Index: webrtc/voice_engine/utility.cc |
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc |
index eb442ecb8f74c8ac607b1f74c85274a0566520b5..605e55369e3d4a31666e17bf69d3b04ccd054630 100644 |
--- a/webrtc/voice_engine/utility.cc |
+++ b/webrtc/voice_engine/utility.cc |
@@ -34,12 +34,12 @@ void RemixAndResample(const AudioFrame& src_frame, |
void RemixAndResample(const int16_t* src_data, |
size_t samples_per_channel, |
- int num_channels, |
+ size_t num_channels, |
int sample_rate_hz, |
PushResampler<int16_t>* resampler, |
AudioFrame* dst_frame) { |
const int16_t* audio_ptr = src_data; |
- int audio_ptr_num_channels = num_channels; |
+ size_t audio_ptr_num_channels = num_channels; |
int16_t mono_audio[AudioFrame::kMaxDataSizeSamples]; |
// Downmix before resampling. |
@@ -68,8 +68,7 @@ void RemixAndResample(const int16_t* src_data, |
<< ", dst_frame->data_ = " << dst_frame->data_; |
assert(false); |
} |
- dst_frame->samples_per_channel_ = |
- static_cast<size_t>(out_length / audio_ptr_num_channels); |
+ dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels; |
// Upmix after resampling. |
if (num_channels == 1 && dst_frame->num_channels_ == 2) { |
@@ -81,9 +80,9 @@ void RemixAndResample(const int16_t* src_data, |
} |
void MixWithSat(int16_t target[], |
- int target_channel, |
+ size_t target_channel, |
const int16_t source[], |
- int source_channel, |
+ size_t source_channel, |
size_t source_len) { |
assert(target_channel == 1 || target_channel == 2); |
assert(source_channel == 1 || source_channel == 2); |