Refactor AudioParameters member setting.

content/browser/speech/speech_recognizer_impl.cc

 // Copyright (c) 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/browser/speech/speech_recognizer_impl.h"
 
 #include "base/basictypes.h"
 #include "base/bind.h"
 #include "base/time/time.h"
 #include "content/browser/browser_main_loop.h"
@@ skipping 221 matching lines @@
 }
 
 bool SpeechRecognizerImpl::IsActive() const {
   // Checking the FSM state from another thread (thus, while the FSM is
   // potentially concurrently evolving) is meaningless.
   DCHECK_CURRENTLY_ON(BrowserThread::IO);
   return state_ != STATE_IDLE && state_ != STATE_ENDED;
 }
 
 bool SpeechRecognizerImpl::IsCapturingAudio() const {
-  DCHECK_CURRENTLY_ON(BrowserThread::IO); // See IsActive().
+  DCHECK_CURRENTLY_ON(BrowserThread::IO);  // See IsActive().
   const bool is_capturing_audio = state_ >= STATE_STARTING &&
                                   state_ <= STATE_RECOGNIZING;
   DCHECK((is_capturing_audio && (audio_controller_.get() != NULL)) ||
          (!is_capturing_audio && audio_controller_.get() == NULL));
   return is_capturing_audio;
 }
 
 const SpeechRecognitionEngine&
 SpeechRecognizerImpl::recognition_engine() const {
   return *(recognition_engine_.get());
@@ skipping 229 matching lines @@
                                 state_ <= STATE_RECOGNIZING;
   const bool clip_detected = DetectClipping(raw_audio);
   float rms = 0.0f;
 
   num_samples_recorded_ += raw_audio.NumSamples();
 
   if (route_to_endpointer)
     endpointer_.ProcessAudio(raw_audio, &rms);
 
   if (route_to_vumeter) {
-    DCHECK(route_to_endpointer); // Depends on endpointer due to |rms|.
+    DCHECK(route_to_endpointer);  // Depends on endpointer due to |rms|.
     UpdateSignalAndNoiseLevels(rms, clip_detected);
   }
   if (route_to_sr_engine) {
     DCHECK(recognition_engine_.get() != NULL);
     recognition_engine_->TakeAudioChunk(raw_audio);
   }
 }
 
 SpeechRecognizerImpl::FSMState
 SpeechRecognizerImpl::StartRecording(const FSMEventArgs&) {
@@ skipping 53 matching lines @@
     // Use native audio parameters but avoid opening up at the native buffer
     // size. Instead use same frame size (in milliseconds) as WebSpeech uses.
     // We rely on internal buffers in the audio back-end to fulfill this request
     // and the idea is to simplify the audio conversion since each Convert()
     // call will then render exactly one ProvideInput() call.
     // in_params.sample_rate()
     frames_per_buffer =
         ((in_params.sample_rate() * chunk_duration_ms) / 1000.0) + 0.5;
     input_parameters.Reset(in_params.format(),
                            in_params.channel_layout(),
-                           in_params.channels(),
                            in_params.sample_rate(),
                            in_params.bits_per_sample(),
                            frames_per_buffer);
+    input_parameters.set_channels_for_discrete(in_params.channels());
     DVLOG(1) << "SRI::input_parameters: "
              << input_parameters.AsHumanReadableString();
   }
 
   // Create an audio converter which converts data between native input format
   // and WebSpeech specific output format.
   audio_converter_.reset(
       new OnDataConverter(input_parameters, output_parameters));
 
   audio_controller_ = AudioInputController::Create(
@@ skipping 247 matching lines @@
 SpeechRecognizerImpl::FSMEventArgs::FSMEventArgs(FSMEvent event_value)
     : event(event_value),
       audio_data(NULL),
       engine_error(SPEECH_RECOGNITION_ERROR_NONE) {
 }
 
 SpeechRecognizerImpl::FSMEventArgs::~FSMEventArgs() {
 }
 
 }  // namespace content