Add new class AudioBufferQueue.

media/base/audio_buffer_unittest.cc

 // Copyright 2013 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/string_util.h"
 #include "base/strings/stringprintf.h"
 #include "media/base/audio_buffer.h"
 #include "media/base/audio_bus.h"
+#include "media/base/test_helpers.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
 
-template <class T>
-static scoped_refptr<AudioBuffer> MakeInterleavedBuffer(
-    SampleFormat format,
-    int channels,
-    T start,
-    T increment,
-    int frames,
-    const base::TimeDelta start_time) {
-  DCHECK(format == kSampleFormatU8 || format == kSampleFormatS16 ||
-         format == kSampleFormatS32 || format == kSampleFormatF32);
-
-  // Create a block of memory with values:
-  //   start
-  //   start + increment
-  //   start + 2 * increment, ...
-  // Since this is interleaved data, channel 0 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;
-  }
-  // Duration is 1 second per frame (for simplicity).
-  base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
-  return AudioBuffer::CopyFrom(
-      format, channels, frames, data, start_time, duration);
-}
-
-template <class T>
-static scoped_refptr<AudioBuffer> MakePlanarBuffer(
-    SampleFormat format,
-    int channels,
-    T start,
-    T increment,
-    int frames,
-    const base::TimeDelta start_time) {
-  DCHECK(format == kSampleFormatPlanarF32 || format == kSampleFormatPlanarS16);
-
-  // Create multiple blocks of data, once for each channel.
-  // Values in channel 0 will be:
-  //   start
-  //   start + increment
-  //   start + 2 * increment, ...
-  // 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;
-      start += increment;
-    }
-  }
-  // Duration is 1 second per frame (for simplicity).
-  base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
-  return AudioBuffer::CopyFrom(
-      format, channels, frames, data.get(), start_time, duration);
-}
-
 static void VerifyResult(float* channel_data,
                          int frames,
                          float start,
                          float increment) {
   for (int i = 0; i < frames; ++i) {
     SCOPED_TRACE(base::StringPrintf(
         "i=%d/%d start=%f, increment=%f", i, frames, start, increment));
     ASSERT_EQ(channel_data[i], start);
     start += increment;
   }
 }
 
 TEST(AudioBufferTest, CopyFrom) {
   const int channels = 1;
   const int frames = 8;
   const base::TimeDelta start_time;
-  scoped_refptr<AudioBuffer> buffer = MakeInterleavedBuffer<uint8>(
+  scoped_refptr<AudioBuffer> buffer = MakeInterleavedAudioBuffer<uint8>(
       kSampleFormatU8, channels, 1, 1, frames, start_time);
   EXPECT_EQ(frames, buffer->frame_count());
   EXPECT_EQ(buffer->timestamp(), start_time);
   EXPECT_EQ(buffer->duration().InSeconds(), frames);
   EXPECT_FALSE(buffer->end_of_stream());
 }
 
 TEST(AudioBufferTest, CreateEOSBuffer) {
   scoped_refptr<AudioBuffer> buffer = AudioBuffer::CreateEOSBuffer();
   EXPECT_TRUE(buffer->end_of_stream());
@@ skipping 13 matching lines @@
 
   buffer = AudioBuffer::CopyFrom(
       kSampleFormatF32, 4, 2, data, kTimestampA, kTimestampB);
   EXPECT_EQ(2, buffer->frame_count());  // now 4 channels of 32-bit data
 }
 
 TEST(AudioBufferTest, ReadU8) {
   const int channels = 4;
   const int frames = 4;
   const base::TimeDelta start_time;
-  scoped_refptr<AudioBuffer> buffer = MakeInterleavedBuffer<uint8>(
+  scoped_refptr<AudioBuffer> buffer = MakeInterleavedAudioBuffer<uint8>(
       kSampleFormatU8, channels, 128, 1, frames, start_time);
 
   // Read all 4 frames from the buffer. Data is interleaved, so ch[0] should be
   // 128, 132, 136, 140, other channels similar. However, values are converted
   // from [0, 255] to [-1.0, 1.0] with a bias of 128. Thus the first buffer
   // value should be 0.0, then 1/127, 2/127, etc.
   scoped_ptr<AudioBus> bus = AudioBus::Create(channels, 100);
   buffer->ReadFrames(frames, 0, 0, bus.get());
   VerifyResult(bus->channel(0), frames, 0.0f, 4.0f / 127.0f);
   VerifyResult(bus->channel(1), frames, 1.0f / 127.0f, 4.0f / 127.0f);
   VerifyResult(bus->channel(2), frames, 2.0f / 127.0f, 4.0f / 127.0f);
   VerifyResult(bus->channel(3), frames, 3.0f / 127.0f, 4.0f / 127.0f);
 }
 
 TEST(AudioBufferTest, ReadS16) {
   const int channels = 2;
   const int frames = 10;
   const base::TimeDelta start_time;
-  scoped_refptr<AudioBuffer> buffer = MakeInterleavedBuffer<int16>(
+  scoped_refptr<AudioBuffer> buffer = MakeInterleavedAudioBuffer<int16>(
       kSampleFormatS16, channels, 1, 1, frames, start_time);
 
   // Read 6 frames from the buffer. Data is interleaved, so ch[0] should be 1,
   // 3, 5, 7, 9, 11, and ch[1] should be 2, 4, 6, 8, 10, 12. Data is converted
   // to float from -1.0 to 1.0 based on int16 range.
   scoped_ptr<AudioBus> bus = AudioBus::Create(channels, 100);
   buffer->ReadFrames(6, 0, 0, bus.get());
   VerifyResult(bus->channel(0), 6, 1.0f / kint16max, 2.0f / kint16max);
   VerifyResult(bus->channel(1), 6, 2.0f / kint16max, 2.0f / kint16max);
 
   // Now read the same data one frame at a time.
   bus = AudioBus::Create(channels, 100);
   for (int i = 0; i < frames; ++i) {
     buffer->ReadFrames(1, i, i, bus.get());
   }
   VerifyResult(bus->channel(0), frames, 1.0f / kint16max, 2.0f / kint16max);
   VerifyResult(bus->channel(1), frames, 2.0f / kint16max, 2.0f / kint16max);
 }
 
 TEST(AudioBufferTest, ReadS32) {
   const int channels = 2;
   const int frames = 6;
   const base::TimeDelta start_time;
-  scoped_refptr<AudioBuffer> buffer = MakeInterleavedBuffer<int32>(
+  scoped_refptr<AudioBuffer> buffer = MakeInterleavedAudioBuffer<int32>(
       kSampleFormatS32, channels, 1, 1, frames, start_time);
 
   // Read 6 frames from the buffer. Data is interleaved, so ch[0] should be 1,
   // 3, 5, 7, 9, 11, and ch[1] should be 2, 4, 6, 8, 10, 12. Data is converted
   // to float from -1.0 to 1.0 based on int32 range.
   scoped_ptr<AudioBus> bus = AudioBus::Create(channels, 100);
   buffer->ReadFrames(frames, 0, 0, bus.get());
   VerifyResult(bus->channel(0), frames, 1.0f / kint32max, 2.0f / kint32max);
   VerifyResult(bus->channel(1), frames, 2.0f / kint32max, 2.0f / kint32max);
 
   // Now read 2 frames starting at frame offset 3. ch[0] should be 7, 9, and
   // ch[1] should be 8, 10.
   buffer->ReadFrames(2, 3, 0, bus.get());
   VerifyResult(bus->channel(0), 2, 7.0f / kint32max, 2.0f / kint32max);
   VerifyResult(bus->channel(1), 2, 8.0f / kint32max, 2.0f / kint32max);
 }
 
 TEST(AudioBufferTest, ReadF32) {
   const int channels = 2;
   const int frames = 20;
   const base::TimeDelta start_time;
-  scoped_refptr<AudioBuffer> buffer = MakeInterleavedBuffer<float>(
+  scoped_refptr<AudioBuffer> buffer = MakeInterleavedAudioBuffer<float>(
       kSampleFormatF32, channels, 1.0f, 1.0f, frames, start_time);
 
   // Read first 10 frames from the buffer. F32 is interleaved, so ch[0] should
   // be 1, 3, 5, ... and ch[1] should be 2, 4, 6, ...
   scoped_ptr<AudioBus> bus = AudioBus::Create(channels, 100);
   buffer->ReadFrames(10, 0, 0, bus.get());
   VerifyResult(bus->channel(0), 10, 1.0f, 2.0f);
   VerifyResult(bus->channel(1), 10, 2.0f, 2.0f);
 
   // Read second 10 frames.
   bus = AudioBus::Create(channels, 100);
   buffer->ReadFrames(10, 10, 0, bus.get());
   VerifyResult(bus->channel(0), 10, 21.0f, 2.0f);
   VerifyResult(bus->channel(1), 10, 22.0f, 2.0f);
 }
 
 TEST(AudioBufferTest, ReadS16Planar) {
   const int channels = 2;
   const int frames = 20;
   const base::TimeDelta start_time;
-  scoped_refptr<AudioBuffer> buffer = MakePlanarBuffer<int16>(
+  scoped_refptr<AudioBuffer> buffer = MakePlanarAudioBuffer<int16>(
       kSampleFormatPlanarS16, channels, 1, 1, frames, start_time);
 
   // Read 6 frames from the buffer. Data is planar, so ch[0] should be 1, 2, 3,
   // 4, 5, 6, and ch[1] should be 21, 22, 23, 24, 25, 26. Data is converted to
   // float from -1.0 to 1.0 based on int16 range.
   scoped_ptr<AudioBus> bus = AudioBus::Create(channels, 100);
   buffer->ReadFrames(6, 0, 0, bus.get());
   VerifyResult(bus->channel(0), 6, 1.0f / kint16max, 1.0f / kint16max);
   VerifyResult(bus->channel(1), 6, 21.0f / kint16max, 1.0f / kint16max);
 
@@ skipping 11 matching lines @@
   buffer->ReadFrames(0, 0, 10, bus.get());
   buffer->ReadFrames(0, 10, 0, bus.get());
   VerifyResult(bus->channel(0), frames, 20.0f / kint16max, -1.0f / kint16max);
   VerifyResult(bus->channel(1), frames, 40.0f / kint16max, -1.0f / kint16max);
 }
 
 TEST(AudioBufferTest, ReadF32Planar) {
   const int channels = 4;
   const int frames = 100;
   const base::TimeDelta start_time;
-  scoped_refptr<AudioBuffer> buffer = MakePlanarBuffer<float>(
+  scoped_refptr<AudioBuffer> buffer = MakePlanarAudioBuffer<float>(
       kSampleFormatPlanarF32, channels, 1.0f, 1.0f, frames, start_time);
 
   // 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
   // channels.
   scoped_ptr<AudioBus> bus = AudioBus::Create(channels, 100);
   buffer->ReadFrames(frames, 0, 0, bus.get());
   VerifyResult(bus->channel(0), frames, 1.0f, 1.0f);
   VerifyResult(bus->channel(1), frames, 101.0f, 1.0f);
   VerifyResult(bus->channel(2), frames, 201.0f, 1.0f);
   VerifyResult(bus->channel(3), frames, 301.0f, 1.0f);
 
   // Now read 20 frames from the middle of the buffer.
   bus = AudioBus::Create(channels, 100);
   buffer->ReadFrames(20, 50, 0, bus.get());
   VerifyResult(bus->channel(0), 20, 51.0f, 1.0f);
   VerifyResult(bus->channel(1), 20, 151.0f, 1.0f);
   VerifyResult(bus->channel(2), 20, 251.0f, 1.0f);
   VerifyResult(bus->channel(3), 20, 351.0f, 1.0f);
 }
 
 }  // namespace media