Revert 46314 - floating point audio support in util functions for volume and folding

BUG=42861
TEST=none

Review URL: http://codereview.chromium.org/1856002

TBR=fbarchard@chromium.org
Review URL: http://codereview.chromium.org/1926001

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@46320 0039d316-1c4b-4281-b951-d872f2087c98

3 files changed, 29 insertions, 65 deletions

diff --git a/media/audio/audio_util.cc b/media/audio/audio_util.cc
index d72cbe7..8774a72 100644
--- a/media/audio/audio_util.cc
+++ b/media/audio/audio_util.cc
@@ -24,20 +24,15 @@ static int ScaleChannel(int channel, int volume) {
 }
 
 template<class Format, class Fixed, int bias>
-static void AdjustVolume(Format* buf_out, int sample_count, int fixed_volume) {
+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);
   }
 }
 
-static void AdjustVolumeFloat(float* buf_out, int sample_count, float volume) {
-  for (int i = 0; i < sample_count; ++i) {
-    buf_out[i] = buf_out[i] * volume;
-  }
-}
-
-
 // Channel order for AAC
 // From http://www.hydrogenaudio.org/forums/lofiversion/index.php/t40046.html
 
@@ -46,7 +41,8 @@ static const int kChannel_L = 1;
 static const int kChannel_R = 2;
 
 template<class Fixed, int min_value, int max_value>
-static int AddChannel(int val, int adder) {
+static int AddChannel(int val,
+               int adder) {
   Fixed sum = static_cast<Fixed>(val) + static_cast<Fixed>(adder);
   if (sum > max_value)
     return max_value;
@@ -66,9 +62,9 @@ static int AddChannel(int val, int adder) {
 
 template<class Format, class Fixed, int min_value, int max_value, int bias>
 static void FoldChannels(Format* buf_out,
-                         int sample_count,
-                         const float volume,
-                         int channels) {
+                  int sample_count,
+                  const float volume,
+                  int channels) {
   Format* buf_in = buf_out;
   const int center_volume = static_cast<int>(volume * 0.707f * 65536);
   const int fixed_volume = static_cast<int>(volume * 65536);
@@ -91,31 +87,6 @@ static void FoldChannels(Format* buf_out,
     buf_in += channels;
   }
 }
-
-static void FoldChannelsFloat(float* buf_out,
-                              int sample_count,
-                              const float volume,
-                              int channels) {
-  float* buf_in = buf_out;
-  const float center_volume = volume * 0.707f;
-
-  for (int i = 0; i < sample_count; ++i) {
-    float center = buf_in[kChannel_C];
-    float left = buf_in[kChannel_L];
-    float right = buf_in[kChannel_R];
-
-    center = center * center_volume;
-    left = left * volume;
-    right = right * volume;
-
-    buf_out[0] = left + center;
-    buf_out[1] = right + center;
-
-    buf_out += 2;
-    buf_in += channels;
-  }
-}
-
 }  // namespace
 
 // AdjustVolume() does an in place audio sample change.
@@ -146,10 +117,9 @@ bool AdjustVolume(void* buf,
                                     fixed_volume);
       return true;
     } else if (bytes_per_sample == 4) {
-      // 4 byte per sample is float.
-      AdjustVolumeFloat(reinterpret_cast<float*>(buf),
-                        sample_count,
-                        volume);
+      AdjustVolume<int32, int64, 0>(reinterpret_cast<int32*>(buf),
+                                    sample_count,
+                                    fixed_volume);
       return true;
     }
   }
@@ -180,9 +150,8 @@ bool FoldChannels(void* buf,
           channels);
       return true;
     } else if (bytes_per_sample == 4) {
-      // 4 byte per sample is float.
-      FoldChannelsFloat(
-          reinterpret_cast<float*>(buf),
+      FoldChannels<int32, int64, 0x80000000, 0x7fffffff, 0>(
+          reinterpret_cast<int32*>(buf),
           sample_count,
           volume,
           channels);
diff --git a/media/audio/audio_util_unittest.cc b/media/audio/audio_util_unittest.cc
index 22e9afe..c0ed0fc 100644
--- a/media/audio/audio_util_unittest.cc
+++ b/media/audio/audio_util_unittest.cc
@@ -66,19 +66,16 @@ TEST(AudioUtilTest, AdjustVolume_s16_one) {
   EXPECT_EQ(0, expected_test);
 }
 
