Allow more than 2 input channels 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.
  */
 
 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
 
 #include <assert.h>
+#include <algorithm>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 }
 #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
@@ skipping 26 matching lines @@
 
 #define RETURN_ON_ERR(expr)  \
   do {                       \
     int err = (expr);        \
     if (err != kNoError) {   \
       return err;            \
     }                        \
   } while (0)
 
 namespace webrtc {
+namespace {
+
+static bool LayoutHasKeyboard(AudioProcessing::ChannelLayout layout) {
+  switch (layout) {
+    case AudioProcessing::kMono:
+    case AudioProcessing::kStereo:
+      return false;
+    case AudioProcessing::kMonoAndKeyboard:
+    case AudioProcessing::kStereoAndKeyboard:
+      return true;
+  }
+
+  assert(false);
+  return false;
+}
+
+}  // namespace
 
 // Throughout webrtc, it's assumed that success is represented by zero.
 static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
 
 // This class has two main functionalities:
 //
 // 1) It is returned instead of the real GainControl after the new AGC has been
 //    enabled in order to prevent an outside user from overriding compression
 //    settings. It doesn't do anything in its implementation, except for
 //    delegating the const methods and Enable calls to the real GainControl, so
 //    AGC can still be disabled.
 //
 // 2) It is injected into AgcManagerDirect and implements volume callbacks for
 //    getting and setting the volume level. It just caches this value to be used
 //    in VoiceEngine later.
 class GainControlForNewAgc : public GainControl, public VolumeCallbacks {
  public:
   explicit GainControlForNewAgc(GainControlImpl* gain_control)
-      : real_gain_control_(gain_control),
-        volume_(0) {
-  }
+      : real_gain_control_(gain_control), volume_(0) {}
 
   // GainControl implementation.
   int Enable(bool enable) override {
     return real_gain_control_->Enable(enable);
   }
   bool is_enabled() const override { return real_gain_control_->is_enabled(); }
   int set_stream_analog_level(int level) override {
     volume_ = level;
     return AudioProcessing::kNoError;
   }
@@ skipping 68 matching lines @@
       gain_control_(NULL),
       high_pass_filter_(NULL),
       level_estimator_(NULL),
       noise_suppression_(NULL),
       voice_detection_(NULL),
       crit_(CriticalSectionWrapper::CreateCriticalSection()),
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
       debug_file_(FileWrapper::Create()),
       event_msg_(new audioproc::Event()),
 #endif
-      fwd_in_format_(kSampleRate16kHz, 1),
+      api_format_({{{kSampleRate16kHz, 1, false},
+                    {kSampleRate16kHz, 1, false},
+                    {kSampleRate16kHz, 1, false}}}),
       fwd_proc_format_(kSampleRate16kHz),
-      fwd_out_format_(kSampleRate16kHz, 1),
-      rev_in_format_(kSampleRate16kHz, 1),
       rev_proc_format_(kSampleRate16kHz, 1),
       split_rate_(kSampleRate16kHz),
       stream_delay_ms_(0),
       delay_offset_ms_(0),
       was_stream_delay_set_(false),
       last_stream_delay_ms_(0),
       last_aec_system_delay_ms_(0),
       stream_delay_jumps_(-1),
       aec_system_delay_jumps_(-1),
       output_will_be_muted_(false),
@@ skipping 63 matching lines @@
   crit_ = NULL;
 }
 
 int AudioProcessingImpl::Initialize() {
   CriticalSectionScoped crit_scoped(crit_);
   return InitializeLocked();
 }
 
 int AudioProcessingImpl::set_sample_rate_hz(int rate) {
   CriticalSectionScoped crit_scoped(crit_);
-  return InitializeLocked(rate,
-                          rate,
-                          rev_in_format_.rate(),
-                          fwd_in_format_.num_channels(),
-                          fwd_out_format_.num_channels(),
-                          rev_in_format_.num_channels());
+
+  ProcessingConfig processing_config = api_format_;
+  processing_config.input_stream().set_sample_rate_hz(rate);
+  processing_config.output_stream().set_sample_rate_hz(rate);
+  return InitializeLocked(processing_config);
 }
 
