1 files changed, 27 insertions, 7 deletions

diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index fde66c8127..e7d71245c6 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -229,6 +229,12 @@ void RtpAudioStream::decode()
 
         size_t decoded_bytes = decode_rtp_packet(rtp_packet, &decode_buff, decoders_hash_, &channels, &sample_rate);
 
+        unsigned rtp_clock_rate = sample_rate;
+        if (rtp_packet->info->info_payload_type == PT_G722) {
+            // G.722 sample rate is 16kHz, but RTP clock rate is 8kHz for historic reasons.
+            rtp_clock_rate = 8000;
+        }
+
         if (decoded_bytes == 0 || sample_rate == 0) {
             // We didn't decode anything. Clean up and prep for the next packet.
             last_sequence = rtp_packet->info->info_seq_num;
@@ -236,7 +242,27 @@ void RtpAudioStream::decode()
             continue;
         }
 
-        if (audio_out_rate_ == 0) { // First non-zero wins
+        if (audio_out_rate_ == 0) {
+            // Use the first non-zero rate we find. Ajust it to match our audio hardware.
+            QAudioDeviceInfo cur_out_device = QAudioDeviceInfo::defaultOutputDevice();
+            QString cur_out_name = parent()->property("currentOutputDeviceName").toString();
+            foreach (QAudioDeviceInfo out_device, QAudioDeviceInfo::availableDevices(QAudio::AudioOutput)) {
+                if (cur_out_name == out_device.deviceName()) {
+                    cur_out_device = out_device;
+                }
+            }
+
+            QAudioFormat format;
+            format.setSampleRate(sample_rate);
+            format.setSampleSize(sample_bytes_ * 8); // bits
+            format.setSampleType(QAudioFormat::SignedInt);
+            format.setChannelCount(1);
+            format.setCodec("audio/pcm");
+
+            if (!cur_out_device.isFormatSupported(format)) {
+                sample_rate = cur_out_device.nearestFormat(format).sampleRate();
+            }
+
             audio_out_rate_ = sample_rate;
             RTP_STREAM_DEBUG("Audio sample rate is %u", audio_out_rate_);
 
@@ -253,12 +279,6 @@ void RtpAudioStream::decode()
         }
         last_sequence = rtp_packet->info->info_seq_num;
 
-        unsigned rtp_clock_rate = sample_rate;
-        if (rtp_packet->info->info_payload_type == PT_G722) {
-            // G.722 sample rate is 16kHz, but RTP clock rate is 8kHz for historic reasons.
-            rtp_clock_rate = 8000;
-        }
-
         double rtp_time = (double)(rtp_packet->info->info_timestamp-start_timestamp)/rtp_clock_rate - start_rtp_time;
         double arrive_time;
         if (timing_mode_ == RtpTimestamp) {