Index: webrtc/modules/audio_coding/acm2/acm_resampler.cc |
diff --git a/webrtc/modules/audio_coding/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/acm2/acm_resampler.cc |
index d7ceb8ac9f2991d0f0f4ba6ced1f6231cd1bf271..dfc3ef7e2746fd4d28ea62d51f058e9afb499a2f 100644 |
--- a/webrtc/modules/audio_coding/acm2/acm_resampler.cc |
+++ b/webrtc/modules/audio_coding/acm2/acm_resampler.cc |
@@ -28,10 +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); |
+ 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); |
@@ -56,7 +56,7 @@ 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 |