 int AudioProcessingImpl::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) {
+  const ProcessingConfig processing_config = {
+      {{input_sample_rate_hz, ChannelsFromLayout(input_layout),
+        LayoutHasKeyboard(input_layout)},
+       {output_sample_rate_hz, ChannelsFromLayout(output_layout),
+        LayoutHasKeyboard(output_layout)},
+       {reverse_sample_rate_hz, ChannelsFromLayout(reverse_layout),
+        LayoutHasKeyboard(reverse_layout)}}};
+
+  return Initialize(processing_config);
+}
+
+int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
   CriticalSectionScoped crit_scoped(crit_);
-  return InitializeLocked(input_sample_rate_hz,
-                          output_sample_rate_hz,
-                          reverse_sample_rate_hz,
-                          ChannelsFromLayout(input_layout),
-                          ChannelsFromLayout(output_layout),
-                          ChannelsFromLayout(reverse_layout));
+  return InitializeLocked(processing_config);
 }
 
 int AudioProcessingImpl::InitializeLocked() {
-  const int fwd_audio_buffer_channels = beamformer_enabled_ ?
-                                        fwd_in_format_.num_channels() :
-                                        fwd_out_format_.num_channels();
-  render_audio_.reset(new AudioBuffer(rev_in_format_.samples_per_channel(),
-                                      rev_in_format_.num_channels(),
-                                      rev_proc_format_.samples_per_channel(),
-                                      rev_proc_format_.num_channels(),
-                                      rev_proc_format_.samples_per_channel()));
-  capture_audio_.reset(new AudioBuffer(fwd_in_format_.samples_per_channel(),
-                                       fwd_in_format_.num_channels(),
-                                       fwd_proc_format_.samples_per_channel(),
-                                       fwd_audio_buffer_channels,
-                                       fwd_out_format_.samples_per_channel()));
+  const int fwd_audio_buffer_channels =
+      beamformer_enabled_ ? api_format_.input_stream().num_channels()
+                          : api_format_.output_stream().num_channels();
+  if (api_format_.reverse_stream().num_channels() > 0) {
+    render_audio_.reset(new AudioBuffer(
+        api_format_.reverse_stream().samples_per_channel(),
+        api_format_.reverse_stream().num_channels(),
+        rev_proc_format_.samples_per_channel(), rev_proc_format_.num_channels(),
+        rev_proc_format_.samples_per_channel()));
+  } else {
+    render_audio_.reset(nullptr);
+  }
+  capture_audio_.reset(new AudioBuffer(
+      api_format_.input_stream().samples_per_channel(),
+      api_format_.input_stream().num_channels(),
+      fwd_proc_format_.samples_per_channel(), fwd_audio_buffer_channels,
+      api_format_.output_stream().samples_per_channel()));
 
   // Initialize all components.
   for (auto item : component_list_) {
     int err = item->Initialize();
     if (err != kNoError) {
       return err;
     }
   }
 
   InitializeExperimentalAgc();
 
   InitializeTransient();
 
   InitializeBeamformer();
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     int err = WriteInitMessage();
     if (err != kNoError) {
       return err;
     }
   }
 #endif
 
   return kNoError;
 }
 
-int AudioProcessingImpl::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) {
-  if (input_sample_rate_hz <= 0 ||
-      output_sample_rate_hz <= 0 ||
-      reverse_sample_rate_hz <= 0) {
-    return kBadSampleRateError;
+int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
+  for (const auto& stream : config.streams) {
+    if (stream.sample_rate_hz() < 0) {
+      return kBadSampleRateError;
+    }
+    if (stream.num_channels() < 0) {
+      return kBadNumberChannelsError;
+    }
   }
-  if (num_output_channels > num_input_channels) {
-    return kBadNumberChannelsError;
-  }
-  // Only mono and stereo supported currently.
-  if (num_input_channels > 2 || num_input_channels < 1 ||
-      num_output_channels > 2 || num_output_channels < 1 ||
-      num_reverse_channels > 2 || num_reverse_channels < 1) {
-    return kBadNumberChannelsError;
-  }
-  if (beamformer_enabled_ &&
-      (static_cast<size_t>(num_input_channels) != array_geometry_.size() ||
-       num_output_channels > 1)) {
+
+  const int num_in_channels = config.input_stream().num_channels();
+  const int num_out_channels = config.output_stream().num_channels();
+
+  // Need at least one input channel.
+  // Need either one output channel or as many outputs as there are inputs.
+  if (num_in_channels == 0 ||
+      !(num_out_channels == 1 || num_out_channels == num_in_channels)) {
     return kBadNumberChannelsError;
   }
 
