Revert of Allow more than 2 input channels 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 
 #include <list>
 #include <string>
-#include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace webrtc {
 
 class AgcManagerDirect;
 class AudioBuffer;
 
@@ skipping 14 matching lines @@
 class VoiceDetectionImpl;
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 namespace audioproc {
 
 class Event;
 
 }  // namespace audioproc
 #endif
 
+class AudioRate {
+ public:
+  explicit AudioRate(int sample_rate_hz) { set(sample_rate_hz); }
+  virtual ~AudioRate() {}
+
+  void set(int rate) {
+    rate_ = rate;
+    samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000;
+  }
+
+  int rate() const { return rate_; }
+  int samples_per_channel() const { return samples_per_channel_; }
+
+ private:
+  int rate_;
+  int samples_per_channel_;
+};
+
+class AudioFormat : public AudioRate {
+ public:
+  AudioFormat(int sample_rate_hz, int num_channels)
+      : AudioRate(sample_rate_hz),
+        num_channels_(num_channels) {}
+  virtual ~AudioFormat() {}
+
+  void set(int rate, int num_channels) {
+    AudioRate::set(rate);
+    num_channels_ = num_channels;
+  }
+
+  int num_channels() const { return num_channels_; }
+
+ private:
+  int num_channels_;
+};
+
 class AudioProcessingImpl : public AudioProcessing {
  public:
   explicit AudioProcessingImpl(const Config& config);
 
   // AudioProcessingImpl takes ownership of beamformer.
   AudioProcessingImpl(const Config& config, Beamformer<float>* beamformer);
   virtual ~AudioProcessingImpl();
 
   // AudioProcessing methods.
   int Initialize() override;
   int Initialize(int input_sample_rate_hz,
                  int output_sample_rate_hz,
                  int reverse_sample_rate_hz,
                  ChannelLayout input_layout,
                  ChannelLayout output_layout,
                  ChannelLayout reverse_layout) override;
-  int Initialize(const ProcessingConfig& processing_config) override;
   void SetExtraOptions(const Config& config) override;
   int set_sample_rate_hz(int rate) override;
   int input_sample_rate_hz() const override;
   int sample_rate_hz() const override;
   int proc_sample_rate_hz() const override;
   int proc_split_sample_rate_hz() const override;
   int num_input_channels() const override;
   int num_output_channels() const override;
   int num_reverse_channels() const override;
   void set_output_will_be_muted(bool muted) override;
   bool output_will_be_muted() const override;
   int ProcessStream(AudioFrame* frame) override;
   int ProcessStream(const float* const* src,
                     int samples_per_channel,
                     int input_sample_rate_hz,
                     ChannelLayout input_layout,
                     int output_sample_rate_hz,
                     ChannelLayout output_layout,
                     float* const* dest) override;
-  int ProcessStream(const float* const* src,
-                    const StreamConfig& input_config,
-                    const StreamConfig& output_config,
-                    float* const* dest) override;
   int AnalyzeReverseStream(AudioFrame* frame) override;
   int AnalyzeReverseStream(const float* const* data,
                            int samples_per_channel,
                            int sample_rate_hz,
                            ChannelLayout layout) override;
-  int AnalyzeReverseStream(const float* const* data,
-                           const StreamConfig& reverse_config) override;
   int set_stream_delay_ms(int delay) override;
   int stream_delay_ms() const override;
   bool was_stream_delay_set() const override;
   void set_delay_offset_ms(int offset) override;
   int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
   bool stream_key_pressed() const override;
   int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
   int StartDebugRecording(FILE* handle) override;
   int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
   int StopDebugRecording() override;
   void UpdateHistogramsOnCallEnd() override;
   EchoCancellation* echo_cancellation() const override;
   EchoControlMobile* echo_control_mobile() const override;
   GainControl* gain_control() const override;
   HighPassFilter* high_pass_filter() const override;
   LevelEstimator* level_estimator() const override;
   NoiseSuppression* noise_suppression() const override;
   VoiceDetection* voice_detection() const override;
 
  protected:
   // Overridden in a mock.
   virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
  private:
-  int InitializeLocked(const ProcessingConfig& config)
+  int InitializeLocked(int input_sample_rate_hz,
+                       int output_sample_rate_hz,
+                       int reverse_sample_rate_hz,
+                       int num_input_channels,
+                       int num_output_channels,
+                       int num_reverse_channels)
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  int MaybeInitializeLocked(const ProcessingConfig& config)
+  int MaybeInitializeLocked(int input_sample_rate_hz,
+                            int output_sample_rate_hz,
+                            int reverse_sample_rate_hz,
+                            int num_input_channels,
+                            int num_output_channels,
+                            int num_reverse_channels)
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
   int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   int AnalyzeReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   bool is_data_processed() const;
   bool output_copy_needed(bool is_data_processed) const;
   bool synthesis_needed(bool is_data_processed) const;
   bool analysis_needed(bool is_data_processed) const;
   void InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   void InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
@@ skipping 16 matching lines @@
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // TODO(andrew): make this more graceful. Ideally we would split this stuff
   // out into a separate class with an "enabled" and "disabled" implementation.
   int WriteMessageToDebugFile();
   int WriteInitMessage();
   rtc::scoped_ptr<FileWrapper> debug_file_;
   rtc::scoped_ptr<audioproc::Event> event_msg_;  // Protobuf message.
   std::string event_str_;  // Memory for protobuf serialization.
 #endif
 
-  // Format of processing streams at input/output call sites.
-  ProcessingConfig api_format_;
-
-  // 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_;
-  StreamConfig rev_proc_format_;
+  AudioFormat fwd_in_format_;
+  // This one is an AudioRate, because the forward processing number of channels
+  // is mutable and is tracked by the capture_audio_.
+  AudioRate fwd_proc_format_;
+  AudioFormat fwd_out_format_;
+  AudioFormat rev_in_format_;
+  AudioFormat rev_proc_format_;
   int split_rate_;
 
   int stream_delay_ms_;
   int delay_offset_ms_;
   bool was_stream_delay_set_;
   int last_stream_delay_ms_;
   int last_aec_system_delay_ms_;
   int stream_delay_jumps_;
   int aec_system_delay_jumps_;
 
@@ skipping 11 matching lines @@
   const bool beamformer_enabled_;
   rtc::scoped_ptr<Beamformer<float>> beamformer_;
   const std::vector<Point> array_geometry_;
 
   const bool supports_48kHz_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_