Added support for multiple sampling rates on Windows.

remoting/host/audio_capturer_win.cc

Index: remoting/host/audio_capturer_win.cc
diff --git a/remoting/host/audio_capturer_win.cc b/remoting/host/audio_capturer_win.cc
index 55bc0855b887d8a05e8a870f8e0093da4fb11942..1d909ff4c6808ca0bc0c56614f4ccef6ed29088c 100644
--- a/remoting/host/audio_capturer_win.cc
+++ b/remoting/host/audio_capturer_win.cc
@@ -23,7 +23,6 @@
 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;
@@ -48,6 +47,8 @@ class AudioCapturerWin : public AudioCapturer {
 
   PacketCapturedCallback callback_;
 
+  AudioPacket::SamplingRate sampling_rate_;
+
   scoped_ptr<base::RepeatingTimer<AudioCapturerWin> > capture_timer_;
   base::TimeDelta audio_device_period_;
 
@@ -62,7 +63,8 @@ class AudioCapturerWin : public AudioCapturer {
   DISALLOW_COPY_AND_ASSIGN(AudioCapturerWin);
 };
 
-AudioCapturerWin::AudioCapturerWin() {
+AudioCapturerWin::AudioCapturerWin()
+  : sampling_rate_(AudioPacket::SAMPLING_RATE_INVALID) {

[Sergey Ulanov, 2012/07/26 03:40:24]

nit: initializer list should be indented 4 spaces (6 spaces after the first
line).

[kxing, 2012/07/26 22:47:17]

Done.

   thread_checker_.DetachFromThread();
 }
 
@@ -131,13 +133,20 @@ bool AudioCapturerWin::Start(const PacketCapturedCallback& callback) {
     case WAVE_FORMAT_IEEE_FLOAT:
       // Intentional fall-through.
     case WAVE_FORMAT_PCM:
+      if (!AudioCapturer::IsValidSampleRate(wave_format_ex_->nSamplesPerSec)) {

[Wez, 2012/07/25 09:23:48]

Rather than have a static test for validity, why not pass a set of acceptable
capture rates to the AudioCapturer when it's started - we can then still
hard-code specific rates e.g. into the AudioScheduler, but we also have the
option of feeding back supported playback rates from the connected client.

[kxing, 2012/07/26 22:47:17]

The set of valid sampling rates is the same across all platforms (since they are
the sampling rates supported by Pepper). I'm not sure if the extra flexibility
of passed-in playback rates is worth it.

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

I discussed this with Kerry. This might be useful in the future, but there is no
point in doing it in this CL - we can add it later when we need it.

+        LOG(ERROR) << "Host sampling rate is neither 44.1 kHz nor 48 kHz.";
+        return false;

[Sergey Ulanov, 2012/07/26 03:40:24]

Can we still try to request some known sampling rate?

[kxing, 2012/07/26 22:47:17]

No, Pepper only supports 44.1 kHz and 48 kHz, and Windows only allows sampling
at the current output rate. Thus, if the output rate is neither 44.1 nor 48 kHz,
we'll have to resample.

+      }
+      sampling_rate_ =
+          static_cast<AudioPacket::SamplingRate>(
+          wave_format_ex_->nSamplesPerSec);
+
       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 =
@@ -145,16 +154,25 @@ bool AudioCapturerWin::Start(const PacketCapturedCallback& callback) {
           static_cast<WAVEFORMATEX*>(wave_format_ex_));
       if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT,
                       wave_format_extensible->SubFormat)) {
+        if (!AudioCapturer::IsValidSampleRate(
+            wave_format_extensible->Format.nSamplesPerSec)) {

[Sergey Ulanov, 2012/07/26 03:40:24]

nit: incorrect indentation. move it four spaces to the right.

[kxing, 2012/07/26 22:47:17]

Done.

+          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);

[Sergey Ulanov, 2012/07/26 03:40:24]

nit: indent 4 spaces relative to the previous line. Or you can fit this
statement in two lines.

[kxing, 2012/07/26 22:47:17]

Done.

+
         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;
@@ -221,6 +239,7 @@ bool AudioCapturerWin::IsRunning() {
 }
 
 void AudioCapturerWin::DoCapture() {
+  DCHECK(AudioCapturer::IsValidSampleRate(sampling_rate_));
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(IsRunning());
 
@@ -249,6 +268,7 @@ void AudioCapturerWin::DoCapture() {
 
     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());