Add vector_math::FMUL. Replace audio_util::AdjustVolume.

Removes the integer based volume adjustment code from the
melting pot that is audio_util in favor of an
AudioBus::AdjustVolume() method which works on float.

The driver behind the method is a new vector_math::FMUL
method which is SSE optimized.  Benchmarks put it in line
with the vector_math::FMAC() method.

Benchmarking 200000 iterations:
FMUL_C took 1962.52ms.
FMUL_SSE (unaligned size) took 493.03ms; which is 3.98x faster than FMUL_C.
FMUL_SSE (aligned size) took 491.79ms; which is 3.99x faster than FMUL_C and 1.00x faster than FMUL_SSE (unaligned size).

BUG=120319, 171540, 226447
TEST=new media_unittests.

Review URL: https://chromiumcodereview.appspot.com/13726011

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@192905 0039d316-1c4b-4281-b951-d872f2087c98

12 files changed, 11 insertions, 233 deletions

diff --git a/media/audio/android/opensles_output.cc b/media/audio/android/opensles_output.cc
index f98baaf..36a1408 100644
--- a/media/audio/android/opensles_output.cc
+++ b/media/audio/android/opensles_output.cc
@@ -5,7 +5,6 @@
 #include "media/audio/android/opensles_output.h"
 
 #include "base/logging.h"
-#include "media/audio/audio_util.h"
 #include "media/audio/android/audio_manager_android.h"
 
 namespace media {
@@ -267,16 +266,10 @@ void OpenSLESOutputStream::FillBufferQueue() {
   DCHECK_LE(static_cast<size_t>(num_filled_bytes), buffer_size_bytes_);
   // Note: If this ever changes to output raw float the data must be clipped and
   // sanitized since it may come from an untrusted source such as NaCl.
+  audio_bus_->Scale(volume_);
   audio_bus_->ToInterleaved(
       frames_filled, format_.bitsPerSample / 8, audio_data_[active_queue_]);
 
-  // Perform in-place, software-volume adjustments.
-  media::AdjustVolume(audio_data_[active_queue_],
-                      num_filled_bytes,
-                      format_.numChannels,
-                      format_.bitsPerSample / 8,
-                      volume_);
-
   // Enqueue the buffer for playback.
   SLresult err = (*simple_buffer_queue_)->Enqueue(
       simple_buffer_queue_,
diff --git a/media/audio/audio_util.cc b/media/audio/audio_util.cc
index 351937e..8758fdc 100644
--- a/media/audio/audio_util.cc
+++ b/media/audio/audio_util.cc
@@ -14,86 +14,17 @@
 
 #include "media/audio/audio_util.h"
 
-#include <algorithm>
-#include <limits>
-
-#include "base/basictypes.h"
 #include "base/command_line.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"
 #include "media/base/media_switches.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)
+#if defined(OS_WIN)
 #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/core_audio_util_win.h"
-#include "media/base/limits.h"
 #endif
 
 namespace media {
 
-// 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) {
-  for (int i = 0; i < sample_count; ++i) {
-    buf_out[i] = static_cast<Format>(ScaleChannel<Fixed>(buf_out[i] - bias,
-                                                         fixed_volume) + bias);
-  }
-}
-
-// AdjustVolume() does an in place audio sample change.
-bool AdjustVolume(void* buf,
-                  size_t buflen,
-                  int channels,
-                  int bytes_per_sample,
-                  float volume) {
-  DCHECK(buf);
-  if (volume < 0.0f || volume > 1.0f)
-    return false;
-  if (volume == 1.0f) {
-    return true;
-  } else if (volume == 0.0f) {
-    memset(buf, 0, buflen);
-    return true;
-  }
-  if (channels > 0 && channels <= 8 && bytes_per_sample > 0) {
-    int sample_count = buflen / bytes_per_sample;
-    const int fixed_volume = static_cast<int>(volume * 65536);
-    if (bytes_per_sample == 1) {
-      AdjustVolume<uint8, int32, 128>(reinterpret_cast<uint8*>(buf),
-                                      sample_count,
-                                      fixed_volume);
-      return true;
-    } else if (bytes_per_sample == 2) {
-      AdjustVolume<int16, int32, 0>(reinterpret_cast<int16*>(buf),
-                                    sample_count,
-                                    fixed_volume);
-      return true;
-    } else if (bytes_per_sample == 4) {
-      AdjustVolume<int32, int64, 0>(reinterpret_cast<int32*>(buf),
-                                    sample_count,
-                                    fixed_volume);
-      return true;
-    }
-  }
-  return false;
-}
-
 // Returns user buffer size as specified on the command line or 0 if no buffer
 // size has been specified.
 int GetUserBufferSize() {
diff --git a/media/audio/audio_util.h b/media/audio/audio_util.h
index 9fdbde8..a11c327 100644
--- a/media/audio/audio_util.h
+++ b/media/audio/audio_util.h
@@ -5,39 +5,11 @@
 #ifndef MEDIA_AUDIO_AUDIO_UTIL_H_
 #define MEDIA_AUDIO_AUDIO_UTIL_H_
 
