Avoids irregular OnMoreData callbacks on Windows using Core Audio

media/audio/audio_util.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.
 
 // Software adjust volume of samples, allows each audio stream its own
 // volume without impacting master volume for chrome and other applications.
 
 // Implemented as templates to allow 8, 16 and 32 bit implementations.
 // 8 bit is unsigned and biased by 128.
 
@@ skipping 194 matching lines @@
   const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
   if (cmd_line->HasSwitch(switches::kEnableExclusiveAudio)) {
     // This sample rate will be combined with a buffer size of 256 samples
     // (see GetAudioHardwareBufferSize()), which corresponds to an output
     // delay of ~5.33ms.
     return 48000;
   }
 
   // Hardware sample-rate on Windows can be configured, so we must query.
   // TODO(henrika): improve possibility to specify an audio endpoint.
-  // Use the default device (same as for Wave) for now to be compatible
-  // or possibly remove the ERole argument completely until it is in use.
-  return WASAPIAudioOutputStream::HardwareSampleRate(eConsole);
+  // Use the default device (same as for Wave) for now to be compatible.
+  return WASAPIAudioOutputStream::HardwareSampleRate();
 #elif defined(OS_ANDROID)
   return 16000;
 #else
   // Hardware for Linux is nearly always 48KHz.
   // TODO(crogers) : return correct value in rare non-48KHz cases.
   return 48000;
 #endif
 }
 
 int GetAudioInputHardwareSampleRate(const std::string& device_id) {
@@ skipping 21 matching lines @@
   // The sizes here were determined by experimentation and are roughly
   // the lowest value (for low latency) that still allowed glitch-free
   // audio under high loads.
   //
   // For Mac OS X and Windows the chromium audio backend uses a low-latency
   // Core Audio API, so a low buffer size is possible. For Linux, further
   // tuning may be needed.
 #if defined(OS_MACOSX)
   return 128;
 #elif defined(OS_WIN)
+  // TODO(henrika): resolve conflict with GetUserBufferSize().
+  // If the user tries to set a buffer size using GetUserBufferSize() it will
+  // most likely fail since only the native/perfect buffer size is allowed.
+
   // Buffer size to use when a proper size can't be determined from the system.
   static const int kFallbackBufferSize = 4096;
 
   if (!CoreAudioUtil::IsSupported()) {
     // Fall back to Windows Wave implementation on Windows XP or lower
     // and assume 48kHz as default sample rate.
     return kFallbackBufferSize;
   }
 
   // TODO(crogers): tune this size to best possible WebAudio performance.
   // WebRTC always uses 10ms for Windows and does not call this method.
   // Note that exclusive mode is experimental.
   const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
   if (cmd_line->HasSwitch(switches::kEnableExclusiveAudio)) {
     return 256;
   }
 
-  // TODO(henrika): remove when the --enable-webaudio-input flag is no longer
-  // utilized.
-  if (cmd_line->HasSwitch(switches::kEnableWebAudioInput)) {
-    AudioParameters params;
-    HRESULT hr = CoreAudioUtil::GetPreferredAudioParameters(eRender, eConsole,
-                                                            &params);
-    return FAILED(hr) ? kFallbackBufferSize : params.frames_per_buffer();
-  }
-
-  // This call must be done on a COM thread configured as MTA.
-  // TODO(tommi): http://code.google.com/p/chromium/issues/detail?id=103835.
-  int mixing_sample_rate =
-      WASAPIAudioOutputStream::HardwareSampleRate(eConsole);
-
-  // Windows will return a sample rate of 0 when no audio output is available
-  // (i.e. via RemoteDesktop with remote audio disabled), but we should never
-  // return a buffer size of zero.
-  if (mixing_sample_rate == 0)
-    return kFallbackBufferSize;
-
-  // Use different buffer sizes depening on the sample rate . The existing
-  // WASAPI implementation is tuned to provide the most stable callback
-  // sequence using these combinations.
-  if (mixing_sample_rate % 11025 == 0)
-    // Use buffer size of ~10.15873 ms.
-    return (112 * (mixing_sample_rate / 11025));
-
-  if (mixing_sample_rate % 8000 == 0)
-    // Use buffer size of 10ms.
-    return (80 * (mixing_sample_rate / 8000));
-
-  // Ensure we always return a buffer size which is somewhat appropriate.
-  LOG(ERROR) << "Unknown sample rate " << mixing_sample_rate << " detected.";
-  if (mixing_sample_rate > limits::kMinSampleRate)
-    return (mixing_sample_rate / 100);
-  return kFallbackBufferSize;
+  AudioParameters params;
+  HRESULT hr = CoreAudioUtil::GetPreferredAudioParameters(eRender, eConsole,
+                                                          &params);
+  return FAILED(hr) ? kFallbackBufferSize : params.frames_per_buffer();
 #else
   return 2048;
 #endif
 }
 
 ChannelLayout GetAudioInputHardwareChannelLayout(const std::string& device_id) {
   // TODO(henrika): add support for device selection on all platforms.
   // Only exists on Windows today.
 #if defined(OS_MACOSX)
   return CHANNEL_LAYOUT_MONO;
@@ skipping 54 matching lines @@
   //    out performance was degraded compared to XP.
   //  - The regression was fixed in Windows 7 and most configurations will work
   //    with 2, but some (e.g., some Sound Blasters) still need 3.
   //  - Some XP configurations (even multi-processor ones) also need 3.
   return (base::win::GetVersion() == base::win::VERSION_VISTA) ? 4 : 3;
 }
 
 #endif
 
 }  // namespace media