Index: content/renderer/media/webrtc_audio_device_impl.h |
diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h |
index 490eaf4a6d90ecd3770d0e4f5611c38b34f54ba0..52cb2a0dd5b34b78e5eb92508ce15ce05f19b36c 100644 |
--- a/content/renderer/media/webrtc_audio_device_impl.h |
+++ b/content/renderer/media/webrtc_audio_device_impl.h |
@@ -380,9 +380,6 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl |
// RegisterAudioCallback(). |
webrtc::AudioTransport* audio_transport_callback_; |
- // Cached value of the current audio delay on the input/capture side. |
- int input_delay_ms_; |
- |
// Cached value of the current audio delay on the output/renderer side. |
int output_delay_ms_; |
@@ -398,10 +395,6 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl |
bool playing_; |
bool recording_; |
- // Stores latest microphone volume received in a CaptureData() callback. |
- // Range is [0, 255]. |
- uint32_t microphone_volume_; |
- |
// Buffer used for temporary storage during render callback. |
// It is only accessed by the audio render thread. |
std::vector<int16_t> render_buffer_; |