;
// Apply gain to each sample with hard limiting at ±31000 (~0.946 * i16::MAX).
const LIMIT: f64 = 31000.0;
let gain = self.current_gain;
for sample in pcm.iter_mut() {
let amplified = (*sample as f64) * gain;
let clamped = amplified.clamp(-LIMIT, LIMIT);
*sample = clamped as i16;
}
}
/// Enable or disable the AGC.
pub fn set_enabled(&mut self, enabled: bool) {
self.enabled = enabled;
}
/// Returns whether the AGC is currently enabled.
pub fn is_enabled(&self) -> bool {
self.enabled
}
/// Current gain expressed in dB.
pub fn current_gain_db(&self) -> f64 {
20.0 * self.current_gain.log10()
}
/// Compute the RMS (root mean square) of a PCM buffer.
fn compute_rms(pcm: &[i16]) -> f64 {
if pcm.is_empty() {
return 0.0;
}
let sum_sq: f64 = pcm.iter().map(|&s| (s as f64) * (s as f64)).sum();
(sum_sq / pcm.len() as f64).sqrt()
}
}
impl Default for AutoGainControl {
fn default() -> Self {
Self::new()
}
}
#[cfg(test)]
mod tests {
use super::*;
#[test]
fn agc_creates_with_defaults() {
let agc = AutoGainControl::new();
assert!(agc.is_enabled());
assert!((agc.current_gain - 1.0).abs() < f64::EPSILON);
}
#[test]
fn agc_passthrough_when_disabled() {
let mut agc = AutoGainControl::new();
agc.set_enabled(false);
let original: Vec<i16> = (0..960).map(|i| (i * 5) as i16).collect();
let mut frame = original.clone();
agc.process_frame(&mut frame);
assert_eq!(frame, original);
}
#[test]
fn agc_does_not_amplify_silence() {
let mut agc = AutoGainControl::new();
let mut frame = vec![0i16; 960];
agc.process_frame(&mut frame);
assert!(frame.iter().all(|&s| s == 0));
// Gain should remain at initial value.
assert!((agc.current_gain - 1.0).abs() < f64::EPSILON);
}
#[test]
fn agc_amplifies_quiet_signal() {
let mut agc = AutoGainControl::new();
// Very quiet signal (RMS ~ 50).
let mut frame: Vec<i16> = (0..960)
.map(|i| {
let t = i as f64 / 48000.0;
(50.0 * (2.0 * std::f64::consts::PI * 440.0 * t).sin()) as i16
})
.collect();
// Process several frames to let the gain ramp up.
for _ in 0..50 {
let mut f = frame.clone();
agc.process_frame(&mut f);
frame = f;
}
// Gain should have increased past 1.0.
assert!(
agc.current_gain > 1.05,
"expected gain > 1.05 for quiet signal, got {}",
agc.current_gain
);
}
#[test]
fn agc_attenuates_loud_signal() {
let mut agc = AutoGainControl::new();
// Loud signal (RMS ~ 20000).
let frame: Vec<i16> = (0..960)
.map(|i| {
let t = i as f64 / 48000.0;
(28000.0 * (2.0 * std::f64::consts::PI * 440.0 * t).sin()) as i16
})
.collect();
// Process several frames.
for _ in 0..20 {
let mut f = frame.clone();
agc.process_frame(&mut f);
}
// Gain should have decreased below 1.0.
assert!(
agc.current_gain < 1.0,
"expected gain < 1.0 for loud signal, got {}",
agc.current_gain
);
}
#[test]
fn agc_output_within_limits() {
let mut agc = AutoGainControl::new();
// Force a high gain by processing many quiet frames first.
for _ in 0..100 {
let mut f: Vec<i16> = vec![100; 960];
agc.process_frame(&mut f);
}
// Now send a louder frame — output should still be within ±31000.
let mut frame: Vec<i16> = vec![20000; 960];
agc.process_frame(&mut frame);
assert!(
frame.iter().all(|&s| s.abs() <= 31000),
"output samples must be within ±31000"
);
}
#[test]
fn agc_gain_db_at_unity() {
let agc = AutoGainControl::new();
let db = agc.current_gain_db();
assert!(
db.abs() < 0.01,
"expected ~0 dB at unity gain, got {db}"
);
}
}

View File

@@ -10,6 +10,8 @@
//! trait-object encoders/decoders that handle adaptive switching internally.
