Index: media/audio/alsa/audio_manager_alsa.cc |
diff --git a/media/audio/alsa/audio_manager_alsa.cc b/media/audio/alsa/audio_manager_alsa.cc |
index beb60bad88b02af515dd1075fb54a43edd226cb6..76248348c464fd88575437bd2ea582ba4f755398 100644 |
--- a/media/audio/alsa/audio_manager_alsa.cc |
+++ b/media/audio/alsa/audio_manager_alsa.cc |
@@ -311,7 +311,6 @@ |
int sample_rate = kDefaultSampleRate; |
int buffer_size = kDefaultOutputBufferSize; |
int bits_per_sample = 16; |
- int input_channels = 0; |
if (input_params.IsValid()) { |
// Some clients, such as WebRTC, have a more limited use case and work |
// acceptably with a smaller buffer size. The check below allows clients |
@@ -321,7 +320,6 @@ |
sample_rate = input_params.sample_rate(); |
bits_per_sample = input_params.bits_per_sample(); |
channel_layout = input_params.channel_layout(); |
- input_channels = input_params.input_channels(); |
buffer_size = std::min(input_params.frames_per_buffer(), buffer_size); |
} |
@@ -330,7 +328,7 @@ |
buffer_size = user_buffer_size; |
return AudioParameters( |
- AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels, |
+ AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, |
sample_rate, bits_per_sample, buffer_size, AudioParameters::NO_EFFECTS); |
} |