-  fwd_in_format_.set(input_sample_rate_hz, num_input_channels);
-  fwd_out_format_.set(output_sample_rate_hz, num_output_channels);
-  rev_in_format_.set(reverse_sample_rate_hz, num_reverse_channels);
+  if (beamformer_enabled_ &&
+      (static_cast<size_t>(config.input_stream().num_channels()) !=
+           array_geometry_.size() || num_out_channels > 1)) {
+    return kBadNumberChannelsError;
+  }
+
+  api_format_ = config;
 
   // We process at the closest native rate >= min(input rate, output rate)...
-  int min_proc_rate = std::min(fwd_in_format_.rate(), fwd_out_format_.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 (supports_48kHz_ && 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;
   }
   // ...with one exception.
   if (echo_control_mobile_->is_enabled() && min_proc_rate > kSampleRate16kHz) {
     fwd_proc_rate = kSampleRate16kHz;
   }
 
-  fwd_proc_format_.set(fwd_proc_rate);
+  fwd_proc_format_ = StreamConfig(fwd_proc_rate);
 
   // We normally process the reverse stream at 16 kHz. Unless...
   int rev_proc_rate = kSampleRate16kHz;
-  if (fwd_proc_format_.rate() == kSampleRate8kHz) {
+  if (fwd_proc_format_.sample_rate_hz() == kSampleRate8kHz) {
     // ...the forward stream is at 8 kHz.
     rev_proc_rate = kSampleRate8kHz;
   } else {
-    if (rev_in_format_.rate() == kSampleRate32kHz) {
+    if (api_format_.reverse_stream().sample_rate_hz() == kSampleRate32kHz) {
       // ...or the input is at 32 kHz, in which case we use the splitting
       // filter rather than the resampler.
       rev_proc_rate = kSampleRate32kHz;
     }
   }
 
   // Always downmix the reverse stream to mono for analysis. This has been
   // demonstrated to work well for AEC in most practical scenarios.
-  rev_proc_format_.set(rev_proc_rate, 1);
+  rev_proc_format_ = StreamConfig(rev_proc_rate, 1);
 
-  if (fwd_proc_format_.rate() == kSampleRate32kHz ||
-      fwd_proc_format_.rate() == kSampleRate48kHz) {
+  if (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
+      fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz) {
     split_rate_ = kSampleRate16kHz;
   } else {
-    split_rate_ = fwd_proc_format_.rate();
+    split_rate_ = fwd_proc_format_.sample_rate_hz();
   }
 
   return InitializeLocked();
 }
 
 // Calls InitializeLocked() if any of the audio parameters have changed from
 // their current values.
-int AudioProcessingImpl::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) {
-  if (input_sample_rate_hz == fwd_in_format_.rate() &&
-      output_sample_rate_hz == fwd_out_format_.rate() &&
-      reverse_sample_rate_hz == rev_in_format_.rate() &&
-      num_input_channels == fwd_in_format_.num_channels() &&
-      num_output_channels == fwd_out_format_.num_channels() &&
-      num_reverse_channels == rev_in_format_.num_channels()) {
+int AudioProcessingImpl::MaybeInitializeLocked(
+    const ProcessingConfig& processing_config) {
+  if (processing_config == api_format_) {
     return kNoError;
   }
-  return InitializeLocked(input_sample_rate_hz,
-                          output_sample_rate_hz,
-                          reverse_sample_rate_hz,
-                          num_input_channels,
-                          num_output_channels,
-                          num_reverse_channels);
+  return InitializeLocked(processing_config);
 }
 
 void AudioProcessingImpl::SetExtraOptions(const Config& config) {
   CriticalSectionScoped crit_scoped(crit_);
   for (auto item : component_list_) {
     item->SetExtraOptions(config);
   }
 
   if (transient_suppressor_enabled_ != config.Get<ExperimentalNs>().enabled) {
     transient_suppressor_enabled_ = config.Get<ExperimentalNs>().enabled;
     InitializeTransient();
   }
 }
 
 int AudioProcessingImpl::input_sample_rate_hz() const {
   CriticalSectionScoped crit_scoped(crit_);
-  return fwd_in_format_.rate();
+  return api_format_.input_stream().sample_rate_hz();
 }
 
 int AudioProcessingImpl::sample_rate_hz() const {
   CriticalSectionScoped crit_scoped(crit_);
-  return fwd_in_format_.rate();
+  return api_format_.input_stream().sample_rate_hz();
 }
 
 int AudioProcessingImpl::proc_sample_rate_hz() const {
-  return fwd_proc_format_.rate();
+  return fwd_proc_format_.sample_rate_hz();
 }
 
 int AudioProcessingImpl::proc_split_sample_rate_hz() const {
   return split_rate_;
 }
 
 int AudioProcessingImpl::num_reverse_channels() const {
   return rev_proc_format_.num_channels();
 }
 
 int AudioProcessingImpl::num_input_channels() const {
-  return fwd_in_format_.num_channels();
+  return api_format_.input_stream().num_channels();
 }
 
 int AudioProcessingImpl::num_output_channels() const {
-  return fwd_out_format_.num_channels();
+  return api_format_.output_stream().num_channels();
 }
 
 void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
   CriticalSectionScoped lock(crit_);
   output_will_be_muted_ = muted;
   if (agc_manager_.get()) {
     agc_manager_->SetCaptureMuted(output_will_be_muted_);
   }
 }
 
 bool AudioProcessingImpl::output_will_be_muted() const {
   CriticalSectionScoped lock(crit_);
   return output_will_be_muted_;
 }
 
 int AudioProcessingImpl::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) {
+  const ProcessingConfig processing_config = {
+      {{
+           input_sample_rate_hz, ChannelsFromLayout(input_layout),
+           LayoutHasKeyboard(input_layout),
+       },
+       {
+           output_sample_rate_hz, ChannelsFromLayout(output_layout),
+           LayoutHasKeyboard(output_layout),
+       },
+       api_format_.reverse_stream()}};
+
+  if (samples_per_channel !=
+      processing_config.input_stream().samples_per_channel()) {
+    return kBadDataLengthError;
+  }
+  return ProcessStream(src, processing_config, dest);
+}
+
+int AudioProcessingImpl::ProcessStream(
+    const float* const* src,
+    const ProcessingConfig& processing_config,
+    float* const* dest) {
   CriticalSectionScoped crit_scoped(crit_);
   if (!src || !dest) {
     return kNullPointerError;
   }
 
-  RETURN_ON_ERR(MaybeInitializeLocked(input_sample_rate_hz,
-                                      output_sample_rate_hz,
-                                      rev_in_format_.rate(),
-                                      ChannelsFromLayout(input_layout),
-                                      ChannelsFromLayout(output_layout),
-                                      rev_in_format_.num_channels()));
-  if (samples_per_channel != fwd_in_format_.samples_per_channel()) {
-    return kBadDataLengthError;
-  }
+  const int samples_per_channel =
+      processing_config.input_stream().samples_per_channel();
+
+  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
+  assert(samples_per_channel ==
+         api_format_.input_stream().samples_per_channel());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     event_msg_->set_type(audioproc::Event::STREAM);
     audioproc::Stream* msg = event_msg_->mutable_stream();
-    const size_t channel_size =
-        sizeof(float) * fwd_in_format_.samples_per_channel();
-    for (int i = 0; i < fwd_in_format_.num_channels(); ++i)
+    const size_t channel_size = sizeof(float) * samples_per_channel;
+    for (int i = 0; i < api_format_.input_stream().num_channels(); ++i)
       msg->add_input_channel(src[i], channel_size);
   }
 #endif
 
-  capture_audio_->CopyFrom(src, samples_per_channel, input_layout);
+  capture_audio_->CopyFrom(src, api_format_.input_stream());
   RETURN_ON_ERR(ProcessStreamLocked());
-  capture_audio_->CopyTo(fwd_out_format_.samples_per_channel(),
-                         output_layout,
-                         dest);
+  capture_audio_->CopyTo(api_format_.output_stream(), dest);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     audioproc::Stream* msg = event_msg_->mutable_stream();
-    const size_t channel_size =
-        sizeof(float) * fwd_out_format_.samples_per_channel();
-    for (int i = 0; i < fwd_out_format_.num_channels(); ++i)
+    const size_t channel_size = sizeof(float) * samples_per_channel;
+    for (int i = 0; i < api_format_.input_stream().num_channels(); ++i)
       msg->add_output_channel(dest[i], channel_size);
     RETURN_ON_ERR(WriteMessageToDebugFile());
   }
 #endif
 
   return kNoError;
 }
 
 int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
   CriticalSectionScoped crit_scoped(crit_);
   if (!frame) {
     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 (echo_control_mobile_->is_enabled() &&
       frame->sample_rate_hz_ > kSampleRate16kHz) {
     LOG(LS_ERROR) << "AECM only supports 16 or 8 kHz sample rates";
     return kUnsupportedComponentError;
   }
 
