Moves channel-dependent audio input processing to separate encoder task queue

webrtc/modules/audio_device/audio_device_buffer.cc

Index: webrtc/modules/audio_device/audio_device_buffer.cc
diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc
index 8b49a3ffd0b9592586e55a546d9ad2375256205a..9322a1459fffb8e4a2638403fdf33ab4763cf77e 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.cc
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc
@@ -322,6 +322,7 @@ int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
 }
 
 int32_t AudioDeviceBuffer::DeliverRecordedData() {
+  int64_t start_time = rtc::TimeMicros();
   RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   if (!audio_transport_cb_) {
     LOG(LS_WARNING) << "Invalid audio transport";
@@ -340,6 +341,14 @@ int32_t AudioDeviceBuffer::DeliverRecordedData() {
   } else {
     LOG(LS_ERROR) << "RecordedDataIsAvailable() failed";
   }
+
+  int64_t diff_time = rtc::TimeMicros() - start_time;
+  acc_diff_times_ += diff_time;
+  num_diff_times_++;
+  if (diff_time > max_diff_time_) {
+    max_diff_time_ = diff_time;
+  }
+
   return 0;
 }
 
@@ -465,6 +474,10 @@ void AudioDeviceBuffer::LogStats(LogState state) {
     last_stats_ = stats;
   }
 
+  double avg_diff_time = (double)acc_diff_times_ / num_diff_times_;
+  LOG(INFO) << "avg_diff_time: " << avg_diff_time;
+  LOG(INFO) << "max_diff_time: " << max_diff_time_;
+
   int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis();
   RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval";