Increase Windows XP hardware buffer size to 4096.

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 "media/audio/audio_util.h"
 
 #include <algorithm>
 #include <limits>
 
 #include "base/basictypes.h"
 #include "base/logging.h"
+#include "base/string_number_conversions.h"
 #include "base/time.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.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/win/windows_version.h"
 #include "media/audio/audio_manager_base.h"
 #include "media/audio/win/audio_low_latency_input_win.h"
 #include "media/audio/win/audio_low_latency_output_win.h"
 #include "media/audio/win/audio_unified_win.h"
 #include "media/base/limits.h"
 #include "media/base/media_switches.h"
 #endif
 
 namespace media {
 
+// Returns user buffer size as specified on the command line or 0 if no buffer
+// size has been specified.
+static int GetUserBufferSize() {
+  const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
+  int buffer_size = 0;
+  std::string buffer_size_str(cmd_line->GetSwitchValueASCII(
+      switches::kAudioBufferSize));
+  if (!buffer_size_str.empty() &&

[scherkus (not reviewing), 2012/10/29 18:23:02]

nit: base::StringToInt() will return 0 on empty string

up to you if you want to roll everything into a single statement

[DaleCurtis, 2012/10/30 21:52:52]

Done.

+      base::StringToInt(buffer_size_str, &buffer_size) && buffer_size > 0) {
+    return buffer_size;
+  }
+
+  return 0;
+}
+
 // TODO(fbarchard): Convert to intrinsics for better efficiency.
 template<class Fixed>
 static int ScaleChannel(int channel, int volume) {
   return static_cast<int>((static_cast<Fixed>(channel) * volume) >> 16);
 }
 
 template<class Format, class Fixed, int bias>
 static void AdjustVolume(Format* buf_out,
                          int sample_count,
                          int fixed_volume) {
@@ skipping 166 matching lines @@
   }
   return WASAPIAudioInputStream::HardwareSampleRate(device_id);
 #elif defined(OS_ANDROID)
   return 16000;
 #else
   return 48000;
 #endif
 }
 
 size_t GetAudioHardwareBufferSize() {
+  int user_buffer_size = GetUserBufferSize();
+  if (user_buffer_size)
+    return user_buffer_size;
+
   // 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)
   // Buffer size to use when a proper size can't be determined from the system.
-  static const int kFallbackBufferSize = 2048;
+  static const int kFallbackBufferSize = 4096;

[scherkus (not reviewing), 2012/10/29 18:23:02]

henrika: this is the buffer size change

[henrika (OOO until Aug 14), 2012/10/29 18:50:33]

Got it. Thanks.

 
   if (!IsWASAPISupported()) {
     // 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.
@@ skipping 57 matching lines @@
   return WASAPIAudioInputStream::HardwareChannelCount(device_id) == 1 ?
       CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
 #else
   return CHANNEL_LAYOUT_STEREO;
 #endif
 }
 
 // Computes a buffer size based on the given |sample_rate|. Must be used in
 // conjunction with AUDIO_PCM_LINEAR.
 size_t GetHighLatencyOutputBufferSize(int sample_rate) {
+  int user_buffer_size = GetUserBufferSize();
+  if (user_buffer_size)
+    return user_buffer_size;
+
   // TODO(vrk/crogers): The buffer sizes that this function computes is probably
   // overly conservative. However, reducing the buffer size to 2048-8192 bytes
   // caused crbug.com/108396. This computation should be revisited while making
   // sure crbug.com/108396 doesn't happen again.
 
   // The minimum number of samples in a hardware packet.
   // This value is selected so that we can handle down to 5khz sample rate.
   static const size_t kMinSamplesPerHardwarePacket = 1024;
 
   // The maximum number of samples in a hardware packet.
@@ skipping 30 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