pub mod adaptive;
pub mod aec;
pub mod agc;
pub mod codec2_dec;
pub mod codec2_enc;
pub mod denoise;
@@ -19,6 +21,8 @@ pub mod resample;
pub mod silence;
pub use adaptive::{AdaptiveDecoder, AdaptiveEncoder};
pub use aec::EchoCanceller;
pub use agc::AutoGainControl;
pub use denoise::NoiseSupressor;
pub use silence::{ComfortNoise, SilenceDetector};
pub use wzp_proto::{AudioDecoder, AudioEncoder, CodecId, QualityProfile};

View File

@@ -40,6 +40,11 @@ impl OpusEncoder {
.set_signal(Signal::Voice)
.map_err(|e| CodecError::EncodeFailed(format!("set signal: {e}")))?;
// Default complexity 7 — good quality/CPU trade-off for VoIP
enc.inner
.set_complexity(7)
.map_err(|e| CodecError::EncodeFailed(format!("set complexity: {e}")))?;
Ok(enc)
}
@@ -56,6 +61,21 @@ impl OpusEncoder {
pub fn frame_samples(&self) -> usize {
(48_000 * self.frame_duration_ms as usize) / 1000
}
/// Set the encoder complexity (0-10). Higher values produce better quality
/// at the cost of more CPU. Default is 7.
pub fn set_complexity(&mut self, complexity: i32) {
let c = (complexity as u8).min(10);
let _ = self.inner.set_complexity(c);
}
/// Hint the encoder about expected packet loss percentage (0-100).
///
/// Higher values cause the encoder to use more redundancy to survive
/// packet loss, at the expense of slightly higher bitrate.
pub fn set_expected_loss(&mut self, loss_pct: u8) {
let _ = self.inner.set_packet_loss_perc(loss_pct.min(100));
}
}
impl AudioEncoder for OpusEncoder {

View File

@@ -1,54 +1,257 @@
//! Simple linear resampler for 48 kHz <-> 8 kHz conversion.
//! Windowed-sinc FIR resampler for 48 kHz <-> 8 kHz conversion.
//!
//! These are basic implementations suitable for voice. For higher quality,
//! replace with the `rubato` crate later.
//! Provides both stateless free functions (backward-compatible) and stateful
//! `Downsampler48to8` / `Upsampler8to48` structs that maintain overlap history
//! between frames for glitch-free streaming.
/// Downsample from 48 kHz to 8 kHz (6:1 decimation with averaging).
///
/// Each output sample is the average of 6 consecutive input samples,
/// providing basic anti-aliasing via a box filter.
pub fn resample_48k_to_8k(input: &[i16]) -> Vec<i16> {
use std::f64::consts::PI;
// ─── FIR kernel parameters ─────────────────────────────────────────────────
/// Number of FIR taps in the anti-alias / interpolation filter.
const FIR_TAPS: usize = 48;
/// Kaiser window beta parameter — controls sidelobe attenuation.
const KAISER_BETA: f64 = 8.0;
/// Cutoff frequency in Hz for the low-pass filter (just below 4 kHz Nyquist of 8 kHz).
const CUTOFF_HZ: f64 = 3800.0;
/// Working sample rate in Hz.
const SAMPLE_RATE: f64 = 48000.0;
/// Decimation / interpolation ratio between 48 kHz and 8 kHz.
const RATIO: usize = 6;
// ─── Kaiser window helpers ─────────────────────────────────────────────────
/// Zeroth-order modified Bessel function of the first kind, I₀(x).
///
/// Computed via the well-known power-series expansion, converging rapidly
/// for the moderate values of x used in Kaiser window design.
fn bessel_i0(x: f64) -> f64 {
let mut sum = 1.0f64;
let mut term = 1.0f64;
let half_x = x / 2.0;
for k in 1..=25 {
term *= (half_x / k as f64) * (half_x / k as f64);
sum += term;
if term < 1e-12 * sum {
break;
}
}
sum
}
/// Build a windowed-sinc low-pass FIR kernel.
///
/// Returns `FIR_TAPS` coefficients normalised so that the DC gain is exactly 1.0.
fn build_fir_kernel() -> [f64; FIR_TAPS] {
let mut kernel = [0.0f64; FIR_TAPS];
let m = (FIR_TAPS - 1) as f64;
let fc = CUTOFF_HZ / SAMPLE_RATE; // normalised cutoff (0..0.5)
let beta_denom = bessel_i0(KAISER_BETA);
for i in 0..FIR_TAPS {
// Sinc
let n = i as f64 - m / 2.0;
let sinc = if n.abs() < 1e-12 {
2.0 * fc
} else {
(2.0 * PI * fc * n).sin() / (PI * n)
};
// Kaiser window
let t = 2.0 * i as f64 / m - 1.0; // range [-1, 1]
let kaiser = bessel_i0(KAISER_BETA * (1.0 - t * t).max(0.0).sqrt()) / beta_denom;
kernel[i] = sinc * kaiser;
}
// Normalise to unity DC gain.
let sum: f64 = kernel.iter().sum();
if sum.abs() > 1e-15 {
for k in kernel.iter_mut() {
*k /= sum;
}
}
kernel
}
// ─── Stateful Downsampler 48→8 ─────────────────────────────────────────────
/// Stateful polyphase FIR downsampler from 48 kHz to 8 kHz.
///
/// Maintains `FIR_TAPS - 1` samples of history between successive calls to
/// `process()` for seamless frame boundaries.
pub struct Downsampler48to8 {
kernel: [f64; FIR_TAPS],
history: Vec<f64>,
}
impl Downsampler48to8 {
pub fn new() -> Self {
Self {
kernel: build_fir_kernel(),
history: vec![0.0; FIR_TAPS - 1],
}
}
/// Downsample a block of 48 kHz samples to 8 kHz.
///
/// The input length should be a multiple of 6; any trailing samples that
/// don't form a complete output sample are consumed into the history.
pub fn process(&mut self, input: &[i16]) -> Vec<i16> {
let hist_len = self.history.len(); // FIR_TAPS - 1
let total_len = hist_len + input.len();
// Build a working buffer: history ++ input (as f64).
let mut work = Vec::with_capacity(total_len);
work.extend_from_slice(&self.history);
work.extend(input.iter().map(|&s| s as f64));
let out_len = input.len() / RATIO;
let mut output = Vec::with_capacity(out_len);
for chunk in input.chunks_exact(RATIO) {
let sum: i32 = chunk.iter().map(|&s| s as i32).sum();
output.push((sum / RATIO as i32) as i16);
for i in 0..out_len {
// The centre of the filter for output sample i sits at
// position hist_len + i*RATIO in the work buffer (aligning
// with the first new input sample at decimation phase 0).
let centre = hist_len + i * RATIO;
let start = centre + 1 - FIR_TAPS; // may be 0 for the first few
let mut acc = 0.0f64;
for k in 0..FIR_TAPS {
let idx = start + k;
if idx < work.len() {
acc += work[idx] * self.kernel[k];
}
}
output.push(acc.round().clamp(-32768.0, 32767.0) as i16);
}
// Update history: keep the last (FIR_TAPS - 1) samples from work.
if work.len() >= hist_len {
self.history
.copy_from_slice(&work[work.len() - hist_len..]);
} else {
// Input was shorter than history — shift.
let shift = hist_len - work.len();
self.history.copy_within(shift.., 0);
for (i, &v) in work.iter().enumerate() {
self.history[hist_len - work.len() + i] = v;
}
}
output
}
/// Upsample from 8 kHz to 48 kHz (1:6 interpolation with linear interp).
///
/// Linearly interpolates between each pair of input samples to produce
/// 6 output samples per input sample.
pub fn resample_8k_to_48k(input: &[i16]) -> Vec<i16> {
const RATIO: usize = 6;
if input.is_empty() {
return Vec::new();
}
let out_len = input.len() * RATIO;
impl Default for Downsampler48to8 {
fn default() -> Self {
Self::new()
}
}
// ─── Stateful Upsampler 8→48 ───────────────────────────────────────────────
/// Stateful FIR upsampler from 8 kHz to 48 kHz.
///
/// Inserts zeros between input samples (zero-stuffing), then applies the
/// low-pass FIR to remove imaging, with gain compensation of `RATIO`.
pub struct Upsampler8to48 {
kernel: [f64; FIR_TAPS],
history: Vec<f64>,
}
impl Upsampler8to48 {
pub fn new() -> Self {
Self {
kernel: build_fir_kernel(),
history: vec![0.0; FIR_TAPS - 1],
}
}
/// Upsample a block of 8 kHz samples to 48 kHz.
pub fn process(&mut self, input: &[i16]) -> Vec<i16> {
let hist_len = self.history.len(); // FIR_TAPS - 1
// Zero-stuff: insert RATIO-1 zeros between each input sample.
let stuffed_len = input.len() * RATIO;
let total_len = hist_len + stuffed_len;
let mut work = Vec::with_capacity(total_len);
work.extend_from_slice(&self.history);
for &s in input {
work.push(s as f64);
for _ in 1..RATIO {
work.push(0.0);
}
}
let out_len = stuffed_len;
let mut output = Vec::with_capacity(out_len);
for i in 0..input.len() {
let current = input[i] as i32;
let next = if i + 1 < input.len() {
input[i + 1] as i32
} else {
current // hold last sample
};
// The gain factor compensates for the zeros introduced by stuffing.
let gain = RATIO as f64;
for j in 0..RATIO {
let interp = current + (next - current) * j as i32 / RATIO as i32;
output.push(interp as i16);
for i in 0..out_len {
let centre = hist_len + i;
let start = centre + 1 - FIR_TAPS;
let mut acc = 0.0f64;
for k in 0..FIR_TAPS {
let idx = start + k;
if idx < work.len() {
acc += work[idx] * self.kernel[k];
}
}
acc *= gain;
output.push(acc.round().clamp(-32768.0, 32767.0) as i16);
}
// Update history.
if work.len() >= hist_len {
self.history
.copy_from_slice(&work[work.len() - hist_len..]);
} else {
let shift = hist_len - work.len();
self.history.copy_within(shift.., 0);
for (i, &v) in work.iter().enumerate() {
self.history[hist_len - work.len() + i] = v;
}
}
output
}
}
impl Default for Upsampler8to48 {
fn default() -> Self {
Self::new()
}
}
// ─── Backward-compatible free functions ─────────────────────────────────────
/// Downsample from 48 kHz to 8 kHz (6:1 decimation with FIR anti-alias filter).
///
/// This is a convenience wrapper that creates a temporary [`Downsampler48to8`].
/// For streaming use, prefer the stateful struct to avoid edge artefacts between
/// frames.
pub fn resample_48k_to_8k(input: &[i16]) -> Vec<i16> {
let mut ds = Downsampler48to8::new();
ds.process(input)
}
/// Upsample from 8 kHz to 48 kHz (1:6 interpolation with FIR imaging filter).
///
/// This is a convenience wrapper that creates a temporary [`Upsampler8to48`].
/// For streaming use, prefer the stateful struct to avoid edge artefacts between
/// frames.
pub fn resample_8k_to_48k(input: &[i16]) -> Vec<i16> {
let mut us = Upsampler8to48::new();
us.process(input)
}
// ─── Tests ──────────────────────────────────────────────────────────────────
#[cfg(test)]
mod tests {
@@ -66,12 +269,28 @@ mod tests {
#[test]
fn dc_signal_preserved() {
// A constant signal should survive resampling
// A constant signal should survive resampling (approximately).
let input = vec![1000i16; 960];
let down = resample_48k_to_8k(&input);
assert!(down.iter().all(|&s| s == 1000));
// Allow some edge transient — check that the middle samples are close.
let mid_start = down.len() / 4;
let mid_end = 3 * down.len() / 4;
for &s in &down[mid_start..mid_end] {
assert!(
(s - 1000).abs() < 50,
"DC downsampled sample {s} too far from 1000"
);
}
let up = resample_8k_to_48k(&down);
assert!(up.iter().all(|&s| s == 1000));
let mid_start_up = up.len() / 4;
let mid_end_up = 3 * up.len() / 4;
for &s in &up[mid_start_up..mid_end_up] {
assert!(
(s - 1000).abs() < 100,
"DC upsampled sample {s} too far from 1000"
);
}
}
#[test]
@@ -79,4 +298,40 @@ mod tests {
assert!(resample_48k_to_8k(&[]).is_empty());
assert!(resample_8k_to_48k(&[]).is_empty());
}
#[test]
fn stateful_downsampler_produces_correct_length() {
let mut ds = Downsampler48to8::new();
let out = ds.process(&vec![0i16; 960]);
assert_eq!(out.len(), 160);
let out2 = ds.process(&vec![0i16; 960]);
assert_eq!(out2.len(), 160);
}
#[test]
fn stateful_upsampler_produces_correct_length() {
let mut us = Upsampler8to48::new();
let out = us.process(&vec![0i16; 160]);
assert_eq!(out.len(), 960);
let out2 = us.process(&vec![0i16; 160]);
assert_eq!(out2.len(), 960);
}
#[test]
fn fir_kernel_has_unity_dc_gain() {
let kernel = build_fir_kernel();
let sum: f64 = kernel.iter().sum();
assert!(
(sum - 1.0).abs() < 1e-10,
"FIR kernel DC gain should be 1.0, got {sum}"
);
}
#[test]
fn bessel_i0_known_values() {
// I₀(0) = 1
assert!((bessel_i0(0.0) - 1.0).abs() < 1e-12);
// I₀(1) ≈ 1.2660658
assert!((bessel_i0(1.0) - 1.2660658).abs() < 1e-5);
}
}

View File

@@ -1,4 +1,5 @@
use std::collections::BTreeMap;
use std::time::{Duration, Instant};
use crate::packet::MediaPacket;
@@ -20,19 +21,29 @@ pub struct AdaptivePlayoutDelay {
max_delay: usize,
/// Exponential moving average of inter-packet arrival jitter (ms).
