fix: use condition variable for audio buffer synchronization

Introduces a std::condition_variable to WebRTCService for improved synchronization of the audio buffer. The audio thread now waits on the condition variable instead of busy-waiting, and the data channel handler notifies the thread when new audio data arrives. This change reduces CPU usage and improves efficiency.

Author: std-microblock

src/client/main.cc

@@ -671,8 +671,9 @@ void ClientContext::init_webrtc_service() {
   if (!audio_thread) {
     audio_thread.emplace([&]() {
       while (true) {
-        std::this_thread::yield();
         std::unique_lock<std::mutex> lock(webrtc_service->audioBufferMutex_);
+        webrtc_service->audioBufferCv_.wait(
+            lock, [&]() { return !webrtc_service->audioBuffer_.empty(); });
         while (webrtc_service->audioBuffer_.size() >= 960) {
           std::vector<uint8_t> buffer(webrtc_service->audioBuffer_.begin(),
                                       webrtc_service->audioBuffer_.begin() +

src/client/webrtc_service.cc

@@ -109,6 +109,7 @@ void WebRTCService::setup_data_channel() {
       }
       std::lock_guard<std::mutex> lock(audioBufferMutex_);
       audioBuffer_.insert(audioBuffer_.end(), data.begin(), data.end());
+      audioBufferCv_.notify_one();
     }
   });
 }

src/client/webrtc_service.h

@@ -3,6 +3,7 @@
 
 #include "httplib.h"
 #include "rtc/configuration.hpp"
+#include <condition_variable>
 #include <deque>
 #include <functional>
 #include <memory>
@@ -39,6 +40,7 @@ struct WebRTCService {
     std::shared_ptr<rtc::PeerConnection> pc_;
     std::shared_ptr<rtc::DataChannel> dc_;
     std::deque<uint8_t> audioBuffer_;
+    std::condition_variable audioBufferCv_;
     std::mutex audioBufferMutex_;
     StatusCallback status_callback_;
     std::string session_id_;