Adds Analog Gain Control (AGC) to the WebRTC client.

content/renderer/media/audio_input_device.cc

Index: content/renderer/media/audio_input_device.cc
diff --git a/content/renderer/media/audio_input_device.cc b/content/renderer/media/audio_input_device.cc
index 24e9da923a14f3f8ad72d060fe3381fcb28efb51..a7304f5de7dfb7a2ce07461eda73c0b5e89b51bc 100644
--- a/content/renderer/media/audio_input_device.cc
+++ b/content/renderer/media/audio_input_device.cc
@@ -47,7 +47,8 @@ AudioInputDevice::AudioInputDevice(const AudioParameters& params,
       volume_(1.0),
       stream_id_(0),
       session_id_(0),
-      pending_device_ready_(false) {
+      pending_device_ready_(false),
+     agc_is_enabled_(false) {

[tommi (sloooow) - chröme, 2012/03/26 15:26:40]

indent is off.

[henrika (OOO until Aug 14), 2012/03/27 09:20:38]

Done.

   filter_ = RenderThreadImpl::current()->audio_input_message_filter();
 }
 
@@ -82,15 +83,27 @@ void AudioInputDevice::Stop() {
 }
 
 bool AudioInputDevice::SetVolume(double volume) {
-  NOTIMPLEMENTED();
-  return false;
+  if (volume < 0 || volume > 1.0)
+    return false;
+
+  message_loop()->PostTask(FROM_HERE,
+      base::Bind(&AudioInputDevice::SetVolumeOnIOThread, this, volume));
+
+  return true;
 }
 
 bool AudioInputDevice::GetVolume(double* volume) {
-  NOTIMPLEMENTED();
+  NOTREACHED();
   return false;
 }
 
+void AudioInputDevice::SetAutomaticGainControl(bool enabled) {
+  DVLOG(1) << "SetAutomaticGainControl(enabled=" << enabled << ")";
+  message_loop()->PostTask(FROM_HERE,
+      base::Bind(&AudioInputDevice::SetAutomaticGainControlOnIOThread,
+          this, enabled));
+}
+
 void AudioInputDevice::InitializeOnIOThread() {
   DCHECK(message_loop()->BelongsToCurrentThread());
   // Make sure we don't call Start() more than once.
@@ -104,7 +117,8 @@ void AudioInputDevice::InitializeOnIOThread() {
   // and create the stream when getting a OnDeviceReady() callback.
   if (!session_id_) {
     Send(new AudioInputHostMsg_CreateStream(
-        stream_id_, audio_parameters_, AudioManagerBase::kDefaultDeviceId));
+        stream_id_, audio_parameters_, AudioManagerBase::kDefaultDeviceId,
+        agc_is_enabled_));
   } else {
     Send(new AudioInputHostMsg_StartDevice(stream_id_, session_id_));
     pending_device_ready_ = true;
@@ -133,6 +147,7 @@ void AudioInputDevice::ShutDownOnIOThread() {
     stream_id_ = 0;
     session_id_ = 0;
     pending_device_ready_ = false;
+    agc_is_enabled_ = false;
   }
 
   // We can run into an issue where ShutDownOnIOThread is called right after
@@ -153,6 +168,18 @@ void AudioInputDevice::SetVolumeOnIOThread(double volume) {
     Send(new AudioInputHostMsg_SetVolume(stream_id_, volume));
 }
 
+void AudioInputDevice::SetAutomaticGainControlOnIOThread(bool enabled) {
+  DCHECK(message_loop()->BelongsToCurrentThread());
+  // The state of the AGC can not be modified while capturing is active.

[tommi (sloooow) - chröme, 2012/03/26 15:26:40]

Are the reasons for this still valid?

[henrika (OOO until Aug 14), 2012/03/27 09:20:38]

Yes, I have removed locks in on the media side under the assumption that this
condition holds.

+  DCHECK_EQ(0, stream_id_);

[scherkus (not reviewing), 2012/03/26 22:41:04]

want to move that comment into a DCHECK() message?

[henrika (OOO until Aug 14), 2012/03/27 09:20:38]

Done.

+  if (stream_id_)
+    return;
+
+  // We simply store the new AGC setting here. This value will be used when
+  // a new stream is initialized and by GetAutomaticGainControl().
+  agc_is_enabled_ = enabled;
+}
+
 void AudioInputDevice::OnStreamCreated(
     base::SharedMemoryHandle handle,
     base::SyncSocket::Handle socket_handle,
@@ -250,7 +277,7 @@ void AudioInputDevice::OnDeviceReady(const std::string& device_id) {
     stream_id_ = 0;
   } else {
     Send(new AudioInputHostMsg_CreateStream(stream_id_, audio_parameters_,
-                                            device_id));
+                                            device_id, agc_is_enabled_));
   }
 
   pending_device_ready_ = false;
@@ -286,8 +313,18 @@ void AudioInputDevice::AudioThreadCallback::MapSharedMemory() {
 }
 
 void AudioInputDevice::AudioThreadCallback::Process(int pending_data) {
+

[tommi (sloooow) - chröme, 2012/03/26 15:26:40]

remove this empty line

[henrika (OOO until Aug 14), 2012/03/27 09:20:38]

Done.

+  // The shared memory represents parameters, size of the data buffer and the
+  // actual data buffer containing audio data. Map the memory into this
+  // structure and parse out parameters and data area.
+  AudioInputBuffer* buffer =
+      reinterpret_cast<AudioInputBuffer*>(shared_memory_.memory());
+  uint32 size = buffer->params.size;
+  DCHECK_EQ(size, memory_length_ - sizeof(AudioInputBufferParameters));
+  double volume = buffer->params.volume;
+
   int audio_delay_milliseconds = pending_data / bytes_per_ms_;
-  int16* memory = reinterpret_cast<int16*>(shared_memory_.memory());
+  int16* memory = reinterpret_cast<int16*>(buffer->audio);

[tommi (sloooow) - chröme, 2012/03/26 15:26:40]

nit: &buffer->audio[0]

[henrika (OOO until Aug 14), 2012/03/27 09:20:38]

Done.

   const size_t number_of_frames = audio_parameters_.frames_per_buffer();
   const int bytes_per_sample = sizeof(memory[0]);
 
@@ -306,5 +343,5 @@ void AudioInputDevice::AudioThreadCallback::Process(int pending_data) {
   // Deliver captured data to the client in floating point format
   // and update the audio-delay measurement.
   capture_callback_->Capture(audio_data_, number_of_frames,
-                             audio_delay_milliseconds);
+                             audio_delay_milliseconds, volume);
 }