-TEST(AudioUtilTest, AdjustVolume_f32) {
+TEST(AudioUtilTest, AdjustVolume_s32) {
   // Test AdjustVolume() on 32 bit samples.
-  float samples_f32[kNumberOfSamples] = { -4.0f, 0.5f, -.05f, 123.0f };
-  float expected_f32[kNumberOfSamples] = { -4.0f * 0.25f,
-                                           0.5f * 0.25f,
-                                           -.05f * 0.25f,
-                                           123.0f * 0.25f };
-  bool result_f32 = media::AdjustVolume(samples_f32, sizeof(samples_f32),
+  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_f32[0]),
+                                        sizeof(samples_s32[0]),
                                         0.25f);
-  EXPECT_EQ(true, result_f32);
-  int expected_test = memcmp(samples_f32, expected_f32, sizeof(expected_f32));
+  EXPECT_EQ(true, result_s32);
+  int expected_test = memcmp(samples_s32, expected_s32, sizeof(expected_s32));
   EXPECT_EQ(0, expected_test);
 }
 
@@ -112,17 +109,17 @@ TEST(AudioUtilTest, FoldChannels_s16) {
   EXPECT_EQ(0, expected_test);
 }
 
-TEST(AudioUtilTest, FoldChannels_f32) {
+TEST(AudioUtilTest, FoldChannels_s32) {
   // Test FoldChannels() on 32 bit samples.
-  float samples_f32[6] = { 12.0f, 1.0f, 3.0f, 7.0f, 13.0f, 17.0f };
-  float expected_f32[2] = { 12.0f * .707f + 1.0f,
-                            12.0f * .707f + 3.0f };
-  bool result_f32 = media::FoldChannels(samples_f32, sizeof(samples_f32),
+  int32 samples_s32[6] = { 12, 1, 3, 7, 13, 17 };
+  int32 expected_s32[2] = { static_cast<int16>(12 * .707 + 1),
+                            static_cast<int16>(12 * .707 + 3) };
+  bool result_s32 = media::FoldChannels(samples_s32, sizeof(samples_s32),
                                         6,  // channels.
-                                        sizeof(samples_f32[0]),
+                                        sizeof(samples_s32[0]),
                                         1.00f);
-  EXPECT_EQ(true, result_f32);
-  int expected_test = memcmp(samples_f32, expected_f32, sizeof(expected_f32));
+  EXPECT_EQ(true, result_s32);
+  int expected_test = memcmp(samples_s32, expected_s32, sizeof(expected_s32));
   EXPECT_EQ(0, expected_test);
 }
 
diff --git a/media/audio/win/waveout_output_win.cc b/media/audio/win/waveout_output_win.cc
index 073d0fc..59f4c1c 100644
--- a/media/audio/win/waveout_output_win.cc
+++ b/media/audio/win/waveout_output_win.cc
@@ -5,7 +5,6 @@
 #include "media/audio/win/waveout_output_win.h"
 
 #include <windows.h>
-#include <mmreg.h>
 #include <mmsystem.h>
 #pragma comment(lib, "winmm.lib")
 
@@ -56,8 +55,7 @@ PCMWaveOutAudioOutputStream::PCMWaveOutAudioOutputStream(
       volume_(1),
       channels_(channels),
       pending_bytes_(0) {
-  format_.wFormatTag = bits_per_sample == 32 ?
-      WAVE_FORMAT_IEEE_FLOAT : WAVE_FORMAT_PCM;
+  format_.wFormatTag = WAVE_FORMAT_PCM;
   format_.nChannels = channels > 2 ? 2 : channels;
   format_.nSamplesPerSec = sampling_rate;
   format_.wBitsPerSample = bits_per_sample;