   // TODO(ajm): The input and output rates and channels are currently
   // constrained to be identical in the int16 interface.
-  RETURN_ON_ERR(MaybeInitializeLocked(frame->sample_rate_hz_,
-                                      frame->sample_rate_hz_,
-                                      rev_in_format_.rate(),
-                                      frame->num_channels_,
-                                      frame->num_channels_,
-                                      rev_in_format_.num_channels()));
-  if (frame->samples_per_channel_ != fwd_in_format_.samples_per_channel()) {
+  ProcessingConfig processing_config = api_format_;
+  processing_config.input_stream().set_sample_rate_hz(frame->sample_rate_hz_);
+  processing_config.input_stream().set_num_channels(frame->num_channels_);
+  processing_config.output_stream().set_sample_rate_hz(frame->sample_rate_hz_);
+  processing_config.output_stream().set_num_channels(frame->num_channels_);
+
+  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
+  if (frame->samples_per_channel_ !=
+      api_format_.input_stream().samples_per_channel()) {
     return kBadDataLengthError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     event_msg_->set_type(audioproc::Event::STREAM);
     audioproc::Stream* msg = event_msg_->mutable_stream();
-    const size_t data_size = sizeof(int16_t) *
-                             frame->samples_per_channel_ *
-                             frame->num_channels_;
+    const size_t data_size =
+        sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_input_data(frame->data_, data_size);
   }
 #endif
 
   capture_audio_->DeinterleaveFrom(frame);
   RETURN_ON_ERR(ProcessStreamLocked());
   capture_audio_->InterleaveTo(frame, output_copy_needed(is_data_processed()));
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     audioproc::Stream* msg = event_msg_->mutable_stream();
-    const size_t data_size = sizeof(int16_t) *
-                             frame->samples_per_channel_ *
-                             frame->num_channels_;
+    const size_t data_size =
+        sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_output_data(frame->data_, data_size);
     RETURN_ON_ERR(WriteMessageToDebugFile());
   }
 #endif
 
   return kNoError;
 }
 
-
 int AudioProcessingImpl::ProcessStreamLocked() {
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     audioproc::Stream* msg = event_msg_->mutable_stream();
     msg->set_delay(stream_delay_ms_);
     msg->set_drift(echo_cancellation_->stream_drift_samples());
     msg->set_level(gain_control()->stream_analog_level());
     msg->set_keypress(key_pressed_);
   }
 #endif
 
   MaybeUpdateHistograms();
 
   AudioBuffer* ca = capture_audio_.get();  // For brevity.
   if (use_new_agc_ && gain_control_->is_enabled()) {
-    agc_manager_->AnalyzePreProcess(ca->channels()[0],
-                                    ca->num_channels(),
+    agc_manager_->AnalyzePreProcess(ca->channels()[0], ca->num_channels(),
                                     fwd_proc_format_.samples_per_channel());
   }
 
   bool data_processed = is_data_processed();
   if (analysis_needed(data_processed)) {
     ca->SplitIntoFrequencyBands();
   }
 
   if (beamformer_enabled_) {
     beamformer_->ProcessChunk(*ca->split_data_f(), ca->split_data_f());
     ca->set_num_channels(1);
   }
 
   RETURN_ON_ERR(high_pass_filter_->ProcessCaptureAudio(ca));
   RETURN_ON_ERR(gain_control_->AnalyzeCaptureAudio(ca));
   RETURN_ON_ERR(noise_suppression_->AnalyzeCaptureAudio(ca));
   RETURN_ON_ERR(echo_cancellation_->ProcessCaptureAudio(ca));
 
   if (echo_control_mobile_->is_enabled() && noise_suppression_->is_enabled()) {
     ca->CopyLowPassToReference();
   }
   RETURN_ON_ERR(noise_suppression_->ProcessCaptureAudio(ca));
   RETURN_ON_ERR(echo_control_mobile_->ProcessCaptureAudio(ca));
   RETURN_ON_ERR(voice_detection_->ProcessCaptureAudio(ca));
 
-  if (use_new_agc_ &&
-      gain_control_->is_enabled() &&
+  if (use_new_agc_ && gain_control_->is_enabled() &&
       (!beamformer_enabled_ || beamformer_->is_target_present())) {
     agc_manager_->Process(ca->split_bands_const(0)[kBand0To8kHz],
-                          ca->num_frames_per_band(),
-                          split_rate_);
+                          ca->num_frames_per_band(), split_rate_);
   }
   RETURN_ON_ERR(gain_control_->ProcessCaptureAudio(ca));
 
   if (synthesis_needed(data_processed)) {
     ca->MergeFrequencyBands();
   }
 
   // TODO(aluebs): Investigate if the transient suppression placement should be
   // before or after the AGC.
   if (transient_suppressor_enabled_) {
     float voice_probability =
         agc_manager_.get() ? agc_manager_->voice_probability() : 1.f;
 
-    transient_suppressor_->Suppress(ca->channels_f()[0],

[mgraczyk, 2015/07/10 00:33:36]

This is just my clang_format disagreeing with yours. Should not be changed.

-                                    ca->num_frames(),
-                                    ca->num_channels(),
-                                    ca->split_bands_const_f(0)[kBand0To8kHz],
-                                    ca->num_frames_per_band(),
-                                    ca->keyboard_data(),
-                                    ca->num_keyboard_frames(),
-                                    voice_probability,
-                                    key_pressed_);
+    transient_suppressor_->Suppress(
+        ca->channels_f()[0], ca->num_frames(), ca->num_channels(),
+        ca->split_bands_const_f(0)[kBand0To8kHz], ca->num_frames_per_band(),
+        ca->keyboard_data(), ca->num_keyboard_frames(), voice_probability,
+        key_pressed_);
   }
 
   // The level estimator operates on the recombined data.
   RETURN_ON_ERR(level_estimator_->ProcessStream(ca));
 
   was_stream_delay_set_ = false;
   return kNoError;
 }
 
 int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
                                               int samples_per_channel,
                                               int sample_rate_hz,
                                               ChannelLayout layout) {
+  const StreamConfig reverse_config = {
+      sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout),
+  };
+  return AnalyzeReverseStream(data, reverse_config);
+}
+
+int AudioProcessingImpl::AnalyzeReverseStream(
+    const float* const* data,
+    const StreamConfig& reverse_config) {
   CriticalSectionScoped crit_scoped(crit_);
   if (data == NULL) {
     return kNullPointerError;
   }
 
-  const int num_channels = ChannelsFromLayout(layout);
-  RETURN_ON_ERR(MaybeInitializeLocked(fwd_in_format_.rate(),
-                                      fwd_out_format_.rate(),
-                                      sample_rate_hz,
-                                      fwd_in_format_.num_channels(),
-                                      fwd_out_format_.num_channels(),
-                                      num_channels));
-  if (samples_per_channel != rev_in_format_.samples_per_channel()) {
+  if (reverse_config.num_channels() <= 0) {
+    return kBadNumberChannelsError;
+  }
+
+  const ProcessingConfig processing_config = {{api_format_.input_stream(),
+                                               api_format_.output_stream(),
+                                               reverse_config}};
+  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
+  if (reverse_config.samples_per_channel() !=
+      api_format_.reverse_stream().samples_per_channel()) {
     return kBadDataLengthError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
     audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
     const size_t channel_size =
-        sizeof(float) * rev_in_format_.samples_per_channel();
-    for (int i = 0; i < num_channels; ++i)
+        sizeof(float) * api_format_.reverse_stream().samples_per_channel();
+    for (int i = 0; i < api_format_.reverse_stream().num_channels(); ++i)
       msg->add_channel(data[i], channel_size);
     RETURN_ON_ERR(WriteMessageToDebugFile());
   }
 #endif
 
-  render_audio_->CopyFrom(data, samples_per_channel, layout);
+  render_audio_->CopyFrom(data, api_format_.reverse_stream());
   return AnalyzeReverseStreamLocked();
 }
 
 int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
   CriticalSectionScoped crit_scoped(crit_);
   if (frame == NULL) {
     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;
   }
   // This interface does not tolerate different forward and reverse rates.
-  if (frame->sample_rate_hz_ != fwd_in_format_.rate()) {
+  if (frame->sample_rate_hz_ != api_format_.input_stream().sample_rate_hz()) {
     return kBadSampleRateError;
   }
 
-  RETURN_ON_ERR(MaybeInitializeLocked(fwd_in_format_.rate(),
-                                      fwd_out_format_.rate(),
-                                      frame->sample_rate_hz_,
-                                      fwd_in_format_.num_channels(),
-                                      fwd_in_format_.num_channels(),
-                                      frame->num_channels_));
-  if (frame->samples_per_channel_ != rev_in_format_.samples_per_channel()) {
+  if (frame->num_channels_ <= 0) {
+    return kBadNumberChannelsError;
+  }
+
+  const ProcessingConfig processing_config = {{
+      api_format_.input_stream(),
+      api_format_.output_stream(),
+      {
+          frame->sample_rate_hz_, frame->num_channels_,
+          api_format_.reverse_stream().has_keyboard(),
+      },
+  }};
+  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
+  if (frame->samples_per_channel_ !=
+      api_format_.reverse_stream().samples_per_channel()) {
     return kBadDataLengthError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
     audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
-    const size_t data_size = sizeof(int16_t) *
-                             frame->samples_per_channel_ *
-                             frame->num_channels_;
+    const size_t data_size =
+        sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_data(frame->data_, data_size);
     RETURN_ON_ERR(WriteMessageToDebugFile());
   }
 #endif
 
   render_audio_->DeinterleaveFrom(frame);
   return AnalyzeReverseStreamLocked();
 }
 
 int AudioProcessingImpl::AnalyzeReverseStreamLocked() {
   AudioBuffer* ra = render_audio_.get();  // For brevity.
-  if (rev_proc_format_.rate() == kSampleRate32kHz) {
+  if (rev_proc_format_.sample_rate_hz() == kSampleRate32kHz) {
     ra->SplitIntoFrequencyBands();
   }
 
   RETURN_ON_ERR(echo_cancellation_->ProcessRenderAudio(ra));
   RETURN_ON_ERR(echo_control_mobile_->ProcessRenderAudio(ra));
   if (!use_new_agc_) {
     RETURN_ON_ERR(gain_control_->ProcessRenderAudio(ra));
   }
 
   return kNoError;
@@ skipping 181 matching lines @@
   } else if (enabled_count == 2) {
     if (level_estimator_->is_enabled() && voice_detection_->is_enabled()) {
       return false;
     }
   }
   return true;
 }
 
 bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
   // Check if we've upmixed or downmixed the audio.
-  return ((fwd_out_format_.num_channels() != fwd_in_format_.num_channels()) ||
+  return ((api_format_.output_stream().num_channels() !=
+           api_format_.input_stream().num_channels()) ||
           is_data_processed || transient_suppressor_enabled_);
 }
 
 bool AudioProcessingImpl::synthesis_needed(bool is_data_processed) const {
-  return (is_data_processed && (fwd_proc_format_.rate() == kSampleRate32kHz ||
-          fwd_proc_format_.rate() == kSampleRate48kHz));
+  return (is_data_processed &&
+          (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
+           fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz));
 }
 
 bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const {
   if (!is_data_processed && !voice_detection_->is_enabled() &&
       !transient_suppressor_enabled_) {
     // Only level_estimator_ is enabled.
     return false;
-  } else if (fwd_proc_format_.rate() == kSampleRate32kHz ||
-             fwd_proc_format_.rate() == kSampleRate48kHz) {
+  } else if (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
+             fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz) {
     // Something besides level_estimator_ is enabled, and we have super-wb.
     return true;
   }
   return false;
 }
 
 void AudioProcessingImpl::InitializeExperimentalAgc() {
   if (use_new_agc_) {
     if (!agc_manager_.get()) {
       agc_manager_.reset(new AgcManagerDirect(gain_control_,
                                               gain_control_for_new_agc_.get(),
                                               agc_startup_min_volume_));
     }
     agc_manager_->Initialize();
     agc_manager_->SetCaptureMuted(output_will_be_muted_);
   }
 }
 
