Drop 48kHz sample rate support in the APM for ARM architecture

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 94 matching lines @@
       : beamformer(beamformer) {}
   // Accessed internally from capture or during initialization
   std::list<ProcessingComponent*> component_list;
   std::unique_ptr<Beamformer<float>> beamformer;
   std::unique_ptr<AgcManagerDirect> agc_manager;
 };
 
 const int AudioProcessing::kNativeSampleRatesHz[] = {
     AudioProcessing::kSampleRate8kHz,
     AudioProcessing::kSampleRate16kHz,
+#ifdef WEBRTC_ARCH_ARM_FAMILY
+    AudioProcessing::kSampleRate32kHz};

[peah-webrtc, 2016/03/05 10:56:41]

What is the impact of this on current platforms? What happens if APM is actually
being opened in 48 kHz on ARM? I guess that there is an error message?

We need to make sure that 
1) Make sure that this change does not break WebRTC for any currently running
ARM platforms.
2) The error message that is caused by this is properly addressed when received
by a caller of APM.

The next question is what to do instead. On Android, I'm quite sure that we are
called with 48 kHz when AECM is not active. Should we then downsample to 32 kHz
internally?

[aluebs-webrtc, 2016/03/07 11:32:15]

What do you mean by "being opened"? If the APM receives a signal of 48kHz it
would just downsample to 32kHz at the input and upsample it again at the end. So
there is no error message.

[peah-webrtc, 2016/03/07 12:50:48]

I think I see it now. So this change has nothing to do with supported sample
rates, right? The kNativeSampleRatesHz array is only used to determine the
minimum processing rate, right? And has nothing to do with the supported sample
rates, right?

Then this change should be fine as you say.

So what then instead happens is that the 3-bandsplitting is replaced by an
down+upsampling? Which should be cheaper than the 3-bandsplitting, right?

[peah-webrtc, 2016/03/07 12:50:48]

I have another question about this code. Currently it is only used as to set the
processing rate as:

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];

...

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;
    }
  }


Why not change that to (the example excluding your changes in this CL)
const int AudioProcessing::kMaxNativeSampleRateHz =
AudioProcessing::kSampleRate48kHz;
...
const int fwd_proc_rate = AudioProcessing::kMaxNativeSampleRateHz;

[aluebs-webrtc, 2016/03/07 13:02:01]

As you say, this would replace the 3-band splitting by a down- and up-sampling.
Regarding your suggestion, this would force to use 48kHz always, which is not
what we want to do if we get and input (or output) sample rate of, let's say,
16kHz.

[peah-webrtc, 2016/03/10 07:01:00]

Ah, then I see. Then the change is fine.

+#else
     AudioProcessing::kSampleRate32kHz,
     AudioProcessing::kSampleRate48kHz};
+#endif  // WEBRTC_ARCH_ARM_FAMILY
 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);
 }
@@ skipping 1323 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