diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index 27c973a06a..51e19e7b0f 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -412,7 +412,7 @@ void RtpAudioStream::decodeAudio(QAudioDeviceInfo out_device)
             // Buffer is in SAMPLEs
             spx_uint32_t in_len = (spx_uint32_t) (write_bytes / SAMPLE_BYTES);
             // Output is audio_out_rate_/sample_rate bigger than input
-            spx_uint32_t out_len = (spx_uint32_t) (in_len * audio_out_rate_ / sample_rate);
+            spx_uint32_t out_len = (spx_uint32_t) ((guint64)in_len * audio_out_rate_ / sample_rate);
             resample_buff = resizeBufferIfNeeded(resample_buff, &resample_buff_bytes, out_len * SAMPLE_BYTES);
 
             speex_resampler_process_int(audio_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
@@ -453,7 +453,7 @@ void RtpAudioStream::decodeVisual()
     // Loop over every frame record
     // readFrameSamples() maintains size of buffer for us
     while (audio_file_->readFrameSamples(&read_buff_bytes, &read_buff, &read_len, &frame_num, &type)) {
-        out_len = (spx_uint32_t)((read_len * visual_sample_rate_ ) / audio_out_rate_);
+        out_len = (spx_uint32_t)(((guint64)read_len * visual_sample_rate_ ) / audio_out_rate_);
 
         if (type == RTP_FRAME_AUDIO) {
             // We resample only audio samples