 void AudioProcessingImpl::InitializeTransient() {
   if (transient_suppressor_enabled_) {
     if (!transient_suppressor_.get()) {
       transient_suppressor_.reset(new TransientSuppressor());
     }
-    transient_suppressor_->Initialize(fwd_proc_format_.rate(),
-                                      split_rate_,
-                                      fwd_out_format_.num_channels());
+    transient_suppressor_->Initialize(
+        fwd_proc_format_.sample_rate_hz(), split_rate_,
+        api_format_.output_stream().num_channels());
   }
 }
 
 void AudioProcessingImpl::InitializeBeamformer() {
   if (beamformer_enabled_) {
     if (!beamformer_) {
       beamformer_.reset(new NonlinearBeamformer(array_geometry_));
     }
     beamformer_->Initialize(kChunkSizeMs, split_rate_);
   }
@@ skipping 22 matching lines @@
         stream_delay_jumps_ = 0;  // Activate counter if needed.
       }
       stream_delay_jumps_++;
     }
     last_stream_delay_ms_ = stream_delay_ms_;
 
     // Detect a jump in AEC system delay and log the difference.
     const int frames_per_ms = rtc::CheckedDivExact(split_rate_, 1000);
     const int aec_system_delay_ms =
         WebRtcAec_system_delay(echo_cancellation()->aec_core()) / frames_per_ms;
-    const int diff_aec_system_delay_ms = aec_system_delay_ms -
-        last_aec_system_delay_ms_;
+    const int diff_aec_system_delay_ms =
+        aec_system_delay_ms - last_aec_system_delay_ms_;
     if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
         last_aec_system_delay_ms_ != 0) {
       RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecSystemDelayJump",
                            diff_aec_system_delay_ms, kMinDiffDelayMs, 1000,
                            100);
       if (aec_system_delay_jumps_ == -1) {
         aec_system_delay_jumps_ = 0;  // Activate counter if needed.
       }
       aec_system_delay_jumps_++;
     }
@@ skipping 18 matching lines @@
   aec_system_delay_jumps_ = -1;
   last_aec_system_delay_ms_ = 0;
 }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 int AudioProcessingImpl::WriteMessageToDebugFile() {
   int32_t size = event_msg_->ByteSize();
   if (size <= 0) {
     return kUnspecifiedError;
   }
 #if defined(WEBRTC_ARCH_BIG_ENDIAN)

[mgraczyk, 2015/07/10 00:33:36]

More clang_format here and above.

-  // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
-  //            pretty safe in assuming little-endian.
+// TODO(ajm): Use little-endian "on the wire". For the moment, we can be
+//            pretty safe in assuming little-endian.
 #endif
 
   if (!event_msg_->SerializeToString(&event_str_)) {
     return kUnspecifiedError;
   }
 
   // Write message preceded by its size.
   if (!debug_file_->Write(&size, sizeof(int32_t))) {
     return kFileError;
   }
   if (!debug_file_->Write(event_str_.data(), event_str_.length())) {
     return kFileError;
   }
 
   event_msg_->Clear();
 
   return kNoError;
 }
 
 int AudioProcessingImpl::WriteInitMessage() {
   event_msg_->set_type(audioproc::Event::INIT);
   audioproc::Init* msg = event_msg_->mutable_init();
-  msg->set_sample_rate(fwd_in_format_.rate());
-  msg->set_num_input_channels(fwd_in_format_.num_channels());
-  msg->set_num_output_channels(fwd_out_format_.num_channels());
-  msg->set_num_reverse_channels(rev_in_format_.num_channels());
-  msg->set_reverse_sample_rate(rev_in_format_.rate());
-  msg->set_output_sample_rate(fwd_out_format_.rate());
+  msg->set_sample_rate(api_format_.input_stream().sample_rate_hz());
+  msg->set_num_input_channels(api_format_.input_stream().num_channels());
+  msg->set_num_output_channels(api_format_.output_stream().num_channels());
+  msg->set_num_reverse_channels(api_format_.reverse_stream().num_channels());
+  msg->set_reverse_sample_rate(api_format_.reverse_stream().sample_rate_hz());
+  msg->set_output_sample_rate(api_format_.output_stream().sample_rate_hz());
 
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc