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

media/audio/win/audio_low_latency_input_win.cc

Index: media/audio/win/audio_low_latency_input_win.cc
diff --git a/media/audio/win/audio_low_latency_input_win.cc b/media/audio/win/audio_low_latency_input_win.cc
index 839ace84c46fa4825166fa0970921285548ba813..18cf6605efa34277d0302ff3e6e64be1846ad1d0 100644
--- a/media/audio/win/audio_low_latency_input_win.cc
+++ b/media/audio/win/audio_low_latency_input_win.cc
@@ -11,6 +11,8 @@
 #include "media/audio/win/audio_manager_win.h"
 #include "media/audio/win/avrt_wrapper_win.h"
 
+static const int kMinIntervalBetweenVolumeUpdatesMs = 1000;
+
 using base::win::ScopedComPtr;
 using base::win::ScopedCOMInitializer;
 
@@ -24,7 +26,9 @@ WASAPIAudioInputStream::WASAPIAudioInputStream(
       started_(false),
       endpoint_buffer_size_frames_(0),
       device_id_(device_id),
-      sink_(NULL) {
+      sink_(NULL),
+      volume_(0.0),
+      agc_is_enabled_(0) {
   DCHECK(manager_);
 
   // Load the Avrt DLL if not already loaded. Required to support MMCSS.
@@ -192,6 +196,7 @@ double WASAPIAudioInputStream::GetMaxVolume() {
 }
 
 void WASAPIAudioInputStream::SetVolume(double volume) {
+  DVLOG(1) << "SetVolume(volume=" << volume << ")";
   DCHECK(CalledOnValidThread());
   DCHECK(volume <= 1.0 && volume >= 0.0);
 
@@ -202,8 +207,18 @@ void WASAPIAudioInputStream::SetVolume(double volume) {
   // Set a new master volume level. Valid volume levels are in the range
   // 0.0 to 1.0. Ignore volume-change events.
   HRESULT hr = simple_audio_volume_->SetMasterVolume(static_cast<float>(volume),
-                                                     NULL);
+    NULL);

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

indent

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

Done.

   DLOG_IF(WARNING, FAILED(hr)) << "Failed to set new input master volume.";
+
+  // We take new volume samples once every second when the AGC is enabled.
+  // To ensure that a new setting has an immediate effect, the new volume
+  // setting is cached here. It will ensure that the next OnData() callback
+  // will contain a new valid volume level. If this approach was not taken,
+  // we could report invalid volume levels to the client for a time period
+  // of up to one second.
+  if (GetAutomaticGainControl()) {
+    volume_ = GetVolume();
+  }
 }
 
 double WASAPIAudioInputStream::GetVolume() {
@@ -220,6 +235,14 @@ double WASAPIAudioInputStream::GetVolume() {
   return static_cast<double>(level);
 }
 
+void WASAPIAudioInputStream::SetAutomaticGainControl(bool enabled) {

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

seems like we have a lot of similar or exactly the same code here as for the
output stream. Can't we have that code in one place somehow?

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

I feel the same way. Let me try to come up with something.

+  base::subtle::Release_Store(&agc_is_enabled_, enabled);
+}
+
+bool WASAPIAudioInputStream::GetAutomaticGainControl() {
+  return (base::subtle::Acquire_Load(&agc_is_enabled_) == 1);
+}
+
 // static
 double WASAPIAudioInputStream::HardwareSampleRate(
     const std::string& device_id) {
@@ -389,6 +412,21 @@ void WASAPIAudioInputStream::Run() {
                 first_audio_frame_timestamp) / 10000.0) * ms_to_frame_count_ +
                 buffer_frame_index - num_frames_to_read;
 
+          // Query a new volume level if AGC is enabled and if it is time to
+          // update the old value.
+          if (GetAutomaticGainControl()) {

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

this code looks the same too

+            base::Time now = base::Time::Now();
+            if ((now - last_volume_update_time_).InMilliseconds() >
+                kMinIntervalBetweenVolumeUpdatesMs) {
+              // Retrieve the current volume level. Range is [0.0,1.0].
+              float level = 0.0f;
+              HRESULT hr = simple_audio_volume_->GetMasterVolume(&level);
+              if (SUCCEEDED(hr))
+                volume_ = static_cast<double>(level);
+              last_volume_update_time_ = now;
+            }
+          }
+
           // Deliver captured data to the registered consumer using a packet
           // size which was specified at construction.
           uint32 delay_frames = static_cast<uint32>(audio_delay_frames + 0.5);
@@ -396,11 +434,13 @@ void WASAPIAudioInputStream::Run() {
             uint8* audio_data =
                 reinterpret_cast<uint8*>(capture_buffer.get());
 
-            // Deliver data packet and delay estimation to the user.
+            // Deliver data packet, delay estimation and volume level to
+            // the user.
             sink_->OnData(this,
                           audio_data,
                           packet_size_bytes_,
-                          delay_frames * frame_size_);
+                          delay_frames * frame_size_,
+                          volume_);
 
             // Store parts of the recorded data which can't be delivered
             // using the current packet size. The stored section will be used