Index: content/renderer/media/webrtc_audio_device_impl.cc |
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc |
index c40e0a2219b773eea21536cfc38d2259f3f966ea..93e024bc6b2ef9b894ca62b0973a90a0298ce0a1 100644 |
--- a/content/renderer/media/webrtc_audio_device_impl.cc |
+++ b/content/renderer/media/webrtc_audio_device_impl.cc |
@@ -59,7 +59,8 @@ int WebRtcAudioDeviceImpl::CaptureData(const std::vector<int>& channels, |
int number_of_frames, |
int audio_delay_milliseconds, |
int current_volume, |
- bool need_audio_processing) { |
+ bool need_audio_processing, |
+ bool key_pressed) { |
int total_delay_ms = 0; |
{ |
base::AutoLock auto_lock(lock_); |
@@ -75,11 +76,9 @@ int WebRtcAudioDeviceImpl::CaptureData(const std::vector<int>& channels, |
// Write audio samples in blocks of 10 milliseconds to the registered |
// webrtc::AudioTransport sink. Keep writing until our internal byte |
// buffer is empty. |
- // TODO(niklase): Wire up the key press detection. |
const int16* audio_buffer = audio_data; |
const int samples_per_10_msec = (sample_rate / 100); |
int accumulated_audio_samples = 0; |
- bool key_pressed = false; |
uint32_t new_volume = 0; |
while (accumulated_audio_samples < number_of_frames) { |
// Deliver 10ms of recorded 16-bit linear PCM audio. |