Index: webrtc/modules/audio_coding/neteq/accelerate.cc |
diff --git a/webrtc/modules/audio_coding/neteq/accelerate.cc b/webrtc/modules/audio_coding/neteq/accelerate.cc |
index ad7423810dfcc1f21c1e55da7238129f817a5dcc..1c36fa8c612a225cbe16e0a1a06bb84be98d744d 100644 |
--- a/webrtc/modules/audio_coding/neteq/accelerate.cc |
+++ b/webrtc/modules/audio_coding/neteq/accelerate.cc |
@@ -18,11 +18,11 @@ Accelerate::ReturnCodes Accelerate::Process(const int16_t* input, |
size_t input_length, |
bool fast_accelerate, |
AudioMultiVector* output, |
- int16_t* length_change_samples) { |
+ size_t* length_change_samples) { |
// Input length must be (almost) 30 ms. |
- static const int k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate. |
- if (num_channels_ == 0 || static_cast<int>(input_length) / num_channels_ < |
- (2 * k15ms - 1) * fs_mult_) { |
+ static const size_t k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate. |
+ if (num_channels_ == 0 || |
+ input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) { |
// Length of input data too short to do accelerate. Simply move all data |
// from input to output. |
output->PushBackInterleaved(input, input_length); |
@@ -34,7 +34,7 @@ Accelerate::ReturnCodes Accelerate::Process(const int16_t* input, |
void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/, |
int16_t* best_correlation, |
- int* /*peak_index*/) const { |
+ size_t* /*peak_index*/) const { |
// When the signal does not contain any active speech, the correlation does |
// not matter. Simply set it to zero. |
*best_correlation = 0; |