jitter_ema: f64,
/// EMA smoothing factor (0.0-1.0, lower = smoother).
alpha: f64,
/// EMA smoothing factor for jitter increases (fast reaction).
alpha_up: f64,
/// EMA smoothing factor for jitter decreases (slow decay).
alpha_down: f64,
/// Last packet arrival timestamp (for computing inter-arrival jitter).
last_arrival_ms: Option<u64>,
/// Last packet expected timestamp.
last_expected_ms: Option<u64>,
/// Safety margin added to jitter-derived target (in packets).
safety_margin: f64,
/// Instant when a jitter spike was detected (handoff detection).
spike_detected_at: Option<Instant>,
/// Duration to hold max_delay after a spike is detected.
spike_cooldown: Duration,
/// Multiplier of jitter_ema that constitutes a spike.
spike_threshold_multiplier: f64,
}
/// Frame duration in milliseconds (20ms Opus/Codec2 frames).
const FRAME_DURATION_MS: f64 = 20.0;
/// Safety margin added to jitter-derived target (in packets).
const SAFETY_MARGIN_PACKETS: f64 = 2.0;
/// Default EMA smoothing factor.
/// Default safety margin in packets.
const DEFAULT_SAFETY_MARGIN: f64 = 2.0;
/// Default EMA smoothing factor (used for both up/down in non-mobile mode).
const DEFAULT_ALPHA: f64 = 0.05;
impl AdaptivePlayoutDelay {
@@ -46,9 +57,14 @@ impl AdaptivePlayoutDelay {
min_delay,
max_delay,
jitter_ema: 0.0,
alpha: DEFAULT_ALPHA,
alpha_up: DEFAULT_ALPHA,
alpha_down: DEFAULT_ALPHA,
last_arrival_ms: None,
last_expected_ms: None,
safety_margin: DEFAULT_SAFETY_MARGIN,
spike_detected_at: None,
spike_cooldown: Duration::from_secs(2),
spike_threshold_multiplier: 3.0,
}
}
@@ -64,14 +80,39 @@ impl AdaptivePlayoutDelay {
let expected_delta = expected_ms as f64 - last_expected as f64;
let jitter = (actual_delta - expected_delta).abs();
// Update EMA
self.jitter_ema = self.alpha * jitter + (1.0 - self.alpha) * self.jitter_ema;
// Spike detection: check before EMA update
if self.jitter_ema > 0.0
&& jitter > self.jitter_ema * self.spike_threshold_multiplier
{
self.spike_detected_at = Some(Instant::now());
}
// Asymmetric EMA update
let alpha = if jitter > self.jitter_ema {
self.alpha_up
} else {
self.alpha_down
};
self.jitter_ema = alpha * jitter + (1.0 - alpha) * self.jitter_ema;
// Check if spike cooldown has expired
if let Some(spike_time) = self.spike_detected_at {
if spike_time.elapsed() >= self.spike_cooldown {
self.spike_detected_at = None;
}
}
// If within spike cooldown, return max_delay
if self.spike_detected_at.is_some() {
self.target_delay = self.max_delay;
} else {
// Convert jitter estimate to target delay in packets
let raw_target = (self.jitter_ema / FRAME_DURATION_MS).ceil() + SAFETY_MARGIN_PACKETS;
let raw_target =
(self.jitter_ema / FRAME_DURATION_MS).ceil() + self.safety_margin;
self.target_delay =
(raw_target as usize).clamp(self.min_delay, self.max_delay);
}
}
self.last_arrival_ms = Some(arrival_ms);
self.last_expected_ms = Some(expected_ms);
@@ -87,6 +128,28 @@ impl AdaptivePlayoutDelay {
pub fn jitter_estimate_ms(&self) -> f64 {
self.jitter_ema
}
/// Enable or disable mobile mode, adjusting parameters for cellular networks.
