| Index: webrtc/modules/audio_coding/acm2/acm_resampler.cc
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| diff --git a/webrtc/modules/audio_coding/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/acm2/acm_resampler.cc
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| index e38cd94a97887da69f31d190f1f931f478e50b65..5df87d2a19bf7021520258f8984d259f8280823a 100644
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| --- a/webrtc/modules/audio_coding/acm2/acm_resampler.cc
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| +++ b/webrtc/modules/audio_coding/acm2/acm_resampler.cc
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| @@ -44,20 +44,16 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio,
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|  
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|    if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
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|                                      num_audio_channels) != 0) {
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| -    LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz,
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| -              num_audio_channels);
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| +    LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", " << out_freq_hz
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| +                  << ", " << num_audio_channels << ") failed.";
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|      return -1;
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|    }
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|  
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|    out_length =
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|        resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
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|    if (out_length == -1) {
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| -    LOG_FERR4(LS_ERROR,
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| -              Resample,
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| -              in_audio,
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| -              in_length,
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| -              out_audio,
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| -              out_capacity_samples);
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| +    LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
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| +                  << out_audio << ", " << out_capacity_samples << ") failed.";
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|      return -1;
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|    }
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|  
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| 
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