Index: content/renderer/media/webrtc_audio_device_unittest.cc |
diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc |
index e988aa281e1c02773b777112852e3c1bb73a32e9..8453f38d04a5e5183e76cd61d716e81441fd506d 100644 |
--- a/content/renderer/media/webrtc_audio_device_unittest.cc |
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc |
@@ -41,7 +41,7 @@ scoped_ptr<media::AudioHardwareConfig> CreateRealHardwareConfig() { |
} |
// Return true if at least one element in the array matches |value|. |
-bool FindElementInArray(int* array, int size, int value) { |
+bool FindElementInArray(const int* array, int size, int value) { |
return (std::find(&array[0], &array[0] + size, value) != &array[size]); |
} |
@@ -56,7 +56,7 @@ bool HardwareSampleRatesAreValid() { |
// The actual WebRTC client can limit these ranges further depending on |
// platform but this is the maximum range we support today. |
int valid_input_rates[] = {16000, 32000, 44100, 48000, 96000}; |
- int valid_output_rates[] = {44100, 48000, 96000}; |
+ int valid_output_rates[] = {16000, 32000, 44100, 48000, 96000}; |
media::AudioHardwareConfig* hardware_config = |
RenderThreadImpl::current()->GetAudioHardwareConfig(); |