Skip to content

use the request_async_id in cpp #43

New issue

Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.

Sign up for GitHub

By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.

Already on GitHub? Sign in to your account

Open
xianshijing-lk wants to merge 3 commits into
base: main
Choose a base branch
from
Open

use the request_async_id in cpp #43

Show file tree
Hide file tree
Changes from all commits
Commits
Show all changes
3 commits
Select commit Hold shift + click to select a range
237a243
use the request_async_id in cpp
xianshijing-lk Jan 19, 2026
a444821
changed the code back to always throw when ffi returns invalid result
xianshijing-lk Jan 19, 2026
5f1bf88
fix the timeout issue and a few bugs related to the SimpleRoom example
xianshijing-lk Jan 29, 2026
File filter

Filter by extension

Filter by extension
Conversations
Failed to load comments.
Loading
Jump to
Jump to file
Failed to load files.
Loading
Diff view
Diff view
2 changes: 1 addition & 1 deletion client-sdk-rust
Open in desktop
Submodule client-sdk-rust updated from 3a3f42 to 6e4773
4 changes: 3 additions & 1 deletion examples/simple_room/main.cpp
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -319,7 +319,7 @@ int main(int argc, char *argv[]) {
<< " Creation time (ms): " << info.creation_time << "\n";

// Setup Audio Source / Track
auto audioSource = std::make_shared<AudioSource>(44100, 1, 10);
auto audioSource = std::make_shared<AudioSource>(44100, 1, 0);
auto audioTrack =
LocalAudioTrack::createLocalAudioTrack("micTrack", audioSource);

Expand Down Expand Up @@ -385,6 +385,8 @@ int main(int argc, char *argv[]) {
// Shutdown the audio / video capture threads.
media.stopMic();
media.stopCamera();
media.stopSpeaker();
media.shutdownRenderer();

// Drain any queued tasks that might still try to update the renderer /
// speaker
Expand Down
1 change: 1 addition & 0 deletions examples/simple_room/sdl_media_manager.cpp
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -31,6 +31,7 @@ SDLMediaManager::~SDLMediaManager() {
stopMic();
stopCamera();
stopSpeaker();
shutdownRenderer();
}

bool SDLMediaManager::ensureSDLInit(Uint32 flags) {
Expand Down
12 changes: 12 additions & 0 deletions examples/simple_room/sdl_video_renderer.cpp
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -22,6 +22,8 @@

using namespace livekit;

constexpr int kMaxFPS = 60;

SDLVideoRenderer::SDLVideoRenderer() = default;

SDLVideoRenderer::~SDLVideoRenderer() { shutdown(); }
Expand Down Expand Up @@ -95,6 +97,16 @@ void SDLVideoRenderer::render() {
return;
}

// Throttle rendering to kMaxFPS
const auto now = std::chrono::steady_clock::now();
if (last_render_time_.time_since_epoch().count() != 0) {
const auto min_interval = std::chrono::microseconds(1'000'000 / kMaxFPS);
if (now - last_render_time_ < min_interval) {
return;
}
}
last_render_time_ = now;

// 3) Read a frame from VideoStream (blocking until one is available)
livekit::VideoFrameEvent vfe;
bool gotFrame = stream_->read(vfe);
Expand Down
1 change: 1 addition & 0 deletions examples/simple_room/sdl_video_renderer.h
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -49,4 +49,5 @@ class SDLVideoRenderer {
std::shared_ptr<livekit::VideoStream> stream_;
int width_ = 0;
int height_ = 0;
std::chrono::steady_clock::time_point last_render_time_{};
};
63 changes: 51 additions & 12 deletions include/livekit/audio_source.h
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -41,8 +41,34 @@ class AudioSource {
* @param sample_rate Sample rate in Hz.
* @param num_channels Number of channels.
* @param queue_size_ms Max buffer duration for the internal queue in ms.
*
* Buffering behavior:
* -------------------
* - queue_size_ms == 0 (recommended for real-time capture):
* Disables internal buffering entirely. Audio frames are forwarded
* directly to WebRTC sinks and consumed synchronously.
*
* This mode is optimized for real-time audio capture driven by hardware
* media callbacks (e.g. microphone capture). The caller is expected to
* provide fixed-size real-time frames (typically 10 ms per call).
*
* Because the native side consumes frames immediately, this mode
* minimizes latency and jitter and is the best choice for live capture
* scenarios.
*
* - queue_size_ms > 0 (buffered / blocking mode):
* Enables an internal queue that buffers audio up to the specified
* duration. Frames are accumulated and flushed asynchronously once the buffer
* reaches its threshold.
*
* This mode is intended for non-real-time producers (e.g. TTS engines,
* file-based audio, or agents generating audio faster or slower than
* real-time). The buffering layer smooths timing and allows the audio to
* be streamed out in real time even if the producer is bursty.
*
* queue_size_ms must be a multiple of 10.
*/
AudioSource(int sample_rate, int num_channels, int queue_size_ms = 1000);
AudioSource(int sample_rate, int num_channels, int queue_size_ms = 0);
virtual ~AudioSource() = default;

AudioSource(const AudioSource &) = delete;
Expand Down Expand Up @@ -86,19 +112,32 @@ class AudioSource {
* callback arrives (recommended for production unless the caller needs
* explicit timeout control).
*
* Notes:
* - This is a blocking call.
* - timeout_ms == 0 (infinite wait) is the safest mode because it
* guarantees the callback completes before the function returns, which in
* turn guarantees that the audio buffer lifetime is fully protected. The
* caller does not need to manage or extend the frame lifetime manually.
* Blocking semantics:
* The blocking behavior of this call depends on the buffering mode selected
* at construction time:
*
* - queue_size_ms == 0 (real-time capture mode):
* Frames are consumed synchronously by the native layer. The FFI callback
* is invoked immediately as part of the capture call, so this function
* returns quickly.
*
* This mode relies on the caller being paced by a real-time media
* callback (e.g. audio hardware interrupt / capture thread). It provides the
* lowest possible latency and is ideal for live microphone capture.
*
* - queue_size_ms > 0 (buffered / non-real-time mode):
* Frames are queued internally and flushed asynchronously. This function
* will block until the buffered audio corresponding to this frame has
* been consumed by the native side and the FFI callback fires.
*
* - May throw std::runtime_error if:
* • the FFI reports an error
* This mode is best suited for non-real-time audio producers (such as TTS
* engines or agents) that generate audio independently of real-time
* pacing, while still streaming audio out in real time.
*
* - The underlying FFI request *must* eventually produce a callback for
* each frame. If the FFI layer is misbehaving or the event loop is stalled,
* a timeout may occur in bounded-wait mode.
* Safety notes:
* May throw std::runtime_error if:
* - the FFI reports an error
* - a timeout occurs in bounded-wait mode
*/
void captureFrame(const AudioFrame &frame, int timeout_ms = 20);

Expand Down
Loading
Loading