Merge 123292 - Increase the buffer size in AudioRendererImpl to fix muted playback rate

This CL temporarily removes GetBufferSizeForSampleRate() and uses
SelectSamplesPerPacket() to determine the size of samples_per_packet in
AudioDevice. This restores the buffer size that AudioRendererImpl
used prior to r113821. SelectSamplesPerPacket() calculates a much larger
samples_per_packet value than GetBufferSizeForSampleRate(), so it fixes
the issue described in 108239.

This CL also does some mild refactoring to move audio_common.cc into
audio_util.cc.

BUG=108239
TEST=media_unittests,content_unittests, manual testing of different playback rates

Review URL: https://chromiumcodereview.appspot.com/9416085

TBR=vrk@google.com
Review URL: https://chromiumcodereview.appspot.com/9479012

git-svn-id: svn://svn.chromium.org/chrome/branches/1025/src@123774 0039d316-1c4b-4281-b951-d872f2087c98

2 files changed, 32 insertions, 0 deletions

diff --git a/media/audio/audio_util.cc b/media/audio/audio_util.cc
index b4b87518..8caced5 100644
--- a/media/audio/audio_util.cc
+++ b/media/audio/audio_util.cc
@@ -14,9 +14,11 @@
 #include "base/basictypes.h"
 #include "base/logging.h"
 #include "base/shared_memory.h"
+#include "base/time.h"
 #if defined(OS_WIN)
 #include "base/win/windows_version.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"
@@ -436,4 +438,29 @@ void Crossfade(int bytes_to_crossfade, int number_of_channels,
   }
 }
 
+// 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 int kMinSamplesPerHardwarePacket = 1024;
+
+// The maximum number of samples in a hardware packet.
+// This value is selected so that we can handle up to 192khz sample rate.
+static const int kMaxSamplesPerHardwarePacket = 64 * 1024;
+
+// This constant governs the hardware audio buffer size, this value should be
+// chosen carefully.
+// This value is selected so that we have 8192 samples for 48khz streams.
+static const int kMillisecondsPerHardwarePacket = 170;
+
+uint32 SelectSamplesPerPacket(int sample_rate) {
+  // Select the number of samples that can provide at least
+  // |kMillisecondsPerHardwarePacket| worth of audio data.
+  int samples = kMinSamplesPerHardwarePacket;
+  while (samples <= kMaxSamplesPerHardwarePacket &&
+         samples * base::Time::kMillisecondsPerSecond <
+         sample_rate * kMillisecondsPerHardwarePacket) {
+    samples *= 2;
+  }
+  return samples;
+}
+
 }  // namespace media
diff --git a/media/audio/audio_util.h b/media/audio/audio_util.h
index d8de479..3730c11 100644
--- a/media/audio/audio_util.h
+++ b/media/audio/audio_util.h
@@ -10,6 +10,8 @@
 #include "base/basictypes.h"
 #include "media/base/media_export.h"
 
+struct AudioParameters;
+
 namespace base {
 class SharedMemory;
 }
@@ -122,6 +124,9 @@ MEDIA_EXPORT void Crossfade(int bytes_to_crossfade, int number_of_channels,
                             int bytes_per_channel, const uint8* src,
                             uint8* dest);
 
+// Calculates a safe hardware buffer size (in number of samples) given a set
+// of audio parameters.
+MEDIA_EXPORT uint32 SelectSamplesPerPacket(int sample_rate);
 
 }  // namespace media