Integrate Intelligibility with APM

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/audio_converter.h"
+#include "webrtc/common_audio/channel_buffer.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"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
 #include "webrtc/modules/audio_processing/common.h"
 #include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
 #include "webrtc/modules/audio_processing/echo_control_mobile_impl.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
 #include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
+#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
 #include "webrtc/modules/audio_processing/level_estimator_impl.h"
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 #include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/file_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/metrics.h"
@@ skipping 135 matching lines @@
       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
       api_format_({{{kSampleRate16kHz, 1, false},
                     {kSampleRate16kHz, 1, false},
+                    {kSampleRate16kHz, 1, false},
                     {kSampleRate16kHz, 1, false}}}),
       fwd_proc_format_(kSampleRate16kHz),
       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),
       key_pressed_(false),
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
       use_new_agc_(false),
 #else
       use_new_agc_(config.Get<ExperimentalAgc>().enabled),
 #endif
       agc_startup_min_volume_(config.Get<ExperimentalAgc>().startup_min_volume),
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
       transient_suppressor_enabled_(false),
 #else
       transient_suppressor_enabled_(config.Get<ExperimentalNs>().enabled),
 #endif
       beamformer_enabled_(config.Get<Beamforming>().enabled),
       beamformer_(beamformer),
-      array_geometry_(config.Get<Beamforming>().array_geometry) {
+      array_geometry_(config.Get<Beamforming>().array_geometry),
+      intelligibility_enabled_(config.Get<Intelligibility>().enabled) {
   echo_cancellation_ = new EchoCancellationImpl(this, crit_);
   component_list_.push_back(echo_cancellation_);
 
   echo_control_mobile_ = new EchoControlMobileImpl(this, crit_);
   component_list_.push_back(echo_control_mobile_);
 
   gain_control_ = new GainControlImpl(this, crit_);
   component_list_.push_back(gain_control_);
 
   high_pass_filter_ = new HighPassFilterImpl(this, crit_);
@@ skipping 51 matching lines @@
   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),
+      {{input_sample_rate_hz,
+        ChannelsFromLayout(input_layout),
         LayoutHasKeyboard(input_layout)},
-       {output_sample_rate_hz, ChannelsFromLayout(output_layout),
+       {output_sample_rate_hz,
+        ChannelsFromLayout(output_layout),
         LayoutHasKeyboard(output_layout)},
-       {reverse_sample_rate_hz, ChannelsFromLayout(reverse_layout),
+       {reverse_sample_rate_hz,
+        ChannelsFromLayout(reverse_layout),
+        LayoutHasKeyboard(reverse_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(processing_config);
 }
 
 int AudioProcessingImpl::InitializeLocked() {
   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) {
+  const int rev_audio_buffer_out_num_frames =
+      api_format_.reverse_output_stream().num_frames() == 0
+          ? rev_proc_format_.num_frames()
+          : api_format_.reverse_output_stream().num_frames();
+  if (api_format_.reverse_input_stream().num_channels() > 0) {
     render_audio_.reset(new AudioBuffer(
-        api_format_.reverse_stream().num_frames(),
-        api_format_.reverse_stream().num_channels(),
+        api_format_.reverse_input_stream().num_frames(),
+        api_format_.reverse_input_stream().num_channels(),
         rev_proc_format_.num_frames(), rev_proc_format_.num_channels(),
-        rev_proc_format_.num_frames()));
+        rev_audio_buffer_out_num_frames));
+    if (rev_conversion_needed()) {
+      render_converter_ = AudioConverter::Create(
+          api_format_.reverse_input_stream().num_channels(),
+          api_format_.reverse_input_stream().num_frames(),
+          api_format_.reverse_output_stream().num_channels(),
+          api_format_.reverse_output_stream().num_frames());
+    } else {
+      render_converter_.reset(nullptr);
+    }
   } else {
     render_audio_.reset(nullptr);
+    render_converter_.reset(nullptr);
   }
   capture_audio_.reset(new AudioBuffer(
       api_format_.input_stream().num_frames(),
       api_format_.input_stream().num_channels(), fwd_proc_format_.num_frames(),
       fwd_audio_buffer_channels, api_format_.output_stream().num_frames()));
 
   // Initialize all components.
   for (auto item : component_list_) {
     int err = item->Initialize();
     if (err != kNoError) {
       return err;
     }
   }
 
   InitializeExperimentalAgc();
 
   InitializeTransient();
 
   InitializeBeamformer();
 
+  InitializeIntelligibility();
+
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     int err = WriteInitMessage();
     if (err != kNoError) {
       return err;
     }
   }
 #endif
 
   return kNoError;
@@ skipping 47 matching lines @@
   }
 
   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_.sample_rate_hz() == kSampleRate8kHz) {
     // ...the forward stream is at 8 kHz.
     rev_proc_rate = kSampleRate8kHz;
   } else {
-    if (api_format_.reverse_stream().sample_rate_hz() == kSampleRate32kHz) {
+    if (api_format_.reverse_input_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_ = StreamConfig(rev_proc_rate, 1);
 
