Fix the maximum native sample rate in AudioProcessing

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 129 matching lines @@
 
   // VolumeCallbacks implementation.
   void SetMicVolume(int volume) override { volume_ = volume; }
   int GetMicVolume() override { return volume_; }
 
  private:
   GainControl* real_gain_control_;
   int volume_;
 };
 
+const int AudioProcessing::kNativeSampleRatesHz[] = {
+    AudioProcessing::kSampleRate8kHz,
+    AudioProcessing::kSampleRate16kHz,
+    AudioProcessing::kSampleRate32kHz,
+    AudioProcessing::kSampleRate48kHz};
+const size_t AudioProcessing::kNumNativeSampleRates =
+    arraysize(AudioProcessing::kNativeSampleRatesHz);
+const int AudioProcessing::kMaxNativeSampleRateHz = AudioProcessing::
+    kNativeSampleRatesHz[AudioProcessing::kNumNativeSampleRates - 1];
+
 AudioProcessing* AudioProcessing::Create() {
   Config config;
   return Create(config, nullptr);
 }
 
 AudioProcessing* AudioProcessing::Create(const Config& config) {
   return Create(config, nullptr);
 }
 
 AudioProcessing* AudioProcessing::Create(const Config& config,
@@ skipping 233 matching lines @@
     return kBadNumberChannelsError;
   }
 
   api_format_ = config;
 
   // We process at the closest native rate >= min(input rate, output rate)...
   const int min_proc_rate =
       std::min(api_format_.input_stream().sample_rate_hz(),
                api_format_.output_stream().sample_rate_hz());
   int fwd_proc_rate;
-  if (min_proc_rate > kSampleRate32kHz) {
-    fwd_proc_rate = kSampleRate48kHz;
-  } else if (min_proc_rate > kSampleRate16kHz) {
-    fwd_proc_rate = kSampleRate32kHz;
-  } else if (min_proc_rate > kSampleRate8kHz) {
-    fwd_proc_rate = kSampleRate16kHz;
-  } else {
-    fwd_proc_rate = kSampleRate8kHz;
+  for (size_t i = 0; i < kNumNativeSampleRates; ++i) {
+    fwd_proc_rate = kNativeSampleRatesHz[i];
+    if (fwd_proc_rate >= min_proc_rate) {
+      break;
+    }
   }
   // ...with one exception.
   if (echo_control_mobile_->is_enabled() && min_proc_rate > kSampleRate16kHz) {
     fwd_proc_rate = kSampleRate16kHz;
   }
 
   fwd_proc_format_ = StreamConfig(fwd_proc_rate);
 
   // We normally process the reverse stream at 16 kHz. Unless...
   int rev_proc_rate = kSampleRate16kHz;
@@ skipping 821 matching lines @@
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc