Lock scheme #6: Preparational work before introducing the locks in order to harmonize the code

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 177 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
-      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),
@@ skipping 91 matching lines @@
         LayoutHasKeyboard(reverse_layout)}}};
 
   return Initialize(processing_config);
 }
 
 int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
   CriticalSectionScoped crit_scoped(crit_);
   return InitializeLocked(processing_config);
 }
 
+// Calls InitializeLocked() if any of the audio parameters have changed from
+// their current values.
+int AudioProcessingImpl::MaybeInitializeLocked(
+    const ProcessingConfig& processing_config) {
+  if (processing_config == shared_state_.api_format_) {
+    return kNoError;
+  }
+  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();
+      beamformer_enabled_
+          ? shared_state_.api_format_.input_stream().num_channels()
+          : shared_state_.api_format_.output_stream().num_channels();
   const int rev_audio_buffer_out_num_frames =
-      api_format_.reverse_output_stream().num_frames() == 0
+      shared_state_.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) {
+          : shared_state_.api_format_.reverse_output_stream().num_frames();
+  if (shared_state_.api_format_.reverse_input_stream().num_channels() > 0) {
     render_audio_.reset(new AudioBuffer(
-        api_format_.reverse_input_stream().num_frames(),
-        api_format_.reverse_input_stream().num_channels(),
+        shared_state_.api_format_.reverse_input_stream().num_frames(),
+        shared_state_.api_format_.reverse_input_stream().num_channels(),
         rev_proc_format_.num_frames(), rev_proc_format_.num_channels(),
         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());
+          shared_state_.api_format_.reverse_input_stream().num_channels(),
+          shared_state_.api_format_.reverse_input_stream().num_frames(),
+          shared_state_.api_format_.reverse_output_stream().num_channels(),
+          shared_state_.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()));
+  capture_audio_.reset(
+      new AudioBuffer(shared_state_.api_format_.input_stream().num_frames(),
+                      shared_state_.api_format_.input_stream().num_channels(),
+                      fwd_proc_format_.num_frames(), fwd_audio_buffer_channels,
+                      shared_state_.api_format_.output_stream().num_frames()));
 
   // Initialize all components.
   for (auto item : component_list_) {
     int err = item->Initialize();
     if (err != kNoError) {
       return err;
     }
   }
 
   InitializeExperimentalAgc();
@@ skipping 35 matching lines @@
       !(num_out_channels == 1 || num_out_channels == num_in_channels)) {
     return kBadNumberChannelsError;
   }
 
   if (beamformer_enabled_ &&
       (static_cast<size_t>(num_in_channels) != array_geometry_.size() ||
        num_out_channels > 1)) {
     return kBadNumberChannelsError;
   }
 
-  api_format_ = config;
+  shared_state_.api_format_ = config;
 
   // We process at the closest native rate >= min(input rate, output rate)...
   const int min_proc_rate =
-      std::min(api_format_.input_stream().sample_rate_hz(),
-               api_format_.output_stream().sample_rate_hz());
+      std::min(shared_state_.api_format_.input_stream().sample_rate_hz(),
+               shared_state_.api_format_.output_stream().sample_rate_hz());
   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;
     }
   }
   // ...with one exception.
   if (echo_control_mobile_->is_enabled() &&
       min_proc_rate > kMaxAECMSampleRateHz) {
     fwd_proc_rate = kMaxAECMSampleRateHz;
   }
 
   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_input_stream().sample_rate_hz() ==
+    if (shared_state_.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);
 
   if (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
       fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz) {
     split_rate_ = kSampleRate16kHz;
   } else {
     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(
-    const ProcessingConfig& processing_config) {
-  if (processing_config == api_format_) {
-    return kNoError;
-  }
-  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::proc_sample_rate_hz() const {
   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 api_format_.input_stream().num_channels();
+  return shared_state_.api_format_.input_stream().num_channels();
 }
 
 int AudioProcessingImpl::num_output_channels() const {
-  return api_format_.output_stream().num_channels();
+  return shared_state_.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_);
   }
 }
 
 
 int AudioProcessingImpl::ProcessStream(const float* const* src,
                                        size_t samples_per_channel,
                                        int input_sample_rate_hz,
                                        ChannelLayout input_layout,
                                        int output_sample_rate_hz,
                                        ChannelLayout output_layout,
                                        float* const* dest) {
   CriticalSectionScoped crit_scoped(crit_);
-  StreamConfig input_stream = api_format_.input_stream();
+  StreamConfig input_stream = shared_state_.api_format_.input_stream();
   input_stream.set_sample_rate_hz(input_sample_rate_hz);
   input_stream.set_num_channels(ChannelsFromLayout(input_layout));
   input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout));
 
