Make audioLevelUpdated parameter optional.

TelegramMessenger · Jan 25, 2024 · b9fa8b8 · b9fa8b8
1 parent df78cf0
commit b9fa8b8
Showing 1 changed file with 10 additions and 12 deletions.
diff --git a/tgcalls/v2/InstanceV2Impl.cpp b/tgcalls/v2/InstanceV2Impl.cpp
@@ -259,15 +259,15 @@ class AudioSinkImpl: public webrtc::AudioSinkInterface {
     AudioSinkImpl(std::function<void(float)> update) :
     _update(update) {
     }
-    
+
     virtual ~AudioSinkImpl() {
     }
-    
+
     virtual void OnData(const Data& audio) override {
         if (_update && audio.channels == 1) {
             const int16_t *samples = (const int16_t *)audio.data;
             int numberOfSamplesInFrame = (int)audio.samples_per_channel;
-            
+
             int16_t currentPeak = 0;
             for (int i = 0; i < numberOfSamplesInFrame; i++) {
                 int16_t sample = samples[i];
@@ -282,7 +282,7 @@ class AudioSinkImpl: public webrtc::AudioSinkInterface {
                 }
                 _peakCount += 1;
             }
-            
+
             if (_peakCount >= 4400) {
                 float level = ((float)(_peak)) / 8000.0f;
                 _peak = 0;
@@ -291,10 +291,10 @@ class AudioSinkImpl: public webrtc::AudioSinkInterface {
             }
         }
     }
-    
+
 private:
     std::function<void(float)> _update;
-    
+
     int _peakCount = 0;
     uint16_t _peak = 0;
 };
@@ -309,7 +309,7 @@ class IncomingV2AudioChannel : public sigslot::has_slots<> {
         webrtc::RtpTransport *rtpTransport,
         rtc::UniqueRandomIdGenerator *randomIdGenerator,
         signaling::MediaContent const &mediaContent,
-        std::function<void(float)> &&onAudioLevelUpdated,
+        std::function<void(float)> onAudioLevelUpdated,
         std::shared_ptr<Threads> threads) :
     _threads(threads),
     _ssrc(mediaContent.ssrc),
@@ -327,7 +327,7 @@ class IncomingV2AudioChannel : public sigslot::has_slots<> {
         _threads->getNetworkThread()->BlockingCall([&]() {
             _audioChannel->SetRtpTransport(rtpTransport);
         });
-        
+
         std::vector<cricket::AudioCodec> codecs;
         for (const auto &payloadType : mediaContent.payloadTypes) {
             cricket::AudioCodec codec(payloadType.id, payloadType.name, payloadType.clockrate, 0, payloadType.channels);
@@ -369,7 +369,7 @@ class IncomingV2AudioChannel : public sigslot::has_slots<> {
             std::string errorDesc;
             _audioChannel->SetLocalContent(outgoingAudioDescription.get(), webrtc::SdpType::kOffer, errorDesc);
             _audioChannel->SetRemoteContent(incomingAudioDescription.get(), webrtc::SdpType::kAnswer, errorDesc);
-            
+
             std::unique_ptr<AudioSinkImpl> audioLevelSink(new AudioSinkImpl(std::move(onAudioLevelUpdated)));
             _audioChannel->media_channel()->SetRawAudioSink(ssrc, std::move(audioLevelSink));
         });
@@ -1517,9 +1517,7 @@ class InstanceV2ImplInternal : public std::enable_shared_from_this<InstanceV2Imp
                             _rtpTransport,
                             _uniqueRandomIdGenerator.get(),
                             content,
-                            [audioLevelUpdated = _audioLevelUpdated](float level) {
-                                audioLevelUpdated(level);
-                            },
+                            _audioLevelUpdated,
                             _threads
                         ));
                     }