///
/// Mobile mode uses:
/// - Asymmetric alpha (fast up=0.3, slow down=0.02) for quicker spike detection
/// - Higher safety margin (3.0 packets) to absorb handoff jitter
/// - Spike detection with 2-second cooldown at 3x threshold
pub fn set_mobile_mode(&mut self, enabled: bool) {
if enabled {
self.safety_margin = 3.0;
self.alpha_up = 0.3;
self.alpha_down = 0.02;
self.spike_threshold_multiplier = 3.0;
self.spike_cooldown = Duration::from_secs(2);
} else {
self.safety_margin = DEFAULT_SAFETY_MARGIN;
self.alpha_up = DEFAULT_ALPHA;
self.alpha_down = DEFAULT_ALPHA;
self.spike_threshold_multiplier = 3.0;
self.spike_cooldown = Duration::from_secs(2);
}
}
}
// ---------------------------------------------------------------------------
@@ -391,6 +454,11 @@ impl JitterBuffer {
self.adaptive.as_ref()
}
/// Get a mutable reference to the adaptive playout delay estimator.
pub fn adaptive_delay_mut(&mut self) -> Option<&mut AdaptivePlayoutDelay> {
self.adaptive.as_mut()
}
/// Adjust target depth based on observed jitter.
pub fn set_target_depth(&mut self, depth: usize) {
self.target_depth = depth.min(self.max_depth);
@@ -720,4 +788,29 @@ mod tests {
let ad = jb.adaptive_delay().unwrap();
assert_eq!(ad.target_delay(), 3);
}
// ---------------------------------------------------------------
// Mobile mode tests
// ---------------------------------------------------------------
#[test]
fn mobile_mode_increases_safety_margin() {
let mut apd = AdaptivePlayoutDelay::new(3, 50);
apd.set_mobile_mode(true);
assert_eq!(apd.safety_margin, 3.0);
assert_eq!(apd.alpha_up, 0.3);
assert_eq!(apd.alpha_down, 0.02);
apd.set_mobile_mode(false);
assert_eq!(apd.safety_margin, DEFAULT_SAFETY_MARGIN);
assert_eq!(apd.alpha_up, DEFAULT_ALPHA);
assert_eq!(apd.alpha_down, DEFAULT_ALPHA);
}
#[test]
fn mobile_mode_accessible_via_jitter_buffer() {
let mut jb = JitterBuffer::new_adaptive(3, 50);
jb.adaptive_delay_mut().unwrap().set_mobile_mode(true);
assert_eq!(jb.adaptive_delay().unwrap().safety_margin, 3.0);
}
}

View File

@@ -29,6 +29,6 @@ pub use packet::{
SignalMessage, TrunkEntry, TrunkFrame, FRAME_TYPE_FULL, FRAME_TYPE_MINI,
};
pub use bandwidth::{BandwidthEstimator, CongestionState};
pub use quality::{AdaptiveQualityController, Tier};
pub use quality::{AdaptiveQualityController, NetworkContext, Tier};
pub use session::{Session, SessionEvent, SessionState};
pub use traits::*;

View File

@@ -1,4 +1,5 @@
use std::collections::VecDeque;
use std::time::{Duration, Instant};
use crate::packet::QualityReport;
use crate::traits::QualityController;
@@ -24,11 +25,31 @@ impl Tier {
}
}
/// Determine which tier a quality report belongs to.
/// Determine which tier a quality report belongs to (default/WiFi thresholds).
pub fn classify(report: &QualityReport) -> Self {
Self::classify_with_context(report, NetworkContext::Unknown)
}
/// Classify with network-context-aware thresholds.
pub fn classify_with_context(report: &QualityReport, context: NetworkContext) -> Self {
let loss = report.loss_percent();
let rtt = report.rtt_ms();
match context {
NetworkContext::CellularLte
| NetworkContext::Cellular5g
| NetworkContext::Cellular3g => {
// Tighter thresholds for cellular networks
if loss > 25.0 || rtt > 500 {
Self::Catastrophic
} else if loss > 8.0 || rtt > 300 {
Self::Degraded
} else {
Self::Good
}
}
NetworkContext::WiFi | NetworkContext::Unknown => {
// Original thresholds
if loss > 40.0 || rtt > 600 {
Self::Catastrophic
} else if loss > 10.0 || rtt > 400 {
@@ -38,10 +59,37 @@ impl Tier {
}
}
}
}
/// Return the next lower (worse) tier, or None if already at the worst.
pub fn downgrade(self) -> Option<Tier> {
match self {
Self::Good => Some(Self::Degraded),
Self::Degraded => Some(Self::Catastrophic),
Self::Catastrophic => None,
}
}
}
/// Describes the network transport type for context-aware quality decisions.
#[derive(Clone, Copy, Debug, PartialEq, Eq)]
pub enum NetworkContext {
WiFi,
CellularLte,
Cellular5g,
Cellular3g,
Unknown,
}
impl Default for NetworkContext {
fn default() -> Self {
Self::Unknown
}
}
/// Adaptive quality controller with hysteresis to prevent tier flapping.
///
/// - Downgrade: 3 consecutive reports in a worse tier
/// - Downgrade: 3 consecutive reports in a worse tier (2 on cellular)
/// - Upgrade: 10 consecutive reports in a better tier
pub struct AdaptiveQualityController {
current_tier: Tier,
@@ -54,14 +102,26 @@ pub struct AdaptiveQualityController {
history: VecDeque<QualityReport>,
/// Whether the profile was manually forced (disables adaptive logic).
forced: bool,
/// Current network context for threshold selection.
network_context: NetworkContext,
/// FEC boost expiry time (set during network handoff).
fec_boost_until: Option<Instant>,
/// FEC boost amount to add during handoff recovery window.
fec_boost_amount: f32,
}
/// Threshold for downgrading (fast reaction to degradation).
const DOWNGRADE_THRESHOLD: u32 = 3;
/// Threshold for downgrading on cellular networks (even faster).
const CELLULAR_DOWNGRADE_THRESHOLD: u32 = 2;
/// Threshold for upgrading (slow, cautious improvement).
const UPGRADE_THRESHOLD: u32 = 10;
/// Maximum history window size.
const HISTORY_SIZE: usize = 20;
/// Default FEC boost amount during handoff recovery.
const DEFAULT_FEC_BOOST: f32 = 0.2;
/// Duration of FEC boost after a network handoff.
const FEC_BOOST_DURATION_SECS: u64 = 10;
impl AdaptiveQualityController {
pub fn new() -> Self {
@@ -72,6 +132,9 @@ impl AdaptiveQualityController {
consecutive_down: 0,
history: VecDeque::with_capacity(HISTORY_SIZE),
forced: false,
network_context: NetworkContext::default(),
fec_boost_until: None,
fec_boost_amount: DEFAULT_FEC_BOOST,
}
}
@@ -80,6 +143,69 @@ impl AdaptiveQualityController {
self.current_tier
}
/// Get the current network context.
pub fn network_context(&self) -> NetworkContext {
self.network_context
}
/// Signal a network transport change (e.g., WiFi to cellular handoff).
///
/// When switching from WiFi to any cellular type, this preemptively
/// downgrades one quality tier and activates a temporary FEC boost.
pub fn signal_network_change(&mut self, new_context: NetworkContext) {
let old = self.network_context;
self.network_context = new_context;
let new_is_cellular = matches!(
new_context,
NetworkContext::CellularLte | NetworkContext::Cellular5g | NetworkContext::Cellular3g
);
// If switching from WiFi to cellular, preemptively downgrade one tier
if old == NetworkContext::WiFi && new_is_cellular {
if let Some(lower_tier) = self.current_tier.downgrade() {
self.current_tier = lower_tier;
self.current_profile = lower_tier.profile();
}
// Reset counters to avoid stale hysteresis state
self.consecutive_up = 0;
self.consecutive_down = 0;
// Un-force so adaptive logic resumes
self.forced = false;
}
// Activate FEC boost for any network change
self.fec_boost_until = Some(Instant::now() + Duration::from_secs(FEC_BOOST_DURATION_SECS));
}
/// Returns the FEC boost amount if within the handoff recovery window, 0.0 otherwise.
///
/// Callers should add this to their base FEC ratio during the boost window.
pub fn fec_boost(&self) -> f32 {
if let Some(until) = self.fec_boost_until {
if Instant::now() < until {
return self.fec_boost_amount;
}
}
0.0
}
/// Reset the hysteresis counters.
pub fn reset_counters(&mut self) {
self.consecutive_up = 0;
self.consecutive_down = 0;
}
/// Get the effective downgrade threshold based on network context.
fn downgrade_threshold(&self) -> u32 {
match self.network_context {
NetworkContext::CellularLte
| NetworkContext::Cellular5g
| NetworkContext::Cellular3g => CELLULAR_DOWNGRADE_THRESHOLD,
_ => DOWNGRADE_THRESHOLD,
}
}
fn try_transition(&mut self, observed_tier: Tier) -> Option<QualityProfile> {
if observed_tier == self.current_tier {
self.consecutive_up = 0;
@@ -96,7 +222,7 @@ impl AdaptiveQualityController {
if is_worse {
self.consecutive_up = 0;
self.consecutive_down += 1;
if self.consecutive_down >= DOWNGRADE_THRESHOLD {
if self.consecutive_down >= self.downgrade_threshold() {
self.current_tier = observed_tier;
self.current_profile = observed_tier.profile();
self.consecutive_down = 0;
@@ -142,7 +268,7 @@ impl QualityController for AdaptiveQualityController {
return None;
}
let observed = Tier::classify(report);
let observed = Tier::classify_with_context(report, self.network_context);
self.try_transition(observed)
}
@@ -246,4 +372,110 @@ mod tests {
assert_eq!(Tier::classify(&make_report(50.0, 200)), Tier::Catastrophic);
assert_eq!(Tier::classify(&make_report(5.0, 700)), Tier::Catastrophic);
}
// ---------------------------------------------------------------
// Network context tests
// ---------------------------------------------------------------
#[test]
fn cellular_tighter_thresholds() {
// 12% loss: Good on WiFi, Degraded on cellular
let report = make_report(12.0, 200);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::WiFi),
Tier::Degraded
);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::CellularLte),
Tier::Degraded
);
// 9% loss: Good on WiFi, Degraded on cellular
let report = make_report(9.0, 200);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::WiFi),
Tier::Good
);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::CellularLte),
Tier::Degraded
);
// 30% loss: Degraded on WiFi, Catastrophic on cellular
let report = make_report(30.0, 200);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::WiFi),
Tier::Degraded
);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::Cellular3g),
Tier::Catastrophic
);
}
#[test]
fn cellular_rtt_thresholds() {
// RTT 350ms: Good on WiFi, Degraded on cellular
let report = make_report(2.0, 348); // rtt_4ms rounds so use 348
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::WiFi),
Tier::Good
);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::CellularLte),
Tier::Degraded
);
}
#[test]
fn cellular_faster_downgrade() {
let mut ctrl = AdaptiveQualityController::new();
ctrl.signal_network_change(NetworkContext::CellularLte);
// Reset tier back to Good for testing downgrade threshold
ctrl.current_tier = Tier::Good;
ctrl.current_profile = Tier::Good.profile();
// On cellular, downgrade threshold is 2 instead of 3
let bad = make_report(50.0, 200);
assert!(ctrl.observe(&bad).is_none()); // 1st bad
let result = ctrl.observe(&bad); // 2nd bad — should trigger on cellular
assert!(result.is_some());
}
#[test]
fn signal_network_change_preemptive_downgrade() {
let mut ctrl = AdaptiveQualityController::new();
assert_eq!(ctrl.tier(), Tier::Good);
// Switch from WiFi to cellular
ctrl.network_context = NetworkContext::WiFi;
ctrl.signal_network_change(NetworkContext::CellularLte);
// Should have downgraded one tier: Good -> Degraded
assert_eq!(ctrl.tier(), Tier::Degraded);
}
#[test]
fn signal_network_change_fec_boost() {
let mut ctrl = AdaptiveQualityController::new();
assert_eq!(ctrl.fec_boost(), 0.0);
ctrl.signal_network_change(NetworkContext::CellularLte);
// FEC boost should be active
assert!(ctrl.fec_boost() > 0.0);
assert_eq!(ctrl.fec_boost(), DEFAULT_FEC_BOOST);
}
#[test]
fn tier_downgrade() {
assert_eq!(Tier::Good.downgrade(), Some(Tier::Degraded));
assert_eq!(Tier::Degraded.downgrade(), Some(Tier::Catastrophic));
assert_eq!(Tier::Catastrophic.downgrade(), None);
}
#[test]
fn network_context_default() {
assert_eq!(NetworkContext::default(), NetworkContext::Unknown);
}
}

View File

@@ -149,6 +149,27 @@ impl PathMonitor {
}
0
}
/// Detect whether a network handoff likely occurred.
///
/// Returns `true` if the most recent RTT jitter measurement exceeds 3x
/// the EWMA-smoothed jitter average, which is characteristic of a cellular
/// network handoff (tower switch, WiFi-to-cellular transition, etc.).
pub fn detect_handoff(&self) -> bool {
// We need at least two RTT observations to have a meaningful jitter value,
// and the EWMA must be non-zero to avoid division/multiplication by zero.
if self.jitter_ewma <= 0.0 {
return false;
}
if let (Some(last_rtt), Some(_)) = (self.last_rtt_ms, Some(self.rtt_ewma)) {
// Compute the most recent instantaneous jitter (RTT deviation from EWMA)
let instant_jitter = (last_rtt - self.rtt_ewma).abs();
instant_jitter > self.jitter_ewma * 3.0
} else {
false
}
}
}
impl Default for PathMonitor {