Add channel_count parameter back to AudioBuffer creation methods.

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/strings/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"
@@ skipping 12 matching lines @@
     ASSERT_EQ(channel_data[i], start);
     start += increment;
   }
 }
 
 TEST(AudioBufferTest, CopyFrom) {
   const ChannelLayout channel_layout = CHANNEL_LAYOUT_MONO;
   const int frames = 8;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
-  scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<uint8>(kSampleFormatU8,
-                                                             channel_layout,
-                                                             kSampleRate,
-                                                             1,
-                                                             1,
-                                                             frames,
-                                                             start_time,
-                                                             duration);
+  scoped_refptr<AudioBuffer> buffer =
+      MakeAudioBuffer<uint8>(kSampleFormatU8,
+                             channel_layout,
+                             ChannelLayoutToChannelCount(channel_layout),
+                             kSampleRate,
+                             1,
+                             1,
+                             frames,
+                             start_time,
+                             duration);
   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());
 }
 
 TEST(AudioBufferTest, FrameSize) {
   const uint8 kTestData[] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14,
                               15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26,
                               27, 28, 29, 30, 31 };
   const base::TimeDelta kTimestampA = base::TimeDelta::FromMicroseconds(1337);
   const base::TimeDelta kTimestampB = base::TimeDelta::FromMicroseconds(1234);
 
   const uint8* const data[] = { kTestData };
   scoped_refptr<AudioBuffer> buffer =
       AudioBuffer::CopyFrom(kSampleFormatU8,
                             CHANNEL_LAYOUT_STEREO,
+                            2,
                             kSampleRate,
                             16,
                             data,
                             kTimestampA,
                             kTimestampB);
   EXPECT_EQ(16, buffer->frame_count());  // 2 channels of 8-bit data
 
   buffer = AudioBuffer::CopyFrom(kSampleFormatF32,
                                  CHANNEL_LAYOUT_4_0,
+                                 4,
                                  kSampleRate,
                                  2,
                                  data,
                                  kTimestampA,
                                  kTimestampB);
   EXPECT_EQ(2, buffer->frame_count());  // now 4 channels of 32-bit data
 }
 
 TEST(AudioBufferTest, ReadU8) {
   const ChannelLayout channel_layout = CHANNEL_LAYOUT_4_0;
   const int channels = ChannelLayoutToChannelCount(channel_layout);
   const int frames = 4;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
   scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<uint8>(kSampleFormatU8,
                                                              channel_layout,
+                                                             channels,
                                                              kSampleRate,
                                                              128,
                                                              1,
                                                              frames,
                                                              start_time,
                                                              duration);
 
   // 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 ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO;
   const int channels = ChannelLayoutToChannelCount(channel_layout);
   const int frames = 10;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
   scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<int16>(kSampleFormatS16,
                                                              channel_layout,
+                                                             channels,
                                                              kSampleRate,
                                                              1,
                                                              1,
                                                              frames,
                                                              start_time,
                                                              duration);
 
   // 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.
@@ skipping 12 matching lines @@
 }
 
 TEST(AudioBufferTest, ReadS32) {
   const ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO;
   const int channels = ChannelLayoutToChannelCount(channel_layout);
   const int frames = 6;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
   scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<int32>(kSampleFormatS32,
                                                              channel_layout,
+                                                             channels,
                                                              kSampleRate,
                                                              1,
                                                              1,
                                                              frames,
                                                              start_time,
                                                              duration);
 
   // 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.
@@ skipping 10 matching lines @@
 }
 
 TEST(AudioBufferTest, ReadF32) {
   const ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO;
   const int channels = ChannelLayoutToChannelCount(channel_layout);
   const int frames = 20;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
   scoped_refptr<AudioBuffer> buffer = MakeAudioBuffer<float>(kSampleFormatF32,
                                                              channel_layout,
+                                                             channels,
                                                              kSampleRate,
                                                              1.0f,
                                                              1.0f,
                                                              frames,
                                                              start_time,
                                                              duration);
 
   // 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);
@@ skipping 10 matching lines @@
 
 TEST(AudioBufferTest, ReadS16Planar) {
   const ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO;
   const int channels = ChannelLayoutToChannelCount(channel_layout);
   const int frames = 20;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
   scoped_refptr<AudioBuffer> buffer =
       MakeAudioBuffer<int16>(kSampleFormatPlanarS16,
                              channel_layout,
+                             channels,
                              kSampleRate,
                              1,
                              1,
                              frames,
                              start_time,
                              duration);
 
   // 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.
@@ skipping 21 matching lines @@
 
 TEST(AudioBufferTest, ReadF32Planar) {
   const ChannelLayout channel_layout = CHANNEL_LAYOUT_4_0;
   const int channels = ChannelLayoutToChannelCount(channel_layout);
   const int frames = 100;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
   scoped_refptr<AudioBuffer> buffer =
       MakeAudioBuffer<float>(kSampleFormatPlanarF32,
                              channel_layout,
+                             channels,
                              kSampleRate,
                              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
   // channels.
@@ skipping 13 matching lines @@
   VerifyResult(bus->channel(3), 20, 351.0f, 1.0f);
 }
 
 TEST(AudioBufferTest, EmptyBuffer) {
   const ChannelLayout channel_layout = CHANNEL_LAYOUT_4_0;
   const int channels = ChannelLayoutToChannelCount(channel_layout);
   const int frames = 100;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
   scoped_refptr<AudioBuffer> buffer = AudioBuffer::CreateEmptyBuffer(
-      channel_layout, kSampleRate, frames, start_time, duration);
+      channel_layout, channels, kSampleRate, frames, start_time, duration);
   EXPECT_EQ(frames, buffer->frame_count());
   EXPECT_EQ(start_time, buffer->timestamp());
   EXPECT_EQ(frames, buffer->duration().InSeconds());
   EXPECT_FALSE(buffer->end_of_stream());
 
   // Read all 100 frames from the buffer. All data should be 0.
   scoped_ptr<AudioBus> bus = AudioBus::Create(channels, 100);
   buffer->ReadFrames(frames, 0, 0, bus.get());
   VerifyResult(bus->channel(0), frames, 0.0f, 0.0f);
   VerifyResult(bus->channel(1), frames, 0.0f, 0.0f);
   VerifyResult(bus->channel(2), frames, 0.0f, 0.0f);
   VerifyResult(bus->channel(3), frames, 0.0f, 0.0f);
 }
 
 TEST(AudioBufferTest, Trim) {
   const ChannelLayout channel_layout = CHANNEL_LAYOUT_4_0;
   const int channels = ChannelLayoutToChannelCount(channel_layout);
   const int frames = 100;
   const base::TimeDelta start_time;
   const base::TimeDelta duration = base::TimeDelta::FromSeconds(frames);
   scoped_refptr<AudioBuffer> buffer =
       MakeAudioBuffer<float>(kSