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 0c2fd4573c58f71897c8fb1321457157e547b463..4c17c34a49ae6afabac3210ca4986f7a46632b30 100644
--- a/content/renderer/media/audio_input_device.cc
+++ b/content/renderer/media/audio_input_device.cc
@@ -99,15 +99,27 @@ 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.
@@ -121,7 +133,8 @@ 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;
@@ -150,6 +163,7 @@ 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
@@ -170,6 +184,21 @@ void AudioInputDevice::SetVolumeOnIOThread(double volume) {
     Send(new AudioInputHostMsg_SetVolume(stream_id_, volume));
 }
 
+void AudioInputDevice::SetAutomaticGainControlOnIOThread(bool enabled) {
+  DCHECK(message_loop()->BelongsToCurrentThread());
+  if (stream_id_) {
+    // We enter this scope if InitializeOnIOThread() has already been called.
+    // It means that an audio stream already exists but AGC is disabled.
+    // We must therefore modify the current AGC setting for the existing
+    // stream using IPC.
+    Send(new AudioInputHostMsg_SetAutomaticGainControl(stream_id_, enabled));
+  }
+
+  // Always store new AGC setting. 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,
@@ -261,7 +290,7 @@ 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;
@@ -298,7 +327,14 @@ void AudioInputDevice::AudioThreadCallback::MapSharedMemory() {
 
 void AudioInputDevice::AudioThreadCallback::Process(int pending_data) {
   int audio_delay_milliseconds = pending_data / bytes_per_ms_;
-  int16* memory = reinterpret_cast<int16*>(shared_memory_.memory());
+
+  AudioInputBuffer* buffer =
+    reinterpret_cast<AudioInputBuffer*>(shared_memory_.memory());

[tommi (sloooow) - chröme, 2012/03/16 13:31:05]

indent

[henrika (OOO until Aug 14), 2012/03/21 10:16:04]

Done.

+
+  // TODO(henrika): uint32 size = buffer->size;

[tommi (sloooow) - chröme, 2012/03/16 13:31:05]

ping

[henrika (OOO until Aug 14), 2012/03/21 10:16:04]

Added DCHECK which uses the size variable.

+
+  double volume = buffer->volume;
+  int16* memory = reinterpret_cast<int16*>(buffer->audio);

[tommi (sloooow) - chröme, 2012/03/16 13:31:05]

nit: &buffer->audio[0]

[henrika (OOO until Aug 14), 2012/03/21 10:16:04]

Done.

   const size_t number_of_frames = audio_parameters_.samples_per_packet;
   const int bytes_per_sample = sizeof(memory[0]);
 
@@ -317,5 +353,5 @@ 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);
 }