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 5df87d2a19bf7021520258f8984d259f8280823a..d7ceb8ac9f2991d0f0f4ba6ced1f6231cd1bf271 100644 |
--- a/webrtc/modules/audio_coding/acm2/acm_resampler.cc |
+++ b/webrtc/modules/audio_coding/acm2/acm_resampler.cc |
@@ -32,7 +32,6 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio, |
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; |
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(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", " |