-  StreamConfig output_stream = api_format_.output_stream();
+  StreamConfig output_stream = shared_state_.api_format_.output_stream();
   output_stream.set_sample_rate_hz(output_sample_rate_hz);
   output_stream.set_num_channels(ChannelsFromLayout(output_layout));
   output_stream.set_has_keyboard(LayoutHasKeyboard(output_layout));
 
   if (samples_per_channel != input_stream.num_frames()) {
     return kBadDataLengthError;
   }
   return ProcessStream(src, input_stream, output_stream, dest);
 }
 
 int AudioProcessingImpl::ProcessStream(const float* const* src,
                                        const StreamConfig& input_config,
                                        const StreamConfig& output_config,
                                        float* const* dest) {
   CriticalSectionScoped crit_scoped(crit_);
   if (!src || !dest) {
     return kNullPointerError;
   }
 
   echo_cancellation_->ReadQueuedRenderData();
   echo_control_mobile_->ReadQueuedRenderData();
   gain_control_->ReadQueuedRenderData();
 
-  ProcessingConfig processing_config = api_format_;
+  ProcessingConfig processing_config = shared_state_.api_format_;
   processing_config.input_stream() = input_config;
   processing_config.output_stream() = output_config;
 
   RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
   assert(processing_config.input_stream().num_frames() ==
-         api_format_.input_stream().num_frames());
+         shared_state_.api_format_.input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     RETURN_ON_ERR(WriteConfigMessage(false));
 
     event_msg_->set_type(audioproc::Event::STREAM);
     audioproc::Stream* msg = event_msg_->mutable_stream();
     const size_t channel_size =
-        sizeof(float) * api_format_.input_stream().num_frames();
-    for (int i = 0; i < api_format_.input_stream().num_channels(); ++i)
+        sizeof(float) * shared_state_.api_format_.input_stream().num_frames();
+    for (int i = 0; i < shared_state_.api_format_.input_stream().num_channels();
+         ++i)
       msg->add_input_channel(src[i], channel_size);
   }
 #endif
 
-  capture_audio_->CopyFrom(src, api_format_.input_stream());
+  capture_audio_->CopyFrom(src, shared_state_.api_format_.input_stream());
   RETURN_ON_ERR(ProcessStreamLocked());
-  capture_audio_->CopyTo(api_format_.output_stream(), dest);
+  capture_audio_->CopyTo(shared_state_.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) * api_format_.output_stream().num_frames();
-    for (int i = 0; i < api_format_.output_stream().num_channels(); ++i)
+        sizeof(float) * shared_state_.api_format_.output_stream().num_frames();
+    for (int i = 0;
+         i < shared_state_.api_format_.output_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_);
   echo_cancellation_->ReadQueuedRenderData();
   echo_control_mobile_->ReadQueuedRenderData();
   gain_control_->ReadQueuedRenderData();
 
   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_ > kMaxAECMSampleRateHz) {
     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.
-  ProcessingConfig processing_config = api_format_;
+  ProcessingConfig processing_config = shared_state_.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().num_frames()) {
+  if (frame->samples_per_channel_ !=
+      shared_state_.api_format_.input_stream().num_frames()) {
     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_;
     msg->set_input_data(frame->data_, data_size);
@@ skipping 109 matching lines @@
 }
 
 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));
   if (is_rev_processed()) {
-    render_audio_->CopyTo(api_format_.reverse_output_stream(), dest);
+    render_audio_->CopyTo(shared_state_.api_format_.reverse_output_stream(),
+                          dest);
   } else if (rev_conversion_needed()) {
     render_converter_->Convert(src, reverse_input_config.num_samples(), dest,
                                reverse_output_config.num_samples());
   } else {
     CopyAudioIfNeeded(src, reverse_input_config.num_frames(),
                       reverse_input_config.num_channels(), dest);
   }
 
   return kNoError;
 }
 
 int AudioProcessingImpl::AnalyzeReverseStream(
     const float* const* src,
     const StreamConfig& reverse_input_config,
     const StreamConfig& reverse_output_config) {
   CriticalSectionScoped crit_scoped(crit_);
   if (src == NULL) {
     return kNullPointerError;
   }
 
   if (reverse_input_config.num_channels() <= 0) {
     return kBadNumberChannelsError;
   }
 
-  ProcessingConfig processing_config = api_format_;
+  ProcessingConfig processing_config = shared_state_.api_format_;
   processing_config.reverse_input_stream() = reverse_input_config;
   processing_config.reverse_output_stream() = reverse_output_config;
 
   RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
   assert(reverse_input_config.num_frames() ==
-         api_format_.reverse_input_stream().num_frames());
+         shared_state_.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_input_stream().num_frames();
-    for (int i = 0; i < api_format_.reverse_input_stream().num_channels(); ++i)
+        sizeof(float) *
+        shared_state_.api_format_.reverse_input_stream().num_frames();
+    for (int i = 0;
+         i < shared_state_.api_format_.reverse_input_stream().num_channels();
+         ++i)
       msg->add_channel(src[i], channel_size);
     RETURN_ON_ERR(WriteMessageToDebugFile());
   }
 #endif
 
-  render_audio_->CopyFrom(src, api_format_.reverse_input_stream());
+  render_audio_->CopyFrom(src,
+                          shared_state_.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()) {
+  if (frame->sample_rate_hz_ !=
+      shared_state_.api_format_.input_stream().sample_rate_hz()) {
     return kBadSampleRateError;
   }
 
   if (frame->num_channels_ <= 0) {
     return kBadNumberChannelsError;
   }
 
-  ProcessingConfig processing_config = api_format_;
+  ProcessingConfig processing_config = shared_state_.api_format_;
   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_input_stream().num_frames()) {
+      shared_state_.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);
@@ skipping 203 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 ((api_format_.output_stream().num_channels() !=
-           api_format_.input_stream().num_channels()) ||
+  return ((shared_state_.api_format_.output_stream().num_channels() !=
+           shared_state_.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_.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_.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 {
-  return (api_format_.reverse_input_stream() !=
-          api_format_.reverse_output_stream());
+  return (shared_state_.api_format_.reverse_input_stream() !=
+          shared_state_.api_format_.reverse_output_stream());
 }
 
 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_.sample_rate_hz(), split_rate_,
-        api_format_.output_stream().num_channels());
+        shared_state_.api_format_.output_stream().num_channels());
   }
 }
 
 void AudioProcessingImpl::InitializeBeamformer() {
   if (beamformer_enabled_) {
     if (!beamformer_) {
       beamformer_.reset(
           new NonlinearBeamformer(array_geometry_, target_direction_));
     }
     beamformer_->Initialize(kChunkSizeMs, split_rate_);
@@ skipping 98 matching lines @@
   }
 
   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(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_sample_rate(
+      shared_state_.api_format_.input_stream().sample_rate_hz());
+  msg->set_num_input_channels(
+      shared_state_.api_format_.input_stream().num_channels());
+  msg->set_num_output_channels(
+      shared_state_.api_format_.output_stream().num_channels());
   msg->set_num_reverse_channels(
-      api_format_.reverse_input_stream().num_channels());
+      shared_state_.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());
+      shared_state_.api_format_.reverse_input_stream().sample_rate_hz());
+  msg->set_output_sample_rate(
+      shared_state_.api_format_.output_stream().sample_rate_hz());
   // TODO(ekmeyerson): Add reverse output fields to event_msg_.
 
   RETURN_ON_ERR(WriteMessageToDebugFile());
   return kNoError;
 }
 
 int AudioProcessingImpl::WriteConfigMessage(bool forced) {
   audioproc::Config config;
 
   config.set_aec_enabled(echo_cancellation_->is_enabled());
@@ skipping 33 matching lines @@
 
   event_msg_->set_type(audioproc::Event::CONFIG);
   event_msg_->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile());
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc