Add basic resample support to WASAPIAudioInputStream.

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 22355580aac45b2c656ae6066bd25fd1b72f1a3f..cfecc91cc41ddbbcca6d5c8d652684b8a31dca6a 100644
--- a/media/audio/win/audio_low_latency_input_win.cc
+++ b/media/audio/win/audio_low_latency_input_win.cc
@@ -15,6 +15,7 @@
 #include "media/audio/win/avrt_wrapper_win.h"
 #include "media/audio/win/core_audio_util_win.h"
 #include "media/base/audio_bus.h"
+#include "media/base/multi_channel_resampler.h"
 
 using base::win::ScopedComPtr;
 using base::win::ScopedCOMInitializer;
@@ -123,9 +124,10 @@ bool WASAPIAudioInputStream::Open() {
   // Initialize the audio stream between the client and the device using
   // shared mode and a lowest possible glitch-free latency.
   hr = InitializeAudioEngine();
+  if (SUCCEEDED(hr) && resampler_.get() != nullptr)

[DaleCurtis, 2017/02/13 19:40:42]

just "&& resampler_" it has safe-bool conversion built in these days.

[tommi (sloooow) - chröme, 2017/02/16 17:08:46]

Done.

+    open_result_ = OPEN_RESULT_OK_WITH_RESAMPLING;
   ReportOpenResult();  // Report before we assign a value to |opened_|.
   opened_ = SUCCEEDED(hr);
-  DCHECK(open_result_ == OPEN_RESULT_OK || !opened_);
 
   return opened_;
 }
@@ -424,8 +426,15 @@ void WASAPIAudioInputStream::Run() {
           // Copy data to audio bus to match the OnData interface.
           uint8_t* audio_data =
               reinterpret_cast<uint8_t*>(capture_buffer.get());
-          audio_bus_->FromInterleaved(audio_data, audio_bus_->frames(),
-                                      format_.wBitsPerSample / 8);
+
+          if (resampler_.get()) {

[DaleCurtis, 2017/02/13 19:40:42]

Ditto.

[tommi (sloooow) - chröme, 2017/02/16 17:08:46]

Done.

+            resample_bus_->FromInterleaved(audio_data, resample_bus_->frames(),
+                                           format_.wBitsPerSample / 8);
+            resampler_->Resample(audio_bus_->frames(), audio_bus_.get());
+          } else {
+            audio_bus_->FromInterleaved(audio_data, audio_bus_->frames(),
+                                        format_.wBitsPerSample / 8);
+          }
 
           // Deliver data packet, delay estimation and volume level to
           // the user.
@@ -595,6 +604,44 @@ bool WASAPIAudioInputStream::DesiredFormatIsSupported() {
                                                 &format_, &closest_match);
   DLOG_IF(ERROR, hr == S_FALSE) << "Format is not supported "
                                 << "but a closest match exists.";
+  if (hr == S_FALSE && closest_match->nChannels == format_.nChannels) {

[DaleCurtis, 2017/02/13 19:40:42]

If you use a AudioConverter it'll handle mismatched channel counts for you.

[DaleCurtis, 2017/02/13 23:59:00]

You should also sanity check the closest match sample rate here. I.e. that it
isn't 0 or something outside the bound of our limits::kMin/MaxSampleRate.

[tommi (sloooow) - chröme, 2017/02/16 17:08:46]

Done. Also switched to AudioConverter.

+    // We want a 1:1 ratio between the buffers we get and the buffers we
+    // give to OnData so that each buffer we receive from the OS can be directly
+    // resampled to a buffer that matches with what the client asked for.
+    const double buffer_ratio =
+        format_.nSamplesPerSec / static_cast<double>(audio_bus_->frames());
+    const size_t new_frames_per_buffer =
+        static_cast<size_t>(closest_match->nSamplesPerSec / buffer_ratio);
+
+    const double sample_rate_ratio =
+        static_cast<double>(closest_match->nSamplesPerSec) /
+        static_cast<double>(format_.nSamplesPerSec);
+    resampler_.reset(new MultiChannelResampler(

[DaleCurtis, 2017/02/13 19:40:42]

I think this works, though not how we've done it in the past see
AudioBufferConverter. Will take a closer look later, have to run now.

[DaleCurtis, 2017/02/13 23:59:00]

Over a long enough time frame I think this might lose a sample every now and
then. I.e. Resample() will consume all but the last frame and then the next
FromInterleaved() call will overwrite it. To make this correct I think you need
to use the AudioBlockFifo as suggested above.

E.g., if the desired is 44.1kHz/1024 frames and the closest match is 16kHz, I
think you'll lose the fractional part on occasion.

[tommi (sloooow) - chröme, 2017/02/16 17:08:47]

I tried several ways of reproducing that, including recording at 16kHz and
converting to 44.1kHz/1024 for 10 minutes, but was unable to reproduce the
problem (ProvideInput was called for all buffers).  Any further hints on how to
test that?

[tommi (sloooow) - chröme, 2017/02/16 17:09:49]

Oh please ignore this, I did more testing and figured out what I was missing.

+        format_.nChannels, sample_rate_ratio, new_frames_per_buffer,
+        base::Bind(&WASAPIAudioInputStream::ProvideInput,
+                   base::Unretained(this))));
+    resampler_->PrimeWithSilence();
+    resample_bus_ = AudioBus::Create(format_.nChannels, new_frames_per_buffer);
+
+    // Now change the format we're going to ask for to better match with what
+    // the OS can provide.  If we succeed in opening the stream with these
+    // params, we can take care of the required resampling.
+    format_.nSamplesPerSec = closest_match->nSamplesPerSec;
+    format_.nAvgBytesPerSec = format_.nSamplesPerSec * format_.nBlockAlign;
+
+    // Update our packet size assumptions based on the new format.
+    const auto new_bytes_per_buffer = resample_bus_->frames() *
+                                      closest_match->nChannels *
+                                      (format_.wBitsPerSample / 8);
+    packet_size_frames_ = new_bytes_per_buffer / format_.nBlockAlign;
+    packet_size_bytes_ = new_bytes_per_buffer;
+    ms_to_frame_count_ =
+        static_cast<double>(closest_match->nSamplesPerSec) / 1000.0;
+
+    // Indicate that we're good to go with a close match.
+    hr = S_OK;
+  }
+
   return (hr == S_OK);
 }
 
@@ -738,4 +785,9 @@ void WASAPIAudioInputStream::ReportOpenResult() const {
                             OPEN_RESULT_MAX + 1);
 }
 
+void WASAPIAudioInputStream::ProvideInput(int resampler_frame_delay,
+                                          AudioBus* audio_bus) {
+  resample_bus_->CopyTo(audio_bus);

[DaleCurtis, 2017/02/13 19:40:42]

Probably want a DCHECK this isn't called twice without resample_bus_ being
refilled.

[tommi (sloooow) - chröme, 2017/02/16 17:08:47]

Done.

+}
+
 }  // namespace media