Remove additional channel constraints when Beamforming is enabled in AudioProcessing

webrtc/modules/audio_processing/audio_processing_impl.h

 /*
  *  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 84 matching lines @@
                            const StreamConfig& reverse_input_config,
                            const StreamConfig& reverse_output_config,
                            float* const* dest) override;
 
   // Methods only accessed from APM submodules or
   // from AudioProcessing tests in a single-threaded manner.
   // Hence there is no need for locks in these.
   int proc_sample_rate_hz() const override;
   int proc_split_sample_rate_hz() const override;
   int num_input_channels() const override;
+  int num_proc_channels() const override;
   int num_output_channels() const override;
   int num_reverse_channels() const override;
   int stream_delay_ms() const override;
   bool was_stream_delay_set() const override
       EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
   // Methods returning pointers to APM submodules.
   // No locks are aquired in those, as those locks
   // would offer no protection (the submodules are
   // created only once in a single-treaded manner
@@ skipping 158 matching lines @@
           agc_startup_min_volume(agc_startup_min_volume),
           use_new_agc(use_new_agc),
           intelligibility_enabled(intelligibility_enabled) {}
     int agc_startup_min_volume;
     bool use_new_agc;
     bool intelligibility_enabled;
   } constants_;
 
   struct ApmCaptureState {
     ApmCaptureState(bool transient_suppressor_enabled,
-                    bool beamformer_enabled,
                     const std::vector<Point>& array_geometry,
                     SphericalPointf target_direction)
         : aec_system_delay_jumps(-1),
           delay_offset_ms(0),
           was_stream_delay_set(false),
           last_stream_delay_ms(0),
           last_aec_system_delay_ms(0),
           stream_delay_jumps(-1),
           output_will_be_muted(false),
           key_pressed(false),
           transient_suppressor_enabled(transient_suppressor_enabled),
-          beamformer_enabled(beamformer_enabled),
           array_geometry(array_geometry),
           target_direction(target_direction),
           fwd_proc_format(kSampleRate16kHz),
           split_rate(kSampleRate16kHz) {}
     int aec_system_delay_jumps;
     int delay_offset_ms;
     bool was_stream_delay_set;
     int last_stream_delay_ms;
     int last_aec_system_delay_ms;
     int stream_delay_jumps;
     bool output_will_be_muted;
     bool key_pressed;
     bool transient_suppressor_enabled;
-    bool beamformer_enabled;
     std::vector<Point> array_geometry;
     SphericalPointf target_direction;
     rtc::scoped_ptr<AudioBuffer> capture_audio;
     // Only the rate and samples fields of fwd_proc_format_ are used because the
     // forward processing number of channels is mutable and is tracked by the
     // capture_audio_.
     StreamConfig fwd_proc_format;
     int split_rate;
   } capture_ GUARDED_BY(crit_capture_);
 
   struct ApmCaptureNonLockedState {
-    ApmCaptureNonLockedState()
+    ApmCaptureNonLockedState(bool beamformer_enabled)
         : fwd_proc_format(kSampleRate16kHz),
           split_rate(kSampleRate16kHz),
-          stream_delay_ms(0) {}
+          stream_delay_ms(0),
+          beamformer_enabled(beamformer_enabled) {}
     // Only the rate and samples fields of fwd_proc_format_ are used because the
     // forward processing number of channels is mutable and is tracked by the
     // capture_audio_.
     StreamConfig fwd_proc_format;
     int split_rate;
     int stream_delay_ms;
+    bool beamformer_enabled;
   } capture_nonlocked_;
 
   struct ApmRenderState {
     rtc::scoped_ptr<AudioConverter> render_converter;
     rtc::scoped_ptr<AudioBuffer> render_audio;
   } render_ GUARDED_BY(crit_render_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_