Drop the 16kHz sample rate restriction on AECM and zero out higher bands

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 100 matching lines @@
 
 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];
-const int AudioProcessing::kMaxAECMSampleRateHz = kSampleRate16kHz;
 
 AudioProcessing* AudioProcessing::Create() {
   Config config;
   return Create(config, nullptr);
 }
 
 AudioProcessing* AudioProcessing::Create(const Config& config) {
   return Create(config, nullptr);
 }
 
@@ skipping 226 matching lines @@
     return kBadNumberChannelsError;
   }
 
   if (capture_nonlocked_.beamformer_enabled &&
       num_in_channels != capture_.array_geometry.size()) {
     return kBadNumberChannelsError;
   }
 
   formats_.api_format = config;
 
-  // We process at the closest native rate >= min(input rate, output rate)...
+  // We process at the closest native rate >= min(input rate, output rate).
   const int min_proc_rate =
       std::min(formats_.api_format.input_stream().sample_rate_hz(),
                formats_.api_format.output_stream().sample_rate_hz());
   int fwd_proc_rate;
   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 (public_submodules_->echo_control_mobile->is_enabled() &&
-      min_proc_rate > kMaxAECMSampleRateHz) {
-    fwd_proc_rate = kMaxAECMSampleRateHz;
-  }
 
   capture_nonlocked_.fwd_proc_format = StreamConfig(fwd_proc_rate);
 
   // We normally process the reverse stream at 16 kHz. Unless...
   int rev_proc_rate = kSampleRate16kHz;
   if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate8kHz) {
     // ...the forward stream is at 8 kHz.
     rev_proc_rate = kSampleRate8kHz;
   } else {
     if (formats_.api_format.reverse_input_stream().sample_rate_hz() ==
@@ skipping 215 matching lines @@
     return kNullPointerError;
   }
   // Must be a native rate.
   if (frame->sample_rate_hz_ != kSampleRate8kHz &&
       frame->sample_rate_hz_ != kSampleRate16kHz &&
       frame->sample_rate_hz_ != kSampleRate32kHz &&
       frame->sample_rate_hz_ != kSampleRate48kHz) {
     return kBadSampleRateError;
   }
 
-  if (public_submodules_->echo_control_mobile->is_enabled() &&
-      frame->sample_rate_hz_ > kMaxAECMSampleRateHz) {
-    LOG(LS_ERROR) << "AECM only supports 16 or 8 kHz sample rates";
-    return kUnsupportedComponentError;
-  }
-
   ProcessingConfig processing_config;
   {
     // Aquire lock for the access of api_format.
     // The lock is released immediately due to the conditional
     // reinitialization.
     rtc::CritScope cs_capture(&crit_capture_);
     // TODO(ajm): The input and output rates and channels are currently
     // constrained to be identical in the int16 interface.
     processing_config = formats_.api_format;
   }
@@ skipping 821 matching lines @@
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &debug_dump_.num_bytes_left_for_log_,
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc