Modifies AudioInputCallback::OnData and use media::AudioBus instead of plain byte vector (Relanding)

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"
@@ skipping 20 matching lines @@
 namespace media {
 
 ACTION_P3(CheckCountAndPostQuitTask, count, limit, loop) {
   if (++*count >= limit) {
     loop->PostTask(FROM_HERE, base::MessageLoop::QuitClosure());
   }
 }
 
 class MockAudioInputCallback : public AudioInputStream::AudioInputCallback {
  public:
-  MOCK_METHOD5(OnData, void(AudioInputStream* stream,
-                            const uint8* src, uint32 size,
-                            uint32 hardware_delay_bytes, double volume));
+  MOCK_METHOD4(OnData,
+               void(AudioInputStream* stream,
+                    const AudioBus* src,
+                    uint32 hardware_delay_bytes,
+                    double volume));
   MOCK_METHOD1(OnError, void(AudioInputStream* stream));
 };
 
 class FakeAudioInputCallback : public AudioInputStream::AudioInputCallback {
  public:
   FakeAudioInputCallback()
-    : error_(false),
-      data_event_(false, false) {
-  }
+      : error_(false),
+        data_event_(false, false),
+        num_received_audio_frames_(0) {}
 
-  const std::vector<uint8>& received_data() const { return received_data_; }
   bool error() const { return error_; }
+  int num_received_audio_frames() const { return num_received_audio_frames_; }
 
   // Waits until OnData() is called on another thread.
   void WaitForData() {
     data_event_.Wait();
   }
 
   virtual void OnData(AudioInputStream* stream,
-                      const uint8* src, uint32 size,
-                      uint32 hardware_delay_bytes, double volume) OVERRIDE {
-    received_data_.insert(received_data_.end(), src, src + size);
+                      const AudioBus* src,
+                      uint32 hardware_delay_bytes,
+                      double volume) OVERRIDE {
+    EXPECT_NE(hardware_delay_bytes, 0u);
+    num_received_audio_frames_ += src->frames();
     data_event_.Signal();
   }
 
   virtual void OnError(AudioInputStream* stream) OVERRIDE {
     error_ = true;
   }
 
  private:
-  std::vector<uint8> received_data_;
+  int num_received_audio_frames_;
   base::WaitableEvent data_event_;
   bool error_;
 
   DISALLOW_COPY_AND_ASSIGN(FakeAudioInputCallback);
 };
 
 // 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.
   static const size_t kMaxBufferSize = 2 * 2 * 480 * 100 * 10;
 
-  explicit WriteToFileAudioSink(const char* file_name)
-      : buffer_(0, kMaxBufferSize),
+  explicit WriteToFileAudioSink(const char* file_name, int bits_per_sample)
+      : bits_per_sample_(bits_per_sample),
+        buffer_(0, kMaxBufferSize),
         bytes_to_write_(0) {
     base::FilePath file_path;
     EXPECT_TRUE(PathService::Get(base::DIR_EXE, &file_path));
     file_path = file_path.AppendASCII(file_name);
     binary_file_ = base::OpenFile(file_path, "wb");
     DLOG_IF(ERROR, !binary_file_) << "Failed to open binary PCM data file.";
     VLOG(0) << ">> Output file: " << file_path.value() << " has been created.";
+    VLOG(0) << "bits_per_sample_:" << bits_per_sample_;
   }
 
   virtual ~WriteToFileAudioSink() {
     size_t bytes_written = 0;
     while (bytes_written < bytes_to_write_) {
       const uint8* chunk;
       int chunk_size;
 
       // Stop writing if no more data is available.
       if (!buffer_.GetCurrentChunk(&chunk, &chunk_size))
         break;
 
       // Write recorded data chunk to the file and prepare for next chunk.
       fwrite(chunk, 1, chunk_size, binary_file_);
       buffer_.Seek(chunk_size);
       bytes_written += chunk_size;
     }
     base::CloseFile(binary_file_);
   }
 
   // AudioInputStream::AudioInputCallback implementation.
   virtual void OnData(AudioInputStream* stream,
-                      const uint8* src,
-                      uint32 size,
+                      const AudioBus* src,
                       uint32 hardware_delay_bytes,
                       double volume) {
+    EXPECT_EQ(bits_per_sample_, 16);
+    const int num_samples = src->frames() * src->channels();
+    scoped_ptr<int16> interleaved(new int16[num_samples]);
+    const int bytes_per_sample = sizeof(*interleaved);
+    src->ToInterleaved(src->frames(), bytes_per_sample, interleaved.get());
+
     // 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)) {
+    const int size = bytes_per_sample * num_samples;
+    if (buffer_.Append((const uint8*)interleaved.get(), size)) {
       bytes_to_write_ += size;
     }
   }
 
   virtual void OnError(AudioInputStream* stream) {}
 
  private:
+  int bits_per_sample_;
   media::SeekableBuffer buffer_;
   FILE* binary_file_;
   size_t bytes_to_write_;
 };
 
 // Convenience method which ensures that we are not running on the build
 // bots and that at least one valid input device can be found. We also
 // verify that we are not running on XP since the low-latency (WASAPI-
 // based) version requires Windows Vista or higher.
 static bool CanRunAudioTests(AudioManager* audio_man) {
@@ skipping 224 matching lines @@
 
   MockAudioInputCallback sink;
 
   // Derive the expected size in bytes of each recorded packet.
   uint32 bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() *
       (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.get(), NotNull(), bytes_per_packet, Gt(bytes_per_packet), _))
+  EXPECT_CALL(sink, OnData(ais.get(), NotNull(), 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 frames_per_buffer_10ms = aisw.frames_per_buffer();
 
   ais.Close();
 
   // 20 ms packet size.
 
   count = 0;
   ais.Reset(aisw.Create(2 * frames_per_buffer_10ms));
   EXPECT_TRUE(ais->Open());
   bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() *
       (aisw.bits_per_sample() / 8);
 
-  EXPECT_CALL(sink, OnData(
-      ais.get(), NotNull(), bytes_per_packet, Gt(bytes_per_packet), _))
+  EXPECT_CALL(sink, OnData(ais.get(), NotNull(), Gt(bytes_per_packet), _))
       .Times(AtLeast(10))
       .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop));
   ais->Start(&sink);
   loop.Run();
   ais->Stop();
   ais.Close();
 
   // 5 ms packet size.
 
   count = 0;
   ais.Reset(aisw.Create(frames_per_buffer_10ms / 2));
   EXPECT_TRUE(ais->Open());
   bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() *
     (aisw.bits_per_sample() / 8);
 
-  EXPECT_CALL(sink, OnData(
-      ais.get(), NotNull(), bytes_per_packet, Gt(bytes_per_packet), _))
+  EXPECT_CALL(sink, OnData(ais.get(), NotNull(), Gt(bytes_per_packet), _))
       .Times(AtLeast(10))
       .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop));
   ais->Start(&sink);
   loop.Run();
   ais->Stop();
   ais.Close();
 }
 
-// Test that we can capture loopback stream.
+// Test that we can capture a stream in loopback.
 TEST(WinAudioInputTest, WASAPIAudioInputStreamLoopback) {
   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
   if (!audio_manager->HasAudioOutputDevices() || !CoreAudioUtil::IsSupported())
     return;
 
   AudioParameters params = audio_manager->GetInputStreamParameters(
       AudioManagerBase::kLoopbackInputDeviceId);
   EXPECT_EQ(params.effects(), 0);
 
   AudioParameters output_params =
       audio_manager->GetOutputStreamParameters(std::string());
   EXPECT_EQ(params.sample_rate(), output_params.sample_rate());
   EXPECT_EQ(params.channel_layout(), output_params.channel_layout());
 
   ScopedAudioInputStream stream(audio_manager->MakeAudioInputStream(
       params, AudioManagerBase::kLoopbackInputDeviceId));
   ASSERT_TRUE(stream->Open());
   FakeAudioInputCallback sink;
   stream->Start(&sink);
   ASSERT_FALSE(sink.error());
 
   sink.WaitForData();
   stream.Close();
 
-  EXPECT_FALSE(sink.received_data().empty());
+  EXPECT_GT(sink.num_received_audio_frames(), 0);
   EXPECT_FALSE(sink.error());
 }
 
 // This test is intended for manual tests and should only be enabled
 // when it is required to store the captured data on a local file.
 // By default, GTest will print out YOU HAVE 1 DISABLED TEST.
 // To include disabled tests in test execution, just invoke the test program
 // with --gtest_also_run_disabled_tests or set the GTEST_ALSO_RUN_DISABLED_TESTS
 // environment variable to a value greater than 0.
 TEST(WinAudioInputTest, DISABLED_WASAPIAudioInputStreamRecordToFile) {
   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
   if (!CanRunAudioTests(audio_manager.get()))
     return;
 
   // Name of the output PCM file containing captured data. The output file
   // will be stored in the directory containing 'media_unittests.exe'.
   // Example of full name: \src\build\Debug\out_stereo_10sec.pcm.
   const char* file_name = "out_stereo_10sec.pcm";
 
   AudioInputStreamWrapper aisw(audio_manager.get());
   ScopedAudioInputStream ais(aisw.Create());
   EXPECT_TRUE(ais->Open());
 
   VLOG(0) << ">> Sample rate: " << aisw.sample_rate() << " [Hz]";
-  WriteToFileAudioSink file_sink(file_name);
+  WriteToFileAudioSink file_sink(file_name, aisw.bits_per_sample());
   VLOG(0) << ">> Speak into the default microphone while recording.";
   ais->Start(&file_sink);
   base::PlatformThread::Sleep(TestTimeouts::action_timeout());
   ais->Stop();
   VLOG(0) << ">> Recording has stopped.";
   ais.Close();
 }
 
 }  // namespace media