Index: webrtc/modules/audio_coding/main/acm2/acm_resampler.cc |
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc |
index 2650725331b94e5db5e110836704d11082657ec6..13c8f9d5e8639231e9816b4873b00f24687b1e5a 100644 |
--- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc |
+++ b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc |
@@ -28,11 +28,10 @@ ACMResampler::~ACMResampler() { |
int ACMResampler::Resample10Msec(const int16_t* in_audio, |
int in_freq_hz, |
int out_freq_hz, |
- int num_audio_channels, |
+ size_t num_audio_channels, |
size_t out_capacity_samples, |
int16_t* out_audio) { |
- size_t in_length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100); |
- int out_length = out_freq_hz * num_audio_channels / 100; |
+ size_t in_length = in_freq_hz * num_audio_channels / 100; |
if (in_freq_hz == out_freq_hz) { |
if (out_capacity_samples < in_length) { |
assert(false); |
@@ -49,7 +48,7 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio, |
return -1; |
} |
- out_length = |
+ int out_length = |
resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples); |
if (out_length == -1) { |
LOG_FERR4(LS_ERROR, |
@@ -61,7 +60,7 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio, |
return -1; |
} |
- return out_length / num_audio_channels; |
+ return static_cast<int>(out_length / num_audio_channels); |
} |
} // namespace acm2 |