@@ skipping 206 matching lines @@
     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(),
                                     fwd_proc_format_.num_frames());
   }
 
   bool data_processed = is_data_processed();
   if (analysis_needed(data_processed)) {
     ca->SplitIntoFrequencyBands();
   }
 
+  if (intelligibility_enabled_) {
+    intelligibility_enhancer_->AnalyzeCaptureAudio(
+        ca->split_channels_f(kBand0To8kHz), split_rate_, ca->num_channels());
+  }
+
   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));
 
@@ skipping 30 matching lines @@
 
   // 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,
+                                              int rev_sample_rate_hz,
                                               ChannelLayout layout) {
   const StreamConfig reverse_config = {
-      sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout),
+      rev_sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout),
   };
   if (samples_per_channel != reverse_config.num_frames()) {
     return kBadDataLengthError;
   }
-  return AnalyzeReverseStream(data, reverse_config);
+  return AnalyzeReverseStream(data, reverse_config, reverse_config, data);
+}
+
+int AudioProcessingImpl::ProcessReverseStream(
+    const float* const* src,
+    const StreamConfig& reverse_input_config,
+    const StreamConfig& reverse_output_config,
+    float* const* dest) {
+  RETURN_ON_ERR(AnalyzeReverseStream(src, reverse_input_config,
+                                     reverse_output_config, dest));
+  if (is_rev_processed()) {
+    render_audio_->CopyTo(api_format_.reverse_output_stream(), dest);
+  } else if (rev_conversion_needed()) {
+    render_converter_->Convert(src, reverse_input_config.num_frames(),
+                               dest, reverse_output_config.num_frames());
+  }
+
+  return kNoError;
 }
 
 int AudioProcessingImpl::AnalyzeReverseStream(
-    const float* const* data,
-    const StreamConfig& reverse_config) {
+    const float* const* src,
+    const StreamConfig& reverse_input_config,
+    const StreamConfig& reverse_output_config,
+    const float* const* dest) {
   CriticalSectionScoped crit_scoped(crit_);
-  if (data == NULL) {
+  if (src == NULL) {
     return kNullPointerError;
   }
 
-  if (reverse_config.num_channels() <= 0) {
+  if (reverse_input_config.num_channels() <= 0) {
     return kBadNumberChannelsError;
   }
 
   ProcessingConfig processing_config = api_format_;
-  processing_config.reverse_stream() = reverse_config;
+  processing_config.reverse_input_stream() = reverse_input_config;
+  processing_config.reverse_output_stream() = reverse_output_config;
 
   RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
-  assert(reverse_config.num_frames() ==
-         api_format_.reverse_stream().num_frames());
+  assert(reverse_input_config.num_frames() ==
+         api_format_.reverse_input_stream().num_frames());
 
 #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) * api_format_.reverse_stream().num_frames();
-    for (int i = 0; i < api_format_.reverse_stream().num_channels(); ++i)
-      msg->add_channel(data[i], channel_size);
+        sizeof(float) * api_format_.reverse_input_stream().num_frames();
+    for (int i = 0; i < api_format_.reverse_input_stream().num_channels(); ++i)
+      msg->add_channel(src[i], channel_size);
     RETURN_ON_ERR(WriteMessageToDebugFile());
   }
 #endif
 
-  render_audio_->CopyFrom(data, api_format_.reverse_stream());
-  return AnalyzeReverseStreamLocked();
+  render_audio_->CopyFrom(src, api_format_.reverse_input_stream());
+  return ProcessReverseStreamLocked();
+}
+
+int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
+  RETURN_ON_ERR(AnalyzeReverseStream(frame));
+  if (is_rev_processed()) {
+    render_audio_->InterleaveTo(frame, true);
+  }
+
+  return kNoError;
 }
 
 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_ != api_format_.input_stream().sample_rate_hz()) {
     return kBadSampleRateError;
   }
 
   if (frame->num_channels_ <= 0) {
     return kBadNumberChannelsError;
   }
 
   ProcessingConfig processing_config = api_format_;
-  processing_config.reverse_stream().set_sample_rate_hz(frame->sample_rate_hz_);
-  processing_config.reverse_stream().set_num_channels(frame->num_channels_);
+  processing_config.reverse_input_stream().set_sample_rate_hz(
+      frame->sample_rate_hz_);
+  processing_config.reverse_input_stream().set_num_channels(
+      frame->num_channels_);
+  processing_config.reverse_output_stream().set_sample_rate_hz(
+      frame->sample_rate_hz_);
+  processing_config.reverse_output_stream().set_num_channels(
+      frame->num_channels_);
 
   RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
   if (frame->samples_per_channel_ !=
-      api_format_.reverse_stream().num_frames()) {
+      api_format_.reverse_input_stream().num_frames()) {
     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_;
     msg->set_data(frame->data_, data_size);
     RETURN_ON_ERR(WriteMessageToDebugFile());
   }
 #endif
-
   render_audio_->DeinterleaveFrom(frame);
-  return AnalyzeReverseStreamLocked();
+  return ProcessReverseStreamLocked();
 }
 
-int AudioProcessingImpl::AnalyzeReverseStreamLocked() {
+int AudioProcessingImpl::ProcessReverseStreamLocked() {
   AudioBuffer* ra = render_audio_.get();  // For brevity.
   if (rev_proc_format_.sample_rate_hz() == kSampleRate32kHz) {
     ra->SplitIntoFrequencyBands();
   }
 
+  if (intelligibility_enabled_) {
+    intelligibility_enhancer_->ProcessRenderAudio(
+        ra->split_channels_f(kBand0To8kHz), split_rate_, ra->num_channels());
+  }
+
   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));
   }
 
+  if (rev_proc_format_.sample_rate_hz() == kSampleRate32kHz &&
+      is_rev_processed()) {
+    ra->MergeFrequencyBands();
+  }
+
   return kNoError;
 }
 
 int AudioProcessingImpl::set_stream_delay_ms(int delay) {
   Error retval = kNoError;
   was_stream_delay_set_ = true;
   delay += delay_offset_ms_;
 
   if (delay < 0) {
     delay = 0;
@@ skipping 196 matching lines @@
     // Only level_estimator_ is enabled.
     return false;
   } 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;
 }
 
+bool AudioProcessingImpl::is_rev_processed() const {
+  return intelligibility_enabled_ && intelligibility_enhancer_->active();
+}
+
+bool AudioProcessingImpl::rev_conversion_needed() const {
+  // Only supports conversions supported by AudioConverter.
+  return ((api_format_.reverse_input_stream() !=
+           api_format_.reverse_output_stream()) &&
+          ((api_format_.reverse_output_stream().num_channels() ==

[ekm, 2015/08/01 04:21:37]

Added these constraints because CommonFormats/AudioProcessingTest tests 2 input
0 output channels case. Maybe better to just declare this case invalid in
audio_processing.h as in ProcessStream case, and fix the test.

[aluebs-webrtc, 2015/08/03 16:30:50]

I think 0 channels just means it doesn't need any output, so you are doing the
right thing. Andrew?

[Andrew MacDonald, 2015/08/05 06:22:39]

If we want to be consistent, I think the test is invalid. If you don't want any
conversion, you should supply the same output as input config.

[ekm, 2015/08/07 06:05:59]

Done.

+            api_format_.reverse_input_stream().num_channels()) ||
+           (api_format_.reverse_output_stream().num_channels() == 1) ||
+           (api_format_.reverse_output_stream().num_channels() == 1)));
+}
+
 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_);
   }
@@ skipping 12 matching lines @@
 
 void AudioProcessingImpl::InitializeBeamformer() {
   if (beamformer_enabled_) {
     if (!beamformer_) {
       beamformer_.reset(new NonlinearBeamformer(array_geometry_));
     }
     beamformer_->Initialize(kChunkSizeMs, split_rate_);
   }
 }
 
+void AudioProcessingImpl::InitializeIntelligibility() {
+  if (intelligibility_enabled_) {
+    IntelligibilityEnhancer::Config config;
+    config.sample_rate_hz = split_rate_;
+    config.num_capture_channels = capture_audio_->num_channels();
+    config.num_render_channels = render_audio_->num_channels();
+    intelligibility_enhancer_.reset(new IntelligibilityEnhancer(config));
+  }
+}
+
 void AudioProcessingImpl::MaybeUpdateHistograms() {
   static const int kMinDiffDelayMs = 60;
 
   if (echo_cancellation()->is_enabled()) {
     // Activate delay_jumps_ counters if we know echo_cancellation is runnning.
     // If a stream has echo we know that the echo_cancellation is in process.
     if (stream_delay_jumps_ == -1 && echo_cancellation()->stream_has_echo()) {
       stream_delay_jumps_ = 0;
     }
     if (aec_system_delay_jumps_ == -1 &&
@@ skipping 78 matching lines @@
 
   return kNoError;
 }
 
 int AudioProcessingImpl::WriteInitMessage() {
   event_msg_->set_type(audioproc::Event::INIT);
   audioproc::Init* msg = event_msg_->mutable_init();
   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_num_reverse_channels(
+      api_format_.reverse_input_stream().num_channels());
+  msg->set_reverse_sample_rate(
+      api_format_.reverse_input_stream().sample_rate_hz());
   msg->set_output_sample_rate(api_format_.output_stream().sample_rate_hz());
+  // TODO(ekmeyerson): Add reverse output fields to event_msg_.
 
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc