Revert 217768 "Adding key press detection in the browser process."

> Adding key press detection in the browser process. > It works like this on the browser side: > A new object KeyPressMonitor is created on BrowserMainLoop and passed to AudioInputRendererHost to pass to AudioInputController. > AudioInputController::DoRecord calls KeyPressMonitor::AddKeyPressListener --> KeyPressMonitor listens to system key events through UserInputMonitor(only implemented on Linux) --> AudioInputController::OnKeyPressed is called and sets key_pressed_ --> When AudioInputController::OnData called, it writes key_pressed_ to shared memory along with the audio data buffer. > On the renderer side a new param "key_pressed" is added through the code path of passing the flag to the webrtc voice engine. > This CL includes all these changes except the implementation of UserInputMonitor for Windows and Mac. The impl of UserInputMonitor is mostly copied from remoting/host/local_input_monitor_linux.cc > > > BUG= > > Review URL: https://chromiumcodereview.appspot.com/21183002 TBR=jiayl@chromium.org Review URL: https://codereview.chromium.org/22871007 git-svn-id: svn://svn.chromium.org/chrome/trunk/src@217774 0039d316-1c4b-4281-b951-d872f2087c98
Denger-Network · Aug 15, 2013 · 1a8d01d · 1a8d01d
1 parent 8c2614c
commit 1a8d01d
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Showing 37 changed files with 159 additions and 931 deletions.
diff --git a/content/browser/browser_main_loop.cc b/content/browser/browser_main_loop.cc
@@ -52,7 +52,6 @@
 #include "crypto/nss_util.h"
 #include "media/audio/audio_manager.h"
 #include "media/base/media.h"
-#include "media/base/user_input_monitor.h"
 #include "media/midi/midi_manager.h"
 #include "net/base/network_change_notifier.h"
 #include "net/socket/client_socket_factory.h"
@@ -873,14 +872,6 @@ int BrowserMainLoop::BrowserThreadsStarted() {
         audio_manager_.get(), media_stream_manager_.get()));
   }
 
-  {
-    TRACE_EVENT0(
-        "startup",
-        "BrowserMainLoop::BrowserThreadsStarted::InitUserInputMonitor");
-    user_input_monitor_ = media::UserInputMonitor::Create(
-        io_thread_->message_loop_proxy(), main_thread_->message_loop_proxy());
-  }
-
   // Alert the clipboard class to which threads are allowed to access the
   // clipboard:
   std::vector<base::PlatformThreadId> allowed_clipboard_threads;

diff --git a/content/browser/browser_main_loop.h b/content/browser/browser_main_loop.h
@@ -26,7 +26,6 @@ class TraceMemoryController;
 namespace media {
 class AudioManager;
 class MIDIManager;
-class UserInputMonitor;
 }  // namespace media
 
 namespace net {
@@ -87,9 +86,6 @@ class CONTENT_EXPORT BrowserMainLoop {
   MediaStreamManager* media_stream_manager() const {
     return media_stream_manager_.get();
   }
-  media::UserInputMonitor* user_input_monitor() const {
-    return user_input_monitor_.get();
-  }
   media::MIDIManager* midi_manager() const { return midi_manager_.get(); }
   base::Thread* indexed_db_thread() const { return indexed_db_thread_.get(); }
 
@@ -126,8 +122,6 @@ class CONTENT_EXPORT BrowserMainLoop {
   scoped_ptr<base::PowerMonitor> power_monitor_;
   scoped_ptr<base::HighResolutionTimerManager> hi_res_timer_manager_;
   scoped_ptr<net::NetworkChangeNotifier> network_change_notifier_;
-  // user_input_monitor_ has to outlive audio_manager_, so declared first.
-  scoped_ptr<media::UserInputMonitor> user_input_monitor_;
   scoped_ptr<media::AudioManager> audio_manager_;
   scoped_ptr<media::MIDIManager> midi_manager_;
   scoped_ptr<AudioMirroringManager> audio_mirroring_manager_;

diff --git a/content/browser/renderer_host/media/audio_input_renderer_host.cc b/content/browser/renderer_host/media/audio_input_renderer_host.cc
@@ -51,12 +51,11 @@ AudioInputRendererHost::AudioEntry::~AudioEntry() {}
 AudioInputRendererHost::AudioInputRendererHost(
     media::AudioManager* audio_manager,
     MediaStreamManager* media_stream_manager,
-    AudioMirroringManager* audio_mirroring_manager,
-    media::UserInputMonitor* user_input_monitor)
+    AudioMirroringManager* audio_mirroring_manager)
     : audio_manager_(audio_manager),
       media_stream_manager_(media_stream_manager),
-      audio_mirroring_manager_(audio_mirroring_manager),
-      user_input_monitor_(user_input_monitor) {}
+      audio_mirroring_manager_(audio_mirroring_manager) {
+}
 
 AudioInputRendererHost::~AudioInputRendererHost() {
   DCHECK(audio_entries_.empty());
@@ -273,23 +272,20 @@ void AudioInputRendererHost::OnCreateStream(
     entry->controller = media::AudioInputController::CreateForStream(
         audio_manager_->GetMessageLoop(),
         this,
-        WebContentsAudioInputStream::Create(device_id,
-                                            audio_params,
-                                            audio_manager_->GetWorkerLoop(),
-                                            audio_mirroring_manager_),
-        entry->writer.get(),
-        user_input_monitor_);
+        WebContentsAudioInputStream::Create(
+            device_id, audio_params, audio_manager_->GetWorkerLoop(),
+            audio_mirroring_manager_),
+        entry->writer.get());
   } else {
     // TODO(henrika): replace CreateLowLatency() with Create() as soon
     // as satish has ensured that Speech Input also uses the default low-
     // latency path. See crbug.com/112472 for details.
-    entry->controller =
-        media::AudioInputController::CreateLowLatency(audio_manager_,
-                                                      this,
-                                                      audio_params,
-                                                      device_id,
-                                                      entry->writer.get(),
-                                                      user_input_monitor_);
+    entry->controller = media::AudioInputController::CreateLowLatency(
+        audio_manager_,
+        this,
+        audio_params,
+        device_id,
+        entry->writer.get());
   }
 
   if (!entry->controller.get()) {

diff --git a/content/browser/renderer_host/media/audio_input_renderer_host.h b/content/browser/renderer_host/media/audio_input_renderer_host.h
@@ -44,7 +44,6 @@
 namespace media {
 class AudioManager;
 class AudioParameters;
-class UserInputMonitor;
 }
 
 namespace content {
@@ -56,11 +55,10 @@ class CONTENT_EXPORT AudioInputRendererHost
       public media::AudioInputController::EventHandler {
  public:
   // Called from UI thread from the owner of this object.
-  // |user_input_monitor| is used for typing detection and can be NULL.
-  AudioInputRendererHost(media::AudioManager* audio_manager,
-                         MediaStreamManager* media_stream_manager,
-                         AudioMirroringManager* audio_mirroring_manager,
-                         media::UserInputMonitor* user_input_monitor);
+  AudioInputRendererHost(
+      media::AudioManager* audio_manager,
+      MediaStreamManager* media_stream_manager,
+      AudioMirroringManager* audio_mirroring_manager);
 
   // BrowserMessageFilter implementation.
   virtual void OnChannelClosing() OVERRIDE;
@@ -156,9 +154,6 @@ class CONTENT_EXPORT AudioInputRendererHost
   // A map of stream IDs to audio sources.
   AudioEntryMap audio_entries_;
 
-  // Raw pointer of the UserInputMonitor.
-  media::UserInputMonitor* user_input_monitor_;
-
   DISALLOW_COPY_AND_ASSIGN(AudioInputRendererHost);
 };
 

diff --git a/content/browser/renderer_host/media/audio_input_sync_writer.cc b/content/browser/renderer_host/media/audio_input_sync_writer.cc
@@ -29,17 +29,14 @@ void AudioInputSyncWriter::UpdateRecordedBytes(uint32 bytes) {
   socket_->Send(&bytes, sizeof(bytes));
 }
 
-uint32 AudioInputSyncWriter::Write(const void* data,
-                                   uint32 size,
-                                   double volume,
-                                   bool key_pressed) {
+uint32 AudioInputSyncWriter::Write(
+    const void* data, uint32 size, double volume) {
   uint8* ptr = static_cast<uint8*>(shared_memory_->memory());
   ptr += current_segment_id_ * shared_memory_segment_size_;
   media::AudioInputBuffer* buffer =
       reinterpret_cast<media::AudioInputBuffer*>(ptr);
   buffer->params.volume = volume;
   buffer->params.size = size;
-  buffer->params.key_pressed = key_pressed;
   memcpy(buffer->audio, data, size);
 
   if (++current_segment_id_ >= shared_memory_segment_count_)

diff --git a/content/browser/renderer_host/media/audio_input_sync_writer.h b/content/browser/renderer_host/media/audio_input_sync_writer.h
@@ -28,10 +28,7 @@ class AudioInputSyncWriter : public media::AudioInputController::SyncWriter {
 
   // media::AudioOutputController::SyncWriter implementation.
   virtual void UpdateRecordedBytes(uint32 bytes) OVERRIDE;
-  virtual uint32 Write(const void* data,
-                       uint32 size,
-                       double volume,
-                       bool key_pressed) OVERRIDE;
+  virtual uint32 Write(const void* data, uint32 size, double volume) OVERRIDE;
   virtual void Close() OVERRIDE;
 
   bool Init();

diff --git a/content/browser/renderer_host/render_process_host_impl.cc b/content/browser/renderer_host/render_process_host_impl.cc
@@ -617,14 +617,11 @@ void RenderProcessHostImpl::CreateMessageFilters() {
   channel_->AddFilter(new AudioInputRendererHost(
       audio_manager,
       media_stream_manager,
-      BrowserMainLoop::GetInstance()->audio_mirroring_manager(),
-      BrowserMainLoop::GetInstance()->user_input_monitor()));
+      BrowserMainLoop::GetInstance()->audio_mirroring_manager()));
   channel_->AddFilter(new AudioRendererHost(
-      GetID(),
-      audio_manager,
+      GetID(), audio_manager,
       BrowserMainLoop::GetInstance()->audio_mirroring_manager(),
-      media_internals,
-      media_stream_manager));
+      media_internals, media_stream_manager));
   channel_->AddFilter(
       new MIDIHost(BrowserMainLoop::GetInstance()->midi_manager()));
   channel_->AddFilter(new MIDIDispatcherHost(GetID(), browser_context));

diff --git a/content/browser/speech/speech_recognizer_impl.cc b/content/browser/speech/speech_recognizer_impl.cc
@@ -564,7 +564,7 @@ SpeechRecognizerImpl::StartRecording(const FSMEventArgs&) {
       new OnDataConverter(input_parameters, output_parameters));
 
   audio_controller_ = AudioInputController::Create(
-      audio_manager, this, input_parameters, device_id_, NULL);
+      audio_manager, this, input_parameters, device_id_);
 
   if (!audio_controller_.get()) {
     return Abort(SpeechRecognitionError(SPEECH_RECOGNITION_ERROR_AUDIO));

diff --git a/content/renderer/media/webaudio_capturer_source.cc b/content/renderer/media/webaudio_capturer_source.cc
@@ -96,7 +96,7 @@ void WebAudioCapturerSource::consumeAudio(
   int capture_frames = params_.frames_per_buffer();
   while (fifo_->frames() >= capture_frames) {
     fifo_->Consume(capture_bus_.get(), 0, capture_frames);
-    callback_->Capture(capture_bus_.get(), 0, 1.0, false);
+    callback_->Capture(capture_bus_.get(), 0, 1.0);
   }
 }
 

diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
@@ -102,16 +102,14 @@ class WebRtcAudioCapturer::TrackOwner
                    int number_of_channels,
                    int number_of_frames,
                    int audio_delay_milliseconds,
-                   int volume,
-                   bool key_pressed) {
+                   int volume) {
     base::AutoLock lock(lock_);
     if (delegate_) {
       delegate_->CaptureData(audio_data,
                              number_of_channels,
                              number_of_frames,
                              audio_delay_milliseconds,
-                             volume,
-                             key_pressed);
+                             volume);
     }
   }
 
@@ -426,11 +424,10 @@ void WebRtcAudioCapturer::SetAutomaticGainControl(bool enable) {
 
 void WebRtcAudioCapturer::Capture(media::AudioBus* audio_source,
                                   int audio_delay_milliseconds,
-                                  double volume,
-                                  bool key_pressed) {
-// This callback is driven by AudioInputDevice::AudioThreadCallback if
-// |source_| is AudioInputDevice, otherwise it is driven by client's
-// CaptureCallback.
+                                  double volume) {
+  // This callback is driven by AudioInputDevice::AudioThreadCallback if
+  // |source_| is AudioInputDevice, otherwise it is driven by client's
+  // CaptureCallback.
 #if defined(OS_WIN) || defined(OS_MACOSX)
   DCHECK_LE(volume, 1.0);
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
@@ -474,11 +471,8 @@ void WebRtcAudioCapturer::Capture(media::AudioBus* audio_source,
        it != tracks.end();
        ++it) {
     (*it)->CaptureData(buffer_ref_while_calling->buffer(),
-                       audio_source->channels(),
-                       audio_source->frames(),
-                       audio_delay_milliseconds,
-                       volume,
-                       key_pressed);
+                       audio_source->channels(), audio_source->frames(),
+                       audio_delay_milliseconds, volume_);
   }
 }
 

diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h
@@ -108,8 +108,7 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // Called on the AudioInputDevice audio thread.
   virtual void Capture(media::AudioBus* audio_source,
                        int audio_delay_milliseconds,
-                       double volume,
-                       bool key_pressed) OVERRIDE;
+                       double volume) OVERRIDE;
   virtual void OnCaptureError() OVERRIDE;
 
   // Reconfigures the capturer with a new buffer size and capture parameters.

diff --git a/content/renderer/media/webrtc_audio_capturer_sink_owner.cc b/content/renderer/media/webrtc_audio_capturer_sink_owner.cc
@@ -18,19 +18,13 @@ int WebRtcAudioCapturerSinkOwner::CaptureData(const std::vector<int>& channels,
                                               int number_of_frames,
                                               int audio_delay_milliseconds,
                                               int current_volume,
-                                              bool need_audio_processing,
-                                              bool key_pressed) {
+                                              bool need_audio_processing) {
   base::AutoLock lock(lock_);
   if (delegate_) {
-    delegate_->CaptureData(channels,
-                           audio_data,
-                           sample_rate,
-                           number_of_channels,
-                           number_of_frames,
-                           audio_delay_milliseconds,
-                           current_volume,
-                           need_audio_processing,
-                           key_pressed);
+    return delegate_->CaptureData(channels, audio_data, sample_rate,
+                                  number_of_channels, number_of_frames,
+                                  audio_delay_milliseconds, current_volume,
+                                  need_audio_processing);
   }
 
   return 0;

diff --git a/content/renderer/media/webrtc_audio_capturer_sink_owner.h b/content/renderer/media/webrtc_audio_capturer_sink_owner.h
@@ -31,8 +31,7 @@ class WebRtcAudioCapturerSinkOwner
                           int number_of_frames,
                           int audio_delay_milliseconds,
                           int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) OVERRIDE;
+                          bool need_audio_processing) OVERRIDE;
 
   virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;
 

diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -59,8 +59,7 @@ int WebRtcAudioDeviceImpl::CaptureData(const std::vector<int>& channels,
                                        int number_of_frames,
                                        int audio_delay_milliseconds,
                                        int current_volume,
-                                       bool need_audio_processing,
-                                       bool key_pressed) {
+                                       bool need_audio_processing) {
   int total_delay_ms = 0;
   {
     base::AutoLock auto_lock(lock_);
@@ -76,9 +75,11 @@ int WebRtcAudioDeviceImpl::CaptureData(const std::vector<int>& channels,
   // Write audio samples in blocks of 10 milliseconds to the registered
   // webrtc::AudioTransport sink. Keep writing until our internal byte
   // buffer is empty.
+  // TODO(niklase): Wire up the key press detection.
   const int16* audio_buffer = audio_data;
   const int samples_per_10_msec = (sample_rate / 100);
   int accumulated_audio_samples = 0;
+  bool key_pressed = false;
   uint32_t new_volume = 0;
   while (accumulated_audio_samples < number_of_frames) {
     // Deliver 10ms of recorded 16-bit linear PCM audio.

diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
@@ -223,8 +223,7 @@ class WebRtcAudioCapturerSink {
                           int number_of_frames,
                           int audio_delay_milliseconds,
                           int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) = 0;
+                          bool need_audio_processing) = 0;
 
   // Set the format for the capture audio parameters.
   virtual void SetCaptureFormat(const media::AudioParameters& params) = 0;
@@ -340,8 +339,7 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
                           int number_of_frames,
                           int audio_delay_milliseconds,
                           int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) OVERRIDE;
+                          bool need_audio_processing) OVERRIDE;
 
   // Called on the main render thread.
   virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;

diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc
@@ -221,8 +221,7 @@ class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
                           int number_of_frames,
                           int audio_delay_milliseconds,
                           int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) OVERRIDE {
+                          bool need_audio_processing) OVERRIDE {
     // Signal that a callback has been received.
     event_->Signal();
     return 0;
@@ -382,13 +381,8 @@ int RunWebRtcLoopbackTimeTest(media::AudioManager* manager,
     capturer_sink->CaptureData(
         voe_channels,
         reinterpret_cast<int16*>(capture_data.get() + input_packet_size * j),
-        params.sample_rate(),
-        params.channels(),
-        params.frames_per_buffer(),
-        kHardwareLatencyInMs,
-        1.0,
-        enable_apm,
-        false);
+        params.sample_rate(), params.channels(), params.frames_per_buffer(),
+        kHardwareLatencyInMs, 1.0, enable_apm);
 
     // Receiving data from WebRtc.
     renderer_source->RenderData(

diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
@@ -52,14 +52,13 @@ void WebRtcLocalAudioRenderer::OnRenderError() {
 
 // content::WebRtcAudioCapturerSink implementation
 int WebRtcLocalAudioRenderer::CaptureData(const std::vector<int>& channels,
-                                          const int16* audio_data,
-                                          int sample_rate,
-                                          int number_of_channels,
-                                          int number_of_frames,
-                                          int audio_delay_milliseconds,
-                                          int current_volume,
-                                          bool need_audio_processing,
-                                          bool key_pressed) {
+                                           const int16* audio_data,
+                                           int sample_rate,
+                                           int number_of_channels,
+                                           int number_of_frames,
+                                           int audio_delay_milliseconds,
+                                           int current_volume,
+                                           bool need_audio_processing) {
   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData");
   base::AutoLock auto_lock(thread_lock_);
 

diff --git a/content/renderer/media/webrtc_local_audio_renderer.h b/content/renderer/media/webrtc_local_audio_renderer.h
@@ -78,8 +78,7 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer
                           int number_of_frames,
                           int audio_delay_milliseconds,
                           int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) OVERRIDE;
+                          bool need_audio_processing) OVERRIDE;
 
   // Can be called on different user thread.
   virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;