-#include <string>
-
 #include "base/basictypes.h"
-#include "media/base/channel_layout.h"
+#include "build/build_config.h"
 #include "media/base/media_export.h"
 
-namespace base {
-class SharedMemory;
-}
-
 namespace media {
-class AudioBus;
-
-// For all audio functions 3 audio formats are supported:
-// 8 bits unsigned 0 to 255.
-// 16 bit signed (little endian).
-// 32 bit signed (little endian)
-
-// AdjustVolume() does a software volume adjustment of a sample buffer.
-// The samples are multiplied by the volume, which should range from
-// 0.0 (mute) to 1.0 (full volume).
-// Using software allows each audio and video to have its own volume without
-// affecting the master volume.
-// In the future the function may be used to adjust the sample format to
-// simplify hardware requirements and to support a wider variety of input
-// formats.
-// The buffer is modified in-place to avoid memory management, as this
-// function may be called in performance critical code.
-MEDIA_EXPORT bool AdjustVolume(void* buf,
-                               size_t buflen,
-                               int channels,
-                               int bytes_per_sample,
-                               float volume);
 
 // Returns user buffer size as specified on the command line or 0 if no buffer
 // size has been specified.
diff --git a/media/audio/audio_util_unittest.cc b/media/audio/audio_util_unittest.cc
deleted file mode 100644
index 679243c..0000000
--- a/media/audio/audio_util_unittest.cc
+++ /dev/null
@@ -1,82 +0,0 @@
-// 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.
-
-#include "base/basictypes.h"
-#include "media/audio/audio_util.h"
-#include "testing/gtest/include/gtest/gtest.h"
-
-// Number of samples in each audio array.
-static const size_t kNumberOfSamples = 4;
-
-namespace media {
-
-TEST(AudioUtilTest, AdjustVolume_u8) {
-  // Test AdjustVolume() on 8 bit samples.
-  uint8 samples_u8[kNumberOfSamples] = { 4, 0x40, 0x80, 0xff };
-  uint8 expected_u8[kNumberOfSamples] = { (4 - 128) / 2 + 128,
-                                          (0x40 - 128) / 2 + 128,
-                                          (0x80 - 128) / 2 + 128,
-                                          (0xff - 128) / 2 + 128 };
-  bool result_u8 = media::AdjustVolume(samples_u8, sizeof(samples_u8),
-                                       1,  // channels.
-                                       sizeof(samples_u8[0]),
-                                       0.5f);
-  EXPECT_TRUE(result_u8);
-  int expected_test = memcmp(samples_u8, expected_u8, sizeof(expected_u8));
-  EXPECT_EQ(0, expected_test);
-}
-
-TEST(AudioUtilTest, AdjustVolume_s16) {
-  // Test AdjustVolume() on 16 bit samples.
-  int16 samples_s16[kNumberOfSamples] = { -4, 0x40, -32768, 123 };
-  int16 expected_s16[kNumberOfSamples] = { -1, 0x10, -8192, 30 };
-  bool result_s16 = media::AdjustVolume(samples_s16, sizeof(samples_s16),
-                                        2,  // channels.
-                                        sizeof(samples_s16[0]),
-                                        0.25f);
-  EXPECT_TRUE(result_s16);
-  int expected_test = memcmp(samples_s16, expected_s16, sizeof(expected_s16));
-  EXPECT_EQ(0, expected_test);
-}
-
-TEST(AudioUtilTest, AdjustVolume_s16_zero) {
-  // Test AdjustVolume() on 16 bit samples.
-  int16 samples_s16[kNumberOfSamples] = { -4, 0x40, -32768, 123 };
-  int16 expected_s16[kNumberOfSamples] = { 0, 0, 0, 0 };
-  bool result_s16 = media::AdjustVolume(samples_s16, sizeof(samples_s16),
-                                        2,  // channels.
-                                        sizeof(samples_s16[0]),
-                                        0.0f);
-  EXPECT_TRUE(result_s16);
-  int expected_test = memcmp(samples_s16, expected_s16, sizeof(expected_s16));
-  EXPECT_EQ(0, expected_test);
-}
-
-TEST(AudioUtilTest, AdjustVolume_s16_one) {
-  // Test AdjustVolume() on 16 bit samples.
-  int16 samples_s16[kNumberOfSamples] = { -4, 0x40, -32768, 123 };
-  int16 expected_s16[kNumberOfSamples] = { -4, 0x40, -32768, 123 };
-  bool result_s16 = media::AdjustVolume(samples_s16, sizeof(samples_s16),
-                                        2,  // channels.
-                                        sizeof(samples_s16[0]),
-                                        1.0f);
-  EXPECT_TRUE(result_s16);
-  int expected_test = memcmp(samples_s16, expected_s16, sizeof(expected_s16));
-  EXPECT_EQ(0, expected_test);
-}
-
-TEST(AudioUtilTest, AdjustVolume_s32) {
-  // Test AdjustVolume() on 32 bit samples.
-  int32 samples_s32[kNumberOfSamples] = { -4, 0x40, -32768, 123 };
-  int32 expected_s32[kNumberOfSamples] = { -1, 0x10, -8192, 30 };
-  bool result_s32 = media::AdjustVolume(samples_s32, sizeof(samples_s32),
-                                        4,  // channels.
-                                        sizeof(samples_s32[0]),
-                                        0.25f);
-  EXPECT_TRUE(result_s32);
-  int expected_test = memcmp(samples_s32, expected_s32, sizeof(expected_s32));
-  EXPECT_EQ(0, expected_test);
-}
-
-}  // namespace media
diff --git a/media/audio/linux/alsa_output.cc b/media/audio/linux/alsa_output.cc
index ea3d1b6..4615d01 100644
--- a/media/audio/linux/alsa_output.cc
+++ b/media/audio/linux/alsa_output.cc
@@ -42,7 +42,6 @@
 #include "base/message_loop.h"
 #include "base/stl_util.h"
 #include "base/time.h"
-#include "media/audio/audio_util.h"
 #include "media/audio/linux/alsa_util.h"
 #include "media/audio/linux/alsa_wrapper.h"
 #include "media/audio/linux/audio_manager_linux.h"
@@ -379,15 +378,10 @@ void AlsaPcmOutputStream::BufferPacket(bool* source_exhausted) {
 
     // Note: If this ever changes to output raw float the data must be clipped
     // and sanitized since it may come from an untrusted source such as NaCl.
+    output_bus->Scale(volume_);
     output_bus->ToInterleaved(
         frames_filled, bytes_per_sample_, packet->GetWritableData());
 
-    media::AdjustVolume(packet->GetWritableData(),
-                        packet_size,
-                        output_bus->channels(),
-                        bytes_per_sample_,
-                        volume_);
-
     if (packet_size > 0) {
       packet->SetDataSize(packet_size);
       // Add the packet to the buffer.
diff --git a/media/audio/mac/audio_auhal_mac.cc b/media/audio/mac/audio_auhal_mac.cc
index 20df6f5..b2a8828 100644
--- a/media/audio/mac/audio_auhal_mac.cc
+++ b/media/audio/mac/audio_auhal_mac.cc
@@ -178,8 +178,6 @@ void AUHALStream::Stop() {
 
 void AUHALStream::SetVolume(double volume) {
   volume_ = static_cast<float>(volume);
-
-  // TODO(crogers): set volume property
 }
 
