Cleanup Make(Planar|Interleaved)AudioBuffer functions.

Removes an unnecessary copy now that we have support for writing
into existing AudioBuffer objects.

BUG=none
TEST=unittests still pass.
NOTRY=true

Review URL: https://codereview.chromium.org/183513007

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@254798 0039d316-1c4b-4281-b951-d872f2087c98

7 files changed, 142 insertions, 192 deletions

diff --git a/media/base/audio_buffer_queue_unittest.cc b/media/base/audio_buffer_queue_unittest.cc
index b95bdca..b765009 100644
--- a/media/base/audio_buffer_queue_unittest.cc
+++ b/media/base/audio_buffer_queue_unittest.cc
@@ -34,12 +34,12 @@ TEST(AudioBufferQueueTest, AppendAndClear) {
   const base::TimeDelta kNoTime = kNoTimestamp();
   AudioBufferQueue buffer;
   EXPECT_EQ(0, buffer.frames());
-  buffer.Append(MakeInterleavedAudioBuffer<uint8>(
+  buffer.Append(MakeAudioBuffer<uint8>(
       kSampleFormatU8, channels, 10, 1, frames, kNoTime, kNoTime));
   EXPECT_EQ(frames, buffer.frames());
   buffer.Clear();
   EXPECT_EQ(0, buffer.frames());
-  buffer.Append(MakeInterleavedAudioBuffer<uint8>(
+  buffer.Append(MakeAudioBuffer<uint8>(
       kSampleFormatU8, channels, 20, 1, frames, kNoTime, kNoTime));
   EXPECT_EQ(frames, buffer.frames());
 }
@@ -51,19 +51,19 @@ TEST(AudioBufferQueueTest, MultipleAppend) {
   AudioBufferQueue buffer;
 
   // Append 40 frames in 5 buffers.
-  buffer.Append(MakeInterleavedAudioBuffer<uint8>(
+  buffer.Append(MakeAudioBuffer<uint8>(
       kSampleFormatU8, channels, 10, 1, frames, kNoTime, kNoTime));
   EXPECT_EQ(8, buffer.frames());
-  buffer.Append(MakeInterleavedAudioBuffer<uint8>(
+  buffer.Append(MakeAudioBuffer<uint8>(
       kSampleFormatU8, channels, 10, 1, frames, kNoTime, kNoTime));
   EXPECT_EQ(16, buffer.frames());
-  buffer.Append(MakeInterleavedAudioBuffer<uint8>(
+  buffer.Append(MakeAudioBuffer<uint8>(
       kSampleFormatU8, channels, 10, 1, frames, kNoTime, kNoTime));
   EXPECT_EQ(24, buffer.frames());
-  buffer.Append(MakeInterleavedAudioBuffer<uint8>(
+  buffer.Append(MakeAudioBuffer<uint8>(
       kSampleFormatU8, channels, 10, 1, frames, kNoTime, kNoTime));
   EXPECT_EQ(32, buffer.frames());
-  buffer.Append(MakeInterleavedAudioBuffer<uint8>(
+  buffer.Append(MakeAudioBuffer<uint8>(
       kSampleFormatU8, channels, 10, 1, frames, kNoTime, kNoTime));
   EXPECT_EQ(40, buffer.frames());
 }
@@ -77,7 +77,7 @@ TEST(AudioBufferQueueTest, IteratorCheck) {
 
   // Append 40 frames in 5 buffers. Intersperse ReadFrames() to make the
   // iterator is pointing to the correct position.
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 10.0f, 1.0f, frames, kNoTime, kNoTime));
   EXPECT_EQ(8, buffer.frames());
 
@@ -85,10 +85,10 @@ TEST(AudioBufferQueueTest, IteratorCheck) {
   EXPECT_EQ(4, buffer.frames());
   VerifyResult(bus->channel(0), 4, 10.0f, 1.0f);
 
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 20.0f, 1.0f, frames, kNoTime, kNoTime));
   EXPECT_EQ(12, buffer.frames());
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 30.0f, 1.0f, frames, kNoTime, kNoTime));
   EXPECT_EQ(20, buffer.frames());
 
@@ -97,10 +97,10 @@ TEST(AudioBufferQueueTest, IteratorCheck) {
   EXPECT_EQ(0, buffer.frames());
   VerifyResult(bus->channel(0), 4, 34.0f, 1.0f);
 
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 40.0f, 1.0f, frames, kNoTime, kNoTime));
   EXPECT_EQ(8, buffer.frames());
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 50.0f, 1.0f, frames, kNoTime, kNoTime));
   EXPECT_EQ(16, buffer.frames());
 
@@ -121,7 +121,7 @@ TEST(AudioBufferQueueTest, Seek) {
   AudioBufferQueue buffer;
 
   // Add 6 frames of data.
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 1.0f, 1.0f, frames, kNoTime, kNoTime));
   EXPECT_EQ(6, buffer.frames());
 
@@ -143,11 +143,11 @@ TEST(AudioBufferQueueTest, ReadF32) {
   AudioBufferQueue buffer;
 
   // Add 76 frames of data.
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 1.0f, 1.0f, 6, kNoTime, kNoTime));
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 13.0f, 1.0f, 10, kNoTime, kNoTime));
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 33.0f, 1.0f, 60, kNoTime, kNoTime));
   EXPECT_EQ(76, buffer.frames());
 
@@ -182,7 +182,7 @@ TEST(AudioBufferQueueTest, ReadU8) {
   AudioBufferQueue buffer;
 
   // Add 4 frames of data.
-  buffer.Append(MakeInterleavedAudioBuffer<uint8>(
+  buffer.Append(MakeAudioBuffer<uint8>(
       kSampleFormatU8, channels, 128, 1, frames, kNoTime, kNoTime));
 
   // Read all 4 frames from the buffer. Data is interleaved, so ch[0] should be
@@ -204,9 +204,9 @@ TEST(AudioBufferQueueTest, ReadS16) {
   AudioBufferQueue buffer;
 
   // Add 24 frames of data.
-  buffer.Append(MakeInterleavedAudioBuffer<int16>(
+  buffer.Append(MakeAudioBuffer<int16>(
       kSampleFormatS16, channels, 1, 1, 4, kNoTime, kNoTime));
-  buffer.Append(MakeInterleavedAudioBuffer<int16>(
+  buffer.Append(MakeAudioBuffer<int16>(
       kSampleFormatS16, channels, 9, 1, 20, kNoTime, kNoTime));
   EXPECT_EQ(24, buffer.frames());
 
@@ -226,9 +226,9 @@ TEST(AudioBufferQueueTest, ReadS32) {
   AudioBufferQueue buffer;
 
   // Add 24 frames of data.
-  buffer.Append(MakeInterleavedAudioBuffer<int32>(
+  buffer.Append(MakeAudioBuffer<int32>(
       kSampleFormatS32, channels, 1, 1, 4, kNoTime, kNoTime));
-  buffer.Append(MakeInterleavedAudioBuffer<int32>(
+  buffer.Append(MakeAudioBuffer<int32>(
       kSampleFormatS32, channels, 9, 1, 20, kNoTime, kNoTime));
   EXPECT_EQ(24, buffer.frames());
 
@@ -254,9 +254,9 @@ TEST(AudioBufferQueueTest, ReadF32Planar) {
   AudioBufferQueue buffer;
 
   // Add 14 frames of data.
-  buffer.Append(MakePlanarAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatPlanarF32, channels, 1.0f, 1.0f, 4, kNoTime, kNoTime));
-  buffer.Append(MakePlanarAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatPlanarF32, channels, 50.0f, 1.0f, 10, kNoTime, kNoTime));
   EXPECT_EQ(14, buffer.frames());
 
@@ -277,9 +277,9 @@ TEST(AudioBufferQueueTest, ReadS16Planar) {
   AudioBufferQueue buffer;
 
   // Add 24 frames of data.
-  buffer.Append(MakePlanarAudioBuffer<int16>(
+  buffer.Append(MakeAudioBuffer<int16>(
       kSampleFormatPlanarS16, channels, 1, 1, 4, kNoTime, kNoTime));
-  buffer.Append(MakePlanarAudioBuffer<int16>(
+  buffer.Append(MakeAudioBuffer<int16>(
       kSampleFormatPlanarS16, channels, 100, 5, 20, kNoTime, kNoTime));
   EXPECT_EQ(24, buffer.frames());
 
@@ -301,17 +301,17 @@ TEST(AudioBufferQueueTest, ReadManyChannels) {
   AudioBufferQueue buffer;
 
   // Add 76 frames of data.
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 0.0f, 1.0f, 6, kNoTime, kNoTime));
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 6.0f * channels, 1.0f, 10, kNoTime, kNoTime));
-  buffer.Append(MakeInterleavedAudioBuffer<float>(kSampleFormatF32,
-                                                  channels,
-                                                  16.0f * channels,
-                                                  1.0f,
-                                                  60,
-                                                  kNoTime,
-                                                  kNoTime));
+  buffer.Append(MakeAudioBuffer<float>(kSampleFormatF32,
+                                       channels,
+                                       16.0f * channels,
+                                       1.0f,
+                                       60,
+                                       kNoTime,
+                                       kNoTime));
   EXPECT_EQ(76, buffer.frames());
 
   // Read 3 frames from the buffer. F32 is interleaved, so ch[0] should be
@@ -330,7 +330,7 @@ TEST(AudioBufferQueueTest, Peek) {
   AudioBufferQueue buffer;
 
   // Add 60 frames of data.
-  buffer.Append(MakeInterleavedAudioBuffer<float>(
+  buffer.Append(MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 0.0f, 1.0f, 60, kNoTime, kNoTime));
   EXPECT_EQ(60, buffer.frames());
 
@@ -381,7 +381,7 @@ TEST(AudioBufferQueueTest, Time) {
   // Add two buffers (second one added later):
   //   first:  start=0s,  duration=10s
   //   second: start=30s, duration=10s
-  buffer.Append(MakeInterleavedAudioBuffer<int16>(
+  buffer.Append(MakeAudioBuffer<int16>(
       kSampleFormatS16, channels, 1, 1, 10, start_time1, duration));
   EXPECT_EQ(10, buffer.frames());
 
@@ -399,7 +399,7 @@ TEST(AudioBufferQueueTest, Time) {
             buffer.current_time());
 
   // Add second buffer for more data.
-  buffer.Append(MakeInterleavedAudioBuffer<int16>(
+  buffer.Append(MakeAudioBuffer<int16>(
       kSampleFormatS16, channels, 1, 1, 10, start_time2, duration));
   EXPECT_EQ(16, buffer.frames());
 
@@ -430,9 +430,9 @@ TEST(AudioBufferQueueTest, NoTime) {
   scoped_ptr<AudioBus> bus = AudioBus::Create(channels, 100);
 
   // Add two buffers with no timestamps. Time should always be unknown.
-  buffer.Append(MakeInterleavedAudioBuffer<int16>(
+  buffer.Append(MakeAudioBuffer<int16>(
       kSampleFormatS16, channels, 1, 1, 10, kNoTime, kNoTime));
-  buffer.Append(MakeInterleavedAudioBuffer<int16>(
+  buffer.Append(MakeAudioBuffer<int16>(
       kSampleFormatS16, channels, 1, 1, 10, kNoTime, kNoTime));
   EXPECT_EQ(20, buffer.frames());
 
diff --git a/media/base/audio_buffer_unittest.cc b/media/base/audio_buffer_unittest.cc
index 473778a..15f6416 100644
--- a/media/base/audio_buffer_unittest.cc
+++ b/media/base/audio_buffer_unittest.cc
@@ -28,7 +28,7 @@ TEST(AudioBufferTest, CopyFrom) {
   const int frames = 8;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
-  scoped_refptr<AudioBuffer> buffer = MakeInterleavedAudioBuffer<uint8>(
+  scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<uint8>(
       kSampleFormatU8, channels, 1, 1, frames, start_time, duration);
   EXPECT_EQ(frames, buffer->frame_count());
   EXPECT_EQ(buffer->timestamp(), start_time);
@@ -63,7 +63,7 @@ TEST(AudioBufferTest, ReadU8) {
   const int frames = 4;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
-  scoped_refptr<AudioBuffer> buffer = MakeInterleavedAudioBuffer<uint8>(
+  scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<uint8>(
       kSampleFormatU8, channels, 128, 1, frames, start_time, duration);
 
   // Read all 4 frames from the buffer. Data is interleaved, so ch[0] should be
@@ -83,7 +83,7 @@ TEST(AudioBufferTest, ReadS16) {
   const int frames = 10;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
-  scoped_refptr<AudioBuffer> buffer = MakeInterleavedAudioBuffer<int16>(
+  scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<int16>(
       kSampleFormatS16, channels, 1, 1, frames, start_time, duration);
 
   // Read 6 frames from the buffer. Data is interleaved, so ch[0] should be 1,
@@ -108,7 +108,7 @@ TEST(AudioBufferTest, ReadS32) {
   const int frames = 6;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
-  scoped_refptr<AudioBuffer> buffer = MakeInterleavedAudioBuffer<int32>(
+  scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<int32>(
       kSampleFormatS32, channels, 1, 1, frames, start_time, duration);
 
   // Read 6 frames from the buffer. Data is interleaved, so ch[0] should be 1,
@@ -131,7 +131,7 @@ TEST(AudioBufferTest, ReadF32) {
   const int frames = 20;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
-  scoped_refptr<AudioBuffer> buffer = MakeInterleavedAudioBuffer<float>(
+  scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<float>(
       kSampleFormatF32, channels, 1.0f, 1.0f, frames, start_time, duration);
 
   // Read first 10 frames from the buffer. F32 is interleaved, so ch[0] should
@@ -153,7 +153,7 @@ TEST(AudioBufferTest, ReadS16Planar) {
   const int frames = 20;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
-  scoped_refptr<AudioBuffer> buffer = MakePlanarAudioBuffer<int16>(
+  scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<int16>(
       kSampleFormatPlanarS16, channels, 1, 1, frames, start_time, duration);
 
   // Read 6 frames from the buffer. Data is planar, so ch[0] should be 1, 2, 3,
@@ -187,13 +187,13 @@ TEST(AudioBufferTest, ReadF32Planar) {
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
   scoped_refptr<AudioBuffer> buffer =
-      MakePlanarAudioBuffer<float>(kSampleFormatPlanarF32,
-                                   channels,
-                                   1.0f,
-                                   1.0f,
-                                   frames,
-                                   start_time,
-                                   duration);
+      MakeAudioBuffer<float>(kSampleFormatPlanarF32,
+                             channels,
+                             1.0f,
+                             1.0f,
+                             frames,
+                             start_time,
+                             duration);
 
   // Read all 100 frames from the buffer. F32 is planar, so ch[0] should be 1,
   // 2, 3, 4, ..., ch[1] should be 101, 102, 103, ..., and so on for all 4
@@ -241,13 +241,13 @@ TEST(AudioBufferTest, Trim) {
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
   scoped_refptr<AudioBuffer> buffer =
-      MakePlanarAudioBuffer<float>(kSampleFormatPlanarF32,
-                                   channels,
-                                   1.0f,
-                                   1.0f,
-                                   frames,
-                                   start_time,
-                                   duration);
+      MakeAudioBuffer<float>(kSampleFormatPlanarF32,
+                             channels,
+                             1.0f,
+                             1.0f,
+                             frames,
+                             start_time,
+                             duration);
   EXPECT_EQ(frames, buffer->frame_count());
   EXPECT_EQ(start_time, buffer->timestamp());
   EXPECT_EQ(frames, buffer->duration().InSeconds());
diff --git a/media/base/audio_splicer_unittest.cc b/media/base/audio_splicer_unittest.cc
index 998a9a3..0acd37e 100644
--- a/media/base/audio_splicer_unittest.cc
+++ b/media/base/audio_splicer_unittest.cc
@@ -34,7 +34,7 @@ class AudioSplicerTest : public ::testing::Test {
   }
 
   scoped_refptr<AudioBuffer> GetNextInputBuffer(float value, int frame_size) {
-    scoped_refptr<AudioBuffer> buffer = MakeInterleavedAudioBuffer<float>(
+    scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<float>(
         kSampleFormat,
         kChannels,
         value,
diff --git a/media/base/test_helpers.cc b/media/base/test_helpers.cc
index 672f8c2..57ac40d 100644
--- a/media/base/test_helpers.cc
+++ b/media/base/test_helpers.cc
@@ -149,100 +149,62 @@ gfx::Size TestVideoConfig::LargeCodedSize() {
 }
 
 template <class T>
-scoped_refptr<AudioBuffer> MakeInterleavedAudioBuffer(
-    SampleFormat format,
-    int channels,
-    T start,
-    T increment,
-    int frames,
-    base::TimeDelta start_time,
-    base::TimeDelta duration) {
-  DCHECK(format == kSampleFormatU8 || format == kSampleFormatS16 ||
-         format == kSampleFormatS32 || format == kSampleFormatF32);
+scoped_refptr<AudioBuffer> MakeAudioBuffer(SampleFormat format,
+                                           int channels,
+                                           T start,
+                                           T increment,
+                                           int frames,
+                                           base::TimeDelta timestamp,
+                                           base::TimeDelta duration) {
+  scoped_refptr<AudioBuffer> output =
+      AudioBuffer::CreateBuffer(format, channels, frames);
+  output->set_timestamp(timestamp);
+  output->set_duration(duration);
 
   // Create a block of memory with values:
   //   start
   //   start + increment
   //   start + 2 * increment, ...
-  // Since this is interleaved data, channel 0 data will be:
+  // For interleaved data, raw data will be:
   //   start
   //   start + channels * increment
   //   start + 2 * channels * increment, ...
-  int buffer_size = frames * channels * sizeof(T);
-  scoped_ptr<uint8[]> memory(new uint8[buffer_size]);
-  uint8* data[] = { memory.get() };
-  T* buffer = reinterpret_cast<T*>(memory.get());
-  for (int i = 0; i < frames * channels; ++i) {
-    buffer[i] = start;
-    start += increment;
-  }
-  return AudioBuffer::CopyFrom(
-      format, channels, frames, data, start_time, duration);
-}
-
-template <class T>
-scoped_refptr<AudioBuffer> MakePlanarAudioBuffer(
-    SampleFormat format,
-    int channels,
-    T start,
-    T increment,
-    int frames,
-    base::TimeDelta start_time,
-    base::TimeDelta duration) {
-  DCHECK(format == kSampleFormatPlanarF32 || format == kSampleFormatPlanarS16);
-
-  // Create multiple blocks of data, one for each channel.
-  // Values in channel 0 will be:
+  //
+  // For planar data, values in channel 0 will be:
   //   start
   //   start + increment
   //   start + 2 * increment, ...
-  // Values in channel 1 will be:
+  // While, values in channel 1 will be:
   //   start + frames * increment
   //   start + (frames + 1) * increment
   //   start + (frames + 2) * increment, ...
-  int buffer_size = frames * sizeof(T);
-  scoped_ptr<uint8*[]> data(new uint8*[channels]);
-  scoped_ptr<uint8[]> memory(new uint8[channels * buffer_size]);
-  for (int i = 0; i < channels; ++i) {
-    data.get()[i] = memory.get() + i * buffer_size;
-    T* buffer = reinterpret_cast<T*>(data.get()[i]);
-    for (int j = 0; j < frames; ++j) {
-      buffer[j] = start;
+  const size_t output_size =
+      output->channel_data().size() == 1 ? frames * channels : frames;
+  for (size_t ch = 0; ch < output->channel_data().size(); ++ch) {
+    T* buffer = reinterpret_cast<T*>(output->channel_data()[ch]);
+    for (size_t i = 0; i < output_size; ++i) {
+      buffer[i] = start;
       start += increment;
     }
   }
-  return AudioBuffer::CopyFrom(
-      format, channels, frames, data.get(), start_time, duration);
-}
-
-// Instantiate all the types of MakeInterleavedAudioBuffer() and
-// MakePlanarAudioBuffer() needed.
-
-#define DEFINE_INTERLEAVED_INSTANCE(type)                               \
-  template scoped_refptr<AudioBuffer> MakeInterleavedAudioBuffer<type>( \
-      SampleFormat format,                                              \
-      int channels,                                                     \
-      type start,                                                       \
-      type increment,                                                   \
-      int frames,                                                       \
-      base::TimeDelta start_time,                                       \
+  return output;
+}
+
+// Instantiate all the types of MakeAudioBuffer() and
+// MakeAudioBuffer() needed.
+#define DEFINE_MAKE_AUDIO_BUFFER_INSTANCE(type)              \
+  template scoped_refptr<AudioBuffer> MakeAudioBuffer<type>( \
+      SampleFormat format,                                   \
+      int channels,                                          \
+      type start,                                            \
+      type increment,                                        \
+      int frames,                                            \
+      base::TimeDelta start_time,                            \
       base::TimeDelta duration)
-DEFINE_INTERLEAVED_INSTANCE(uint8);
-DEFINE_INTERLEAVED_INSTANCE(int16);
-DEFINE_INTERLEAVED_INSTANCE(int32);
-DEFINE_INTERLEAVED_INSTANCE(float);
-
-#define DEFINE_PLANAR_INSTANCE(type)                               \
-  template scoped_refptr<AudioBuffer> MakePlanarAudioBuffer<type>( \
-      SampleFormat format,                                         \
-      int channels,                                                \
-      type start,                                                  \
-      type increment,                                              \
-      int frames,                                                  \
-      base::TimeDelta start_time,                                  \
-      base::TimeDelta duration);
-DEFINE_PLANAR_INSTANCE(int16);
-DEFINE_PLANAR_INSTANCE(float);
+DEFINE_MAKE_AUDIO_BUFFER_INSTANCE(uint8);
+DEFINE_MAKE_AUDIO_BUFFER_INSTANCE(int16);
+DEFINE_MAKE_AUDIO_BUFFER_INSTANCE(int32);
+DEFINE_MAKE_AUDIO_BUFFER_INSTANCE(float);
 
 static const char kFakeVideoBufferHeader[] = "FakeVideoBufferForTest";
 
diff --git a/media/base/test_helpers.h b/media/base/test_helpers.h
index 872d08d..ee18f53 100644
--- a/media/base/test_helpers.h
+++ b/media/base/test_helpers.h
@@ -85,9 +85,11 @@ class TestVideoConfig {
 };
 
 // Create an AudioBuffer containing |frames| frames of data, where each sample
-// is of type T. Each frame will have the data from |channels| channels
-// interleaved. |start| and |increment| are used to specify the values for the
-// samples. Since this is interleaved data, channel 0 data will be:
+// is of type T.
+//
+// For interleaved formats, each frame will have the data from |channels|
+// channels interleaved. |start| and |increment| are used to specify the values
+// for the samples. Since this is interleaved data, channel 0 data will be:
 //   |start|
 //   |start| + |channels| * |increment|
 //   |start| + 2 * |channels| * |increment|, and so on.
@@ -95,23 +97,10 @@ class TestVideoConfig {
 // requires data to be of type T, but it is verified that |format| is an
 // interleaved format.
 //
-// |start_time| will be used as the start time for the samples. |duration| is
-// the duration.
-template <class T>
-scoped_refptr<AudioBuffer> MakeInterleavedAudioBuffer(
-    SampleFormat format,
-    int channels,
-    T start,
-    T increment,
-    int frames,
-    base::TimeDelta start_time,
-    base::TimeDelta duration);
-
-// Create an AudioBuffer containing |frames| frames of data, where each sample
-// is of type T. Since this is planar data, there will be a block for each of
-// |channel| channels. |start| and |increment| are used to specify the values
-// for the samples, which are created in channel order. Since this is planar
-// data, channel 0 data will be:
+// For planar formats, there will be a block for each of |channel| channels.
+// |start| and |increment| are used to specify the values for the samples, which
+// are created in channel order. Since this is planar data, channel 0 data will
+// be:
 //   |start|
 //   |start| + |increment|
 //   |start| + 2 * |increment|, and so on.
@@ -122,14 +111,13 @@ scoped_refptr<AudioBuffer> MakeInterleavedAudioBuffer(
 // |start_time| will be used as the start time for the samples. |duration| is
 // the duration.
 template <class T>
-scoped_refptr<AudioBuffer> MakePlanarAudioBuffer(
-    SampleFormat format,
-    int channels,
-    T start,
-    T increment,
-    int frames,
-    base::TimeDelta start_time,
-    base::TimeDelta duration);
+scoped_refptr<AudioBuffer> MakeAudioBuffer(SampleFormat format,
+                                           int channels,
+                                           T start,
+                                           T increment,
+                                           int frames,
+                                           base::TimeDelta timestamp,
+                                           base::TimeDelta duration);
 
 // Create a fake video DecoderBuffer for testing purpose. The buffer contains
 // part of video decoder config info embedded so that the testing code can do
diff --git a/media/filters/audio_renderer_algorithm_unittest.cc b/media/filters/audio_renderer_algorithm_unittest.cc
index aab4a9d..b05e64a 100644
--- a/media/filters/audio_renderer_algorithm_unittest.cc
+++ b/media/filters/audio_renderer_algorithm_unittest.cc
@@ -101,31 +101,31 @@ class AudioRendererAlgorithmTest : public testing::Test {
     while (!algorithm_.IsQueueFull()) {
       switch (sample_format_) {
         case kSampleFormatU8:
-          buffer = MakeInterleavedAudioBuffer<uint8>(sample_format_,
-                                                     channels_,
-                                                     1,
-                                                     1,
-                                                     kFrameSize,
-                                                     kNoTimestamp(),
-                                                     kNoTimestamp());
+          buffer = MakeAudioBuffer<uint8>(sample_format_,
+                                          channels_,
+                                          1,
+                                          1,
+                                          kFrameSize,
+                                          kNoTimestamp(),
+                                          kNoTimestamp());
           break;
         case kSampleFormatS16:
-          buffer = MakeInterleavedAudioBuffer<int16>(sample_format_,
-                                                     channels_,
-                                                     1,
-                                                     1,
-                                                     kFrameSize,
-                                                     kNoTimestamp(),
-                                                     kNoTimestamp());
+          buffer = MakeAudioBuffer<int16>(sample_format_,
+                                          channels_,
+                                          1,
+                                          1,
+                                          kFrameSize,
+                                          kNoTimestamp(),
+                                          kNoTimestamp());
           break;
         case kSampleFormatS32:
-          buffer = MakeInterleavedAudioBuffer<int32>(sample_format_,
-                                                     channels_,
-                                                     1,
-                                                     1,
-                                                     kFrameSize,
-                                                     kNoTimestamp(),
-                                                     kNoTimestamp());
+          buffer = MakeAudioBuffer<int32>(sample_format_,
+                                          channels_,
+                                          1,
+                                          1,
+                                          kFrameSize,
+                                          kNoTimestamp(),
+                                          kNoTimestamp());
           break;
         default:
           NOTREACHED() << "Unrecognized format " << sample_format_;
diff --git a/media/filters/audio_renderer_impl_unittest.cc b/media/filters/audio_renderer_impl_unittest.cc
index c84ccba..ef215a0 100644
--- a/media/filters/audio_renderer_impl_unittest.cc
+++ b/media/filters/audio_renderer_impl_unittest.cc
@@ -302,13 +302,13 @@ class AudioRendererImplTest : public ::testing::Test {
     CHECK(!read_cb_.is_null());
 
     scoped_refptr<AudioBuffer> buffer =
-        MakePlanarAudioBuffer<float>(kSampleFormat,
-                                     kChannels,
-                                     kPlayingAudio,
-                                     0.0f,
-                                     size,
-                                     next_timestamp_->GetTimestamp(),
-                                     next_timestamp_->GetFrameDuration(size));
+        MakeAudioBuffer<float>(kSampleFormat,
+                               kChannels,
+                               kPlayingAudio,
+                               0.0f,
+                               size,
+                               next_timestamp_->GetTimestamp(),
+                               next_timestamp_->GetFrameDuration(size));
     next_timestamp_->AddFrames(size);
 
     DeliverBuffer(AudioDecoder::kOk, buffer);