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Restore legacy Wi-Fi streaming stability #31

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4495711
test: align contracts with stream optimization
YashasVM Aug 5, 2026
65e3259
Restore legacy Wi-Fi streaming stability
YashasVM Aug 5, 2026
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18 changes: 11 additions & 7 deletions README.md
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Original file line number Diff line number Diff line change
Expand Up @@ -21,6 +21,10 @@
> [!IMPORTANT]
> OpenStream V1.0.1 is a fixed patch release for local Wi-Fi camera workflows. Device-specific camera behavior and network quality can still vary; please report bugs in [GitHub Issues](https://github.com/YashasVM/OpenStream/issues).

The supported V1.0.1 path is the legacy Android app plus the legacy OBS camera
source/dock over local Wi-Fi. USB transport and USB camera-source support are
not included in this release.

## Quick Downloads

| Step | Download | Install on |
Expand All @@ -38,7 +42,7 @@ Need the non-technical walkthrough with screenshots? Start with [`docs/set-up.md
OpenStream V1.0.1 sends your Android phone camera directly into OBS Studio over local Wi-Fi. It uses Camera2, MediaCodec video/audio encoding, MPEG-TS muxing, SRT transport, two-way LAN discovery, source-slot pairing, and a native OBS source plugin.

```text
Phone camera -> HEVC/H.264 + AAC -> SRT over Wi-Fi -> OpenStream camera slot in OBS
Phone camera -> hardware AVC/H.264 + AAC -> SRT over Wi-Fi -> OpenStream camera slot in OBS
```

### V1.0.1 Release
Expand Down Expand Up @@ -97,8 +101,8 @@ Use OBS as usual. Open **Docks → OpenStream Camera Control** for the camera co

| Feature | Details |
|---|---|
| **Full HD Streaming** | Streams a 1080p camera feed at up to 60 fps over SRT. |
| **Hardware Encoding** | Uses MediaCodec HEVC/H.265 with H.264 fallback. |
| **Full HD Streaming** | Streams a sustainable 1080p30 camera feed over SRT, with a 720p30 fallback profile. |
| **Hardware Encoding** | Uses an explicit hardware AVC/H.264 surface encoder; unsupported hardware is reported instead of silently using software video encoding. |
| **Audio Streaming** | Sends microphone audio with the video stream as AAC. |
| **Multi-Lens Switching** | Supports rear, ultrawide, telephoto, and front cameras when available. |
| **Pinch-to-Zoom** | Smooth digital zoom with a live zoom indicator. |
Expand Down Expand Up @@ -147,7 +151,7 @@ Use OBS as usual. Open **Docks → OpenStream Camera Control** for the camera co
```text
Android phone
Camera2 preview/capture
MediaCodec HEVC/H.264 video
MediaCodec hardware AVC/H.264 video
MediaCodec AAC audio
MPEG-TS muxer
libsrt sender
Expand All @@ -168,9 +172,9 @@ Discovery uses UDP port `51515`. Camera remote controls use the phone HTTP contr
| Parameter | Default | Notes |
|---|---|---|
| Resolution | `1920x1080` | Full HD capture target |
| Frame rate | `60 fps` | 30-60 fps depending on device support |
| Video codec | HEVC/H.265 | H.264 fallback |
| Bitrate | `50 Mbps` | High-quality local Wi-Fi default; lower if the link drops frames |
| Frame rate | `30 fps` | Bounded default to reduce heat and receiver backlog |
| Video codec | AVC/H.264 | Explicit hardware surface path |
| Bitrate | `12 Mbps` | Valid tuning range is `8-50 Mbps`; lower to 720p30 if a phone still runs hot |
| SRT latency | `120 ms` | 80-200 ms useful range |
| Discovery port | `51515/udp` | LAN discovery beacon |
| SRT port | `9000` | Media stream |
Expand Down
127 changes: 117 additions & 10 deletions android/app/src/main/cpp/openstream_srt.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -4,14 +4,18 @@

#include <algorithm>
#include <array>
#include <atomic>
#include <charconv>
#include <condition_variable>
#include <cstdlib>
#include <cstdint>
#include <cstring>
#include <deque>
#include <map>
#include <mutex>
#include <optional>
#include <string>
#include <thread>
#include <utility>
#include <vector>

Expand All @@ -27,6 +31,8 @@
namespace {

constexpr const char *kTag = "OpenStreamSRT";
constexpr int kMinSrtLatencyMs = 80;
constexpr int kMaxSrtLatencyMs = 200;
constexpr int kMediaCodecBufferFlagKeyFrame = 1;
constexpr int kMediaCodecBufferFlagCodecConfig = 2;
constexpr int kAudioSampleRate = 48000;
Expand Down Expand Up @@ -467,7 +473,7 @@ std::optional<SrtUrl> parseSrtUrl(const std::string &url) {
std::from_chars(value.data(), value.data() + value.size(), latency);
if (latencyResult.ec == std::errc{} &&
latencyResult.ptr == value.data() + value.size()) {
parsed.latencyMs = std::clamp(latency, 80, 200);
parsed.latencyMs = std::clamp(latency, kMinSrtLatencyMs, kMaxSrtLatencyMs);
}
}
if (amp == std::string::npos) {
Expand All @@ -482,6 +488,13 @@ std::optional<SrtUrl> parseSrtUrl(const std::string &url) {

class NativeSender {
public:
NativeSender() : sendWorker_(&NativeSender::runSendWorker, this) {}

~NativeSender() {
stopSendWorker();
disconnect();
}

bool connect(const std::string &url) {
#if OPENSTREAM_HAVE_LIBSRT
disconnect();
Expand All @@ -506,11 +519,19 @@ class NativeSender {
int yes = 1;
int transportType = SRTT_LIVE;
int payloadSize = 188 * 7;
int sendTimeoutMs = 500;
// Never let a congested Wi-Fi link stall MediaCodec callbacks. SRT's
// live late-packet drop recovers latency instead of growing it.
int sendTimeoutMs = 120;
int tooLatePacketDrop = 1;
int connectTimeoutMs = 2000;
int peerIdleTimeoutMs = 4000;
srt_setsockopt(socket, 0, SRTO_TRANSTYPE, &transportType, sizeof transportType);
srt_setsockopt(socket, 0, SRTO_SENDER, &yes, sizeof yes);
srt_setsockopt(socket, 0, SRTO_PAYLOADSIZE, &payloadSize, sizeof payloadSize);
srt_setsockopt(socket, 0, SRTO_SNDTIMEO, &sendTimeoutMs, sizeof sendTimeoutMs);
srt_setsockopt(socket, 0, SRTO_TLPKTDROP, &tooLatePacketDrop, sizeof tooLatePacketDrop);
srt_setsockopt(socket, 0, SRTO_CONNTIMEO, &connectTimeoutMs, sizeof connectTimeoutMs);
srt_setsockopt(socket, 0, SRTO_PEERIDLETIMEO, &peerIdleTimeoutMs, sizeof peerIdleTimeoutMs);
const int latency = parsed->latencyMs;
srt_setsockopt(socket, 0, SRTO_LATENCY, &latency, sizeof latency);
srt_setsockopt(socket, 0, SRTO_PEERLATENCY, &latency, sizeof latency);
Expand Down Expand Up @@ -572,13 +593,17 @@ class NativeSender {
int yes = 1;
int transportType = SRTT_LIVE;
int payloadSize = 188 * 7;
int sendTimeoutMs = 500;
int sendTimeoutMs = 120;
int tooLatePacketDrop = 1;
int peerIdleTimeoutMs = 4000;
const int latency = parsed->latencyMs;
srt_setsockopt(listenerSocket, 0, SRTO_TRANSTYPE, &transportType, sizeof transportType);
srt_setsockopt(listenerSocket, 0, SRTO_SENDER, &yes, sizeof yes);
srt_setsockopt(listenerSocket, 0, SRTO_REUSEADDR, &yes, sizeof yes);
srt_setsockopt(listenerSocket, 0, SRTO_PAYLOADSIZE, &payloadSize, sizeof payloadSize);
srt_setsockopt(listenerSocket, 0, SRTO_SNDTIMEO, &sendTimeoutMs, sizeof sendTimeoutMs);
srt_setsockopt(listenerSocket, 0, SRTO_TLPKTDROP, &tooLatePacketDrop, sizeof tooLatePacketDrop);
srt_setsockopt(listenerSocket, 0, SRTO_PEERIDLETIMEO, &peerIdleTimeoutMs, sizeof peerIdleTimeoutMs);
srt_setsockopt(listenerSocket, 0, SRTO_LATENCY, &latency, sizeof latency);
srt_setsockopt(listenerSocket, 0, SRTO_PEERLATENCY, &latency, sizeof latency);

Expand Down Expand Up @@ -609,6 +634,8 @@ class NativeSender {
return false;
}
srt_setsockopt(acceptedSocket, 0, SRTO_SNDTIMEO, &sendTimeoutMs, sizeof sendTimeoutMs);
srt_setsockopt(acceptedSocket, 0, SRTO_TLPKTDROP, &tooLatePacketDrop, sizeof tooLatePacketDrop);
srt_setsockopt(acceptedSocket, 0, SRTO_PEERIDLETIMEO, &peerIdleTimeoutMs, sizeof peerIdleTimeoutMs);
setSocket(acceptedSocket);
logInfo("OBS connected to Android SRT listener");
return true;
Expand All @@ -619,10 +646,10 @@ class NativeSender {
#endif
}

bool send(const std::vector<uint8_t> &bytes) {
bool sendNow(const std::vector<uint8_t> &bytes) {
#if OPENSTREAM_HAVE_LIBSRT
// Closing an SRT socket or tearing down libsrt while another codec thread
// is inside srt_sendmsg is unsafe. Serialize the whole send with teardown,
// Closing an SRT socket or tearing down libsrt while the send worker is
// inside srt_sendmsg is unsafe. Serialize the whole send with teardown,
// not just the socket-handle lookup.
std::lock_guard<std::mutex> ioLock(ioMutex_);
const SRTSOCKET socket = currentSocket();
Expand Down Expand Up @@ -655,8 +682,45 @@ class NativeSender {
#endif
}

bool send(std::vector<uint8_t> bytes) {
#if OPENSTREAM_HAVE_LIBSRT
if (!healthy_.load()) return false;
const size_t byteCount = bytes.size();
{
std::lock_guard<std::mutex> lock(sendQueueMutex_);
// This is a byte-bounded queue, not a packet-count guess: one large
// keyframe cannot turn a short receiver stall into unbounded latency.
if (sendWorkerStopping_ ||
byteCount > kMaximumSendQueueBytes ||
sendQueueBytes_ > kMaximumSendQueueBytes - byteCount) {
healthy_ = false;
sendQueue_.clear();
sendQueueBytes_ = 0;
__android_log_print(
ANDROID_LOG_WARN,
kTag,
"SRT send queue saturated; dropping the session to avoid latency growth");
return false;
}
sendQueue_.push_back(std::move(bytes));
sendQueueBytes_ += byteCount;
}
sendQueueWake_.notify_one();
return true;
#else
(void)bytes;
return false;
#endif
}

void disconnect() {
#if OPENSTREAM_HAVE_LIBSRT
healthy_ = false;
{
std::lock_guard<std::mutex> lock(sendQueueMutex_);
sendQueue_.clear();
sendQueueBytes_ = 0;
}
std::lock_guard<std::mutex> ioLock(ioMutex_);
const SRTSOCKET listenerSocket = takeListenerSocket();
const SRTSOCKET socket = takeSocket();
Expand All @@ -671,6 +735,39 @@ class NativeSender {
}

private:
void runSendWorker() {
for (;;) {
std::vector<uint8_t> bytes;
{
std::unique_lock<std::mutex> lock(sendQueueMutex_);
sendQueueWake_.wait(lock, [this] {
return sendWorkerStopping_ || !sendQueue_.empty();
});
if (sendWorkerStopping_) return;
bytes = std::move(sendQueue_.front());
sendQueue_.pop_front();
sendQueueBytes_ -= bytes.size();
}
if (!sendNow(bytes)) {
healthy_ = false;
std::lock_guard<std::mutex> lock(sendQueueMutex_);
sendQueue_.clear();
sendQueueBytes_ = 0;
}
}
}

void stopSendWorker() {
{
std::lock_guard<std::mutex> lock(sendQueueMutex_);
sendWorkerStopping_ = true;
sendQueue_.clear();
sendQueueBytes_ = 0;
}
sendQueueWake_.notify_one();
if (sendWorker_.joinable()) sendWorker_.join();
}

#if OPENSTREAM_HAVE_LIBSRT
SRTSOCKET currentSocket() const {
std::lock_guard<std::mutex> lock(socketMutex_);
Expand All @@ -680,6 +777,7 @@ class NativeSender {
void setSocket(SRTSOCKET socket) {
std::lock_guard<std::mutex> lock(socketMutex_);
socket_ = socket;
healthy_ = true;
}

SRTSOCKET takeSocket() {
Expand All @@ -695,6 +793,7 @@ class NativeSender {
std::lock_guard<std::mutex> lock(socketMutex_);
if (socket_ == socket) {
socket_ = SRT_INVALID_SOCK;
healthy_ = false;
ownsSocket = true;
}
}
Expand Down Expand Up @@ -734,6 +833,14 @@ class NativeSender {
SRTSOCKET socket_ = SRT_INVALID_SOCK;
SRTSOCKET listener_socket_ = SRT_INVALID_SOCK;
#endif
static constexpr size_t kMaximumSendQueueBytes = 768 * 1024;
std::atomic<bool> healthy_{false};
std::mutex sendQueueMutex_;
std::condition_variable sendQueueWake_;
std::deque<std::vector<uint8_t>> sendQueue_;
size_t sendQueueBytes_ = 0;
bool sendWorkerStopping_ = false;
std::thread sendWorker_;
};

struct StreamState {
Expand Down Expand Up @@ -863,9 +970,9 @@ Java_dev_openstream_app_stream_SrtNativeBridge_sendVideo(
annexB = std::move(withConfig);
}

const std::vector<uint8_t> ts =
std::vector<uint8_t> ts =
g_state.muxer->muxAccessUnit(annexB, static_cast<int64_t>(presentation_time_us), keyFrame);
const bool sent = g_state.sender.send(ts);
const bool sent = g_state.sender.send(std::move(ts));
if (!sent) {
g_state.connected = false;
}
Expand Down Expand Up @@ -905,9 +1012,9 @@ Java_dev_openstream_app_stream_SrtNativeBridge_sendAudio(
return JNI_TRUE;
}

const std::vector<uint8_t> ts = g_state.muxer->muxAudioAccessUnit(
std::vector<uint8_t> ts = g_state.muxer->muxAudioAccessUnit(
bytes, g_state.audioCodecConfig, static_cast<int64_t>(presentation_time_us));
const bool sent = g_state.sender.send(ts);
const bool sent = g_state.sender.send(std::move(ts));
if (!sent) {
g_state.connected = false;
}
Expand Down
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