Added support for multiple sampling rates on Windows.

remoting/host/audio_capturer_win.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 <audioclient.h>
 #include <avrt.h>
 #include <mmdeviceapi.h>
 #include <mmreg.h>
 #include <mmsystem.h>
 
 #include "base/basictypes.h"
 #include "base/logging.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/message_loop.h"
 #include "base/timer.h"
 #include "base/win/scoped_co_mem.h"
 #include "base/win/scoped_com_initializer.h"
 #include "base/win/scoped_comptr.h"
 #include "remoting/host/audio_capturer.h"
 #include "remoting/proto/audio.pb.h"
 
 namespace {
 const int kChannels = 2;
 const int kBitsPerSample = 16;
-const int kSamplesPerSecond = 44100;
 const int kBitsPerByte = 8;
 // Conversion factor from 100ns to 1ms.
 const int kHnsToMs = 10000;
 }  // namespace
 
 namespace remoting {
 
 class AudioCapturerWin : public AudioCapturer {
  public:
   AudioCapturerWin();
   virtual ~AudioCapturerWin();
 
   // AudioCapturer interface.
   virtual bool Start(const PacketCapturedCallback& callback) OVERRIDE;
   virtual void Stop() OVERRIDE;
   virtual bool IsRunning() OVERRIDE;
 
  private:
   // Receives all packets from the audio capture endpoint buffer and pushes them
   // to the network.
   void DoCapture();
 
   PacketCapturedCallback callback_;
 
+  AudioPacket::SamplingRate sampling_rate_;
+
   scoped_ptr<base::RepeatingTimer<AudioCapturerWin> > capture_timer_;
   base::TimeDelta audio_device_period_;
 
   base::win::ScopedCoMem<WAVEFORMATEX> wave_format_ex_;
   base::win::ScopedComPtr<IAudioCaptureClient> audio_capture_client_;
   base::win::ScopedComPtr<IAudioClient> audio_client_;
   base::win::ScopedComPtr<IMMDevice> mm_device_;
   scoped_ptr<base::win::ScopedCOMInitializer> com_initializer_;
 
   base::ThreadChecker thread_checker_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioCapturerWin);
 };
 
-AudioCapturerWin::AudioCapturerWin() {
-  thread_checker_.DetachFromThread();
+AudioCapturerWin::AudioCapturerWin()
+    : sampling_rate_(AudioPacket::SAMPLING_RATE_INVALID) {
+      thread_checker_.DetachFromThread();
 }
 
 AudioCapturerWin::~AudioCapturerWin() {
 }
 
 bool AudioCapturerWin::Start(const PacketCapturedCallback& callback) {
   DCHECK(!audio_capture_client_.get());
   DCHECK(!audio_client_.get());
   DCHECK(!mm_device_.get());
   DCHECK(static_cast<PWAVEFORMATEX>(wave_format_ex_) == NULL);
@@ skipping 47 matching lines @@
   if (FAILED(hr)) {
     LOG(ERROR) << "Failed to get WAVEFORMATEX. Error " << hr;
     return false;
   }
 
   // Set the wave format
   switch (wave_format_ex_->wFormatTag) {
     case WAVE_FORMAT_IEEE_FLOAT:
       // Intentional fall-through.
     case WAVE_FORMAT_PCM:
+      if (!AudioCapturer::IsValidSampleRate(wave_format_ex_->nSamplesPerSec)) {
+        LOG(ERROR) << "Host sampling rate is neither 44.1 kHz nor 48 kHz.";
+        return false;
+      }
+      sampling_rate_ =
+          static_cast<AudioPacket::SamplingRate>(
+          wave_format_ex_->nSamplesPerSec);

[Sergey Ulanov, 2012/07/27 04:22:38]

nit: incorrect indentation

[kxing, 2012/07/27 14:39:52]

Done.

+
       wave_format_ex_->wFormatTag = WAVE_FORMAT_PCM;
       wave_format_ex_->nChannels = kChannels;
-      wave_format_ex_->nSamplesPerSec = kSamplesPerSecond;
       wave_format_ex_->wBitsPerSample = kBitsPerSample;
       wave_format_ex_->nBlockAlign = kChannels * kBitsPerSample / kBitsPerByte;
       wave_format_ex_->nAvgBytesPerSec =
-          kSamplesPerSecond * kChannels * kBitsPerSample / kBitsPerByte;
+          sampling_rate_ * kChannels * kBitsPerSample / kBitsPerByte;
       break;
     case WAVE_FORMAT_EXTENSIBLE: {
       PWAVEFORMATEXTENSIBLE wave_format_extensible =
           reinterpret_cast<WAVEFORMATEXTENSIBLE*>(
           static_cast<WAVEFORMATEX*>(wave_format_ex_));
       if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT,
                       wave_format_extensible->SubFormat)) {
+        if (!AudioCapturer::IsValidSampleRate(
+                wave_format_extensible->Format.nSamplesPerSec)) {
+          LOG(ERROR) << "Host sampling rate is neither 44.1 kHz nor 48 kHz.";
+          return false;
+        }
+        sampling_rate_ = static_cast<AudioPacket::SamplingRate>(
+            wave_format_extensible->Format.nSamplesPerSec);
+
         wave_format_extensible->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
         wave_format_extensible->Samples.wValidBitsPerSample = kBitsPerSample;
 
         wave_format_extensible->Format.nChannels = kChannels;
-        wave_format_extensible->Format.nSamplesPerSec = kSamplesPerSecond;
+        wave_format_extensible->Format.nSamplesPerSec = sampling_rate_;
         wave_format_extensible->Format.wBitsPerSample = kBitsPerSample;
         wave_format_extensible->Format.nBlockAlign =
             kChannels * kBitsPerSample / kBitsPerByte;
         wave_format_extensible->Format.nAvgBytesPerSec =
-            kSamplesPerSecond * kChannels * kBitsPerSample / kBitsPerByte;
+            sampling_rate_ * kChannels * kBitsPerSample / kBitsPerByte;
       } else {
         LOG(ERROR) << "Failed to force 16-bit samples";
         return false;
       }
       break;
     }
     default:
-      LOG(ERROR) << "Failed to force 16-bit samples";
+      LOG(ERROR) << "Failed to force 16-bit PCM";
       return false;
   }
 
   // Initialize the IAudioClient.
   hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_SHARED,
                                  AUDCLNT_STREAMFLAGS_LOOPBACK,
                                  0,
                                  0,
                                  wave_format_ex_,
                                  NULL);
@@ skipping 38 matching lines @@
 
   thread_checker_.DetachFromThread();
 }
 
 bool AudioCapturerWin::IsRunning() {
   DCHECK(thread_checker_.CalledOnValidThread());
   return capture_timer_.get() != NULL;
 }
 
 void AudioCapturerWin::DoCapture() {
+  DCHECK(AudioCapturer::IsValidSampleRate(sampling_rate_));
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(IsRunning());
 
   // Fetch all packets from the audio capture endpoint buffer.
   while (true) {
     UINT32 next_packet_size;
     HRESULT hr = audio_capture_client_->GetNextPacketSize(&next_packet_size);
     if (FAILED(hr)) {
       LOG(ERROR) << "Failed to GetNextPacketSize. Error " << hr;
       return;
     }
 
     if (next_packet_size <= 0) {
       return;
     }
 
     BYTE* data;
     UINT32 frames;
     DWORD flags;
     hr = audio_capture_client_->GetBuffer(
         &data, &frames, &flags, NULL, NULL);
     if (FAILED(hr)) {
       LOG(ERROR) << "Failed to GetBuffer. Error " << hr;
       return;
     }
 
     scoped_ptr<AudioPacket> packet = scoped_ptr<AudioPacket>(new AudioPacket());
     packet->set_data(data, frames * wave_format_ex_->nBlockAlign);
+    packet->set_sampling_rate(sampling_rate_);
 
     callback_.Run(packet.Pass());
 
     hr = audio_capture_client_->ReleaseBuffer(frames);
     if (FAILED(hr)) {
       LOG(ERROR) << "Failed to ReleaseBuffer. Error " << hr;
       return;
     }
   }
 }
 
 scoped_ptr<AudioCapturer> AudioCapturer::Create() {
   return scoped_ptr<AudioCapturer>(new AudioCapturerWin());
 }
 
 }  // namespace remoting