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

media/audio/win/audio_low_latency_input_win_unittest.cc

 // Copyright (c) 2012 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 <windows.h>
 #include <mmsystem.h>
 
 #include "base/basictypes.h"
 #include "base/environment.h"
 #include "base/file_util.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/message_loop.h"
 #include "base/path_service.h"
 #include "base/test/test_timeouts.h"
 #include "base/win/scoped_com_initializer.h"
 #include "media/audio/audio_io.h"
 #include "media/audio/audio_manager_base.h"
 #include "media/audio/win/audio_low_latency_input_win.h"
 #include "media/base/seekable_buffer.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 using base::win::ScopedCOMInitializer;
+using ::testing::_;
 using ::testing::AnyNumber;
 using ::testing::AtLeast;
 using ::testing::Gt;
 using ::testing::NotNull;
 
 ACTION_P3(CheckCountAndPostQuitTask, count, limit, loop) {
   if (++*count >= limit) {
     loop->PostTask(FROM_HERE, MessageLoop::QuitClosure());
   }
 }
 
 class MockAudioInputCallback : public AudioInputStream::AudioInputCallback {
  public:
-  MOCK_METHOD4(OnData, void(AudioInputStream* stream,
+  MOCK_METHOD5(OnData, void(AudioInputStream* stream,
       const uint8* src, uint32 size,
-      uint32 hardware_delay_bytes));
+      uint32 hardware_delay_bytes, double volume));
   MOCK_METHOD1(OnClose, void(AudioInputStream* stream));
   MOCK_METHOD2(OnError, void(AudioInputStream* stream, int code));
 };
 
 // This audio sink implementation should be used for manual tests only since
 // the recorded data is stored on a raw binary data file.
 class WriteToFileAudioSink : public AudioInputStream::AudioInputCallback {
  public:
   // Allocate space for ~10 seconds of data @ 48kHz in stereo:
   // 2 bytes per sample, 2 channels, 10ms @ 48kHz, 10 seconds <=> 1920000 bytes.
@@ skipping 26 matching lines @@
       buffer_.Seek(chunk_size);
       bytes_written += chunk_size;
     }
     file_util::CloseFile(binary_file_);
   }
 
   // AudioInputStream::AudioInputCallback implementation.
   virtual void OnData(AudioInputStream* stream,
                       const uint8* src,
                       uint32 size,
-                      uint32 hardware_delay_bytes) {
+                      uint32 hardware_delay_bytes,
+                      double volume) {
     // Store data data in a temporary buffer to avoid making blocking
     // fwrite() calls in the audio callback. The complete buffer will be
     // written to file in the destructor.
     if (buffer_.Append(src, size)) {
       bytes_to_write_ += size;
     }
   }
 
   virtual void OnClose(AudioInputStream* stream) {}
   virtual void OnError(AudioInputStream* stream, int code) {}
@@ skipping 202 matching lines @@
   MockAudioInputCallback sink;
 
   // Derive the expected size in bytes of each recorded packet.
   uint32 bytes_per_packet = aisw.channels() * aisw.samples_per_packet() *
       (aisw.bits_per_sample() / 8);
 
   // We use 10ms packets and will run the test until ten packets are received.
   // All should contain valid packets of the same size and a valid delay
   // estimate.
   EXPECT_CALL(sink, OnData(
-      ais, NotNull(), bytes_per_packet, Gt(bytes_per_packet)))
+      ais, NotNull(), bytes_per_packet, Gt(bytes_per_packet), _))
       .Times(AtLeast(10))
       .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop));
   ais->Start(&sink);
   loop.Run();
   ais->Stop();
 
   // Store current packet size (to be used in the subsequent tests).
   int samples_per_packet_10ms = aisw.samples_per_packet();
 
   EXPECT_CALL(sink, OnClose(ais))
       .Times(1);
   ais->Close();
 
   // 20 ms packet size.
 
   count = 0;
   ais = aisw.Create(2 * samples_per_packet_10ms);
   EXPECT_TRUE(ais->Open());
   bytes_per_packet = aisw.channels() * aisw.samples_per_packet() *
       (aisw.bits_per_sample() / 8);
 
   EXPECT_CALL(sink, OnData(
-      ais, NotNull(), bytes_per_packet, Gt(bytes_per_packet)))
+      ais, NotNull(), bytes_per_packet, Gt(bytes_per_packet), _))
       .Times(AtLeast(10))
       .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop));
   ais->Start(&sink);
   loop.Run();
   ais->Stop();
 
   EXPECT_CALL(sink, OnClose(ais))
       .Times(1);
   ais->Close();
 
   // 5 ms packet size.
 
   count = 0;
   ais = aisw.Create(samples_per_packet_10ms / 2);
   EXPECT_TRUE(ais->Open());
   bytes_per_packet = aisw.channels() * aisw.samples_per_packet() *
     (aisw.bits_per_sample() / 8);
 
   EXPECT_CALL(sink, OnData(
-      ais, NotNull(), bytes_per_packet, Gt(bytes_per_packet)))
+      ais, NotNull(), bytes_per_packet, Gt(bytes_per_packet), _))
       .Times(AtLeast(10))
       .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop));
   ais->Start(&sink);
   loop.Run();
   ais->Stop();
 
   EXPECT_CALL(sink, OnClose(ais))
       .Times(1);
   ais->Close();
 }
@@ skipping 20 matching lines @@
 
   LOG(INFO) << ">> Sample rate: " << aisw.sample_rate() << " [Hz]";
   WriteToFileAudioSink file_sink(file_name);
   LOG(INFO) << ">> Speak into the default microphone while recording.";
   ais->Start(&file_sink);
   base::PlatformThread::Sleep(TestTimeouts::action_timeout_ms());
   ais->Stop();
   LOG(INFO) << ">> Recording has stopped.";
   ais->Close();
 }