Adding experimental exclusive-mode streaming to WASAPIAudioOutputStream

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.
 
 // TODO(vrk): This file has been running pretty wild and free, and it's likely
 // that a lot of the functions can be simplified and made more elegant. Revisit
 // after other audio cleanup is done. (crbug.com/120319)
 
 #include <algorithm>
 #include <limits>
 
 #include "base/atomicops.h"
 #include "base/basictypes.h"
 #include "base/logging.h"
 #include "base/shared_memory.h"
 #include "base/time.h"
-#if defined(OS_WIN)
+#include "media/audio/audio_parameters.h"
+#include "media/audio/audio_util.h"
+
+#if defined(OS_MACOSX)
+#include "media/audio/mac/audio_low_latency_input_mac.h"
+#include "media/audio/mac/audio_low_latency_output_mac.h"
+#elif defined(OS_WIN)
+#include "base/command_line.h"
 #include "base/sys_info.h"
 #include "base/win/windows_version.h"
 #include "media/audio/audio_manager_base.h"
-#endif
-#include "media/audio/audio_parameters.h"
-#include "media/audio/audio_util.h"
-#if defined(OS_MACOSX)
-#include "media/audio/mac/audio_low_latency_input_mac.h"
-#include "media/audio/mac/audio_low_latency_output_mac.h"
-#endif
-#if defined(OS_WIN)
 #include "media/audio/win/audio_low_latency_input_win.h"
 #include "media/audio/win/audio_low_latency_output_win.h"
+#include "media/base/media_switches.h"
 #endif
 
 using base::subtle::Atomic32;
 
 const uint32 kUnknownDataSize = static_cast<uint32>(-1);
 
 namespace media {
 
 // TODO(fbarchard): Convert to intrinsics for better efficiency.
 template<class Fixed>
@@ skipping 297 matching lines @@
 #if defined(OS_MACOSX)
   // Hardware sample-rate on the Mac can be configured, so we must query.
   return AUAudioOutputStream::HardwareSampleRate();
 #elif defined(OS_WIN)
   if (!IsWASAPISupported()) {
     // Fall back to Windows Wave implementation on Windows XP or lower
     // and use 48kHz as default input sample rate.
     return 48000;
   }
 
+  // TODO(crogers): tune this rate for best possible WebAudio performance.
+  // WebRTC works well at 48kHz and a buffer size of 480 samples will be used
+  // for this case. Note that exclusive mode is experimental.
+  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 audio endpoint.
-  // Use the default device (same as for Wave) for now to be compatible.
+  // 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);
 #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;
+  // 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) {
   // TODO(henrika): add support for device selection on all platforms.
   // Only exists on Windows today.
 #if defined(OS_MACOSX)
   return AUAudioInputStream::HardwareSampleRate();
 #elif defined(OS_WIN)
   if (!IsWASAPISupported()) {
@@ skipping 16 matching lines @@
   // 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)
   if (!IsWASAPISupported()) {
     // Fall back to Windows Wave implementation on Windows XP or lower
     // and assume 48kHz as default sample rate.
     return 2048;
   }
+
+  // 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;
+  }
+
   // 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);
   if (mixing_sample_rate == 48000)
     return 480;
   else if (mixing_sample_rate == 44100)
     return 448;
   else
     return 960;
@@ skipping 120 matching lines @@
   if ((base::SysInfo::NumberOfProcessors() < 2) ||
       (base::win::GetVersion() == base::win::VERSION_VISTA)) {
     return 3;
   }
   return 2;
 }
 
 #endif
 
 }  // namespace media