 void AUHALStream::GetVolume(double* volume) {
@@ -251,6 +249,7 @@ OSStatus AUHALStream::Render(
         input_bus_.get(),
         output_bus_.get(),
         AudioBuffersState(0, hardware_pending_bytes));
+    output_bus_->Scale(volume_);
   }
 
   return noErr;
diff --git a/media/audio/mac/audio_low_latency_output_mac.cc b/media/audio/mac/audio_low_latency_output_mac.cc
index 4b82bef..592a6a1 100644
--- a/media/audio/mac/audio_low_latency_output_mac.cc
+++ b/media/audio/mac/audio_low_latency_output_mac.cc
@@ -10,7 +10,6 @@
 #include "base/command_line.h"
 #include "base/logging.h"
 #include "base/mac/mac_logging.h"
-#include "media/audio/audio_util.h"
 #include "media/audio/mac/audio_manager_mac.h"
 #include "media/base/media_switches.h"
 
@@ -284,16 +283,9 @@ OSStatus AUAudioOutputStream::Render(UInt32 number_of_frames,
 
   // Note: If this ever changes to output raw float the data must be clipped and
   // sanitized since it may come from an untrusted source such as NaCl.
+  audio_bus_->Scale(volume_);
   audio_bus_->ToInterleaved(
       frames_filled, format_.mBitsPerChannel / 8, audio_data);
-  uint32 filled = frames_filled * format_.mBytesPerFrame;
-
-  // Perform in-place, software-volume adjustments.
-  media::AdjustVolume(audio_data,
-                      filled,
-                      audio_bus_->channels(),
-                      format_.mBitsPerChannel / 8,
-                      volume_);
 
   return noErr;
 }
diff --git a/media/audio/pulse/pulse_output.cc b/media/audio/pulse/pulse_output.cc
index eccf463..1734954 100644
--- a/media/audio/pulse/pulse_output.cc
+++ b/media/audio/pulse/pulse_output.cc
@@ -9,7 +9,6 @@
 #include "base/message_loop.h"
 #include "media/audio/audio_manager_base.h"
 #include "media/audio/audio_parameters.h"
-#include "media/audio/audio_util.h"
 #include "media/audio/pulse/pulse_util.h"
 
 namespace media {
@@ -143,10 +142,9 @@ void PulseAudioOutputStream::FulfillWriteRequest(size_t requested_bytes) {
 
     // Note: If this ever changes to output raw float the data must be clipped
     // and sanitized since it may come from an untrusted source such as NaCl.
+    audio_bus_->Scale(volume_);
     audio_bus_->ToInterleaved(
         audio_bus_->frames(), params_.bits_per_sample() / 8, buffer);
-    media::AdjustVolume(buffer, bytes_to_fill, params_.channels(),
-                        params_.bits_per_sample() / 8, volume_);
 
     if (pa_stream_write(pa_stream_, buffer, bytes_to_fill, NULL, 0LL,
                         PA_SEEK_RELATIVE) < 0) {
diff --git a/media/audio/pulse/pulse_unified.cc b/media/audio/pulse/pulse_unified.cc
index 791a3a8..87bb6ae 100644
--- a/media/audio/pulse/pulse_unified.cc
+++ b/media/audio/pulse/pulse_unified.cc
@@ -8,7 +8,6 @@
 #include "base/time.h"
 #include "media/audio/audio_manager_base.h"
 #include "media/audio/audio_parameters.h"
-#include "media/audio/audio_util.h"
 #include "media/audio/pulse/pulse_util.h"
 #include "media/base/seekable_buffer.h"
 
@@ -175,10 +174,9 @@ void PulseAudioUnifiedStream::WriteData(size_t requested_bytes) {
 
   // Note: If this ever changes to output raw float the data must be clipped
   // and sanitized since it may come from an untrusted source such as NaCl.
+  output_bus_->Scale(volume_);
   output_bus_->ToInterleaved(
       output_bus_->frames(), params_.bits_per_sample() / 8, buffer);
-  media::AdjustVolume(buffer, requested_bytes, params_.channels(),
-                      params_.bits_per_sample() / 8, volume_);
 
   if (pa_stream_write(output_stream_, buffer, requested_bytes, NULL, 0LL,
                       PA_SEEK_RELATIVE) < 0) {
diff --git a/media/audio/win/audio_low_latency_output_win.cc b/media/audio/win/audio_low_latency_output_win.cc
index b5a4c47..04b5dd2 100644
--- a/media/audio/win/audio_low_latency_output_win.cc
+++ b/media/audio/win/audio_low_latency_output_win.cc
@@ -13,7 +13,6 @@
 #include "base/metrics/histogram.h"
 #include "base/utf_string_conversions.h"
 #include "base/win/scoped_propvariant.h"
-#include "media/audio/audio_util.h"
 #include "media/audio/win/audio_manager_win.h"
 #include "media/audio/win/avrt_wrapper_win.h"
 #include "media/audio/win/core_audio_util_win.h"
@@ -581,15 +580,10 @@ void WASAPIAudioOutputStream::RenderAudioFromSource(
     // clipped and sanitized since it may come from an untrusted
     // source such as NaCl.
     const int bytes_per_sample = format_.Format.wBitsPerSample >> 3;
+    audio_bus_->Scale(volume_);
     audio_bus_->ToInterleaved(
         frames_filled, bytes_per_sample, audio_data);
 
-    // Perform in-place, software-volume adjustments.
-    media::AdjustVolume(audio_data,
-                        num_filled_bytes,
-                        audio_bus_->channels(),
-                        bytes_per_sample,
-                        volume_);
 
     // Release the buffer space acquired in the GetBuffer() call.
     // Render silence if we were not able to fill up the buffer totally.
diff --git a/media/audio/win/audio_unified_win.cc b/media/audio/win/audio_unified_win.cc
index e7877f5..59b98d0 100644
--- a/media/audio/win/audio_unified_win.cc
+++ b/media/audio/win/audio_unified_win.cc
@@ -9,7 +9,6 @@
 #include "base/debug/trace_event.h"
 #include "base/time.h"
 #include "base/win/scoped_com_initializer.h"
-#include "media/audio/audio_util.h"
 #include "media/audio/win/audio_manager_win.h"
 #include "media/audio/win/avrt_wrapper_win.h"
 #include "media/audio/win/core_audio_util_win.h"
@@ -529,16 +528,10 @@ void WASAPIUnifiedStream::Run() {
 
           // Convert the audio bus content to interleaved integer data using
           // |audio_data| as destination.
+          render_bus_->Scale(volume_);
           render_bus_->ToInterleaved(
               packet_size_frames_, bytes_per_sample, audio_data);
 
-          // Perform in-place, software-volume adjustments.
-          media::AdjustVolume(audio_data,
-                              frames_filled * format_.Format.nBlockAlign,
-                              render_bus_->channels(),
-                              bytes_per_sample,
-                              volume_);
-
           // Release the buffer space acquired in the GetBuffer() call.
           audio_render_client_->ReleaseBuffer(packet_size_frames_, 0);
           DLOG_IF(ERROR, FAILED(hr)) << "Failed to release render buffer";
diff --git a/media/audio/win/waveout_output_win.cc b/media/audio/win/waveout_output_win.cc
index f42d948..9b03c31 100644
--- a/media/audio/win/waveout_output_win.cc
+++ b/media/audio/win/waveout_output_win.cc
@@ -13,7 +13,6 @@
 #include "base/debug/trace_event.h"
 #include "base/logging.h"
 #include "media/audio/audio_io.h"
-#include "media/audio/audio_util.h"
 #include "media/audio/win/audio_manager_win.h"
 
 namespace media {
@@ -360,14 +359,11 @@ void PCMWaveOutAudioOutputStream::QueueNextPacket(WAVEHDR *buffer) {
   if (used <= buffer_size_) {
     // Note: If this ever changes to output raw float the data must be clipped
     // and sanitized since it may come from an untrusted source such as NaCl.
+    audio_bus_->Scale(volume_);
     audio_bus_->ToInterleaved(
         frames_filled, format_.Format.wBitsPerSample / 8, buffer->lpData);
 
     buffer->dwBufferLength = used * format_.Format.nChannels / channels_;
-    media::AdjustVolume(buffer->lpData, used,
-                        format_.Format.nChannels,
-                        format_.Format.wBitsPerSample >> 3,
-                        volume_);
   } else {
     HandleError(0);
     return;