| 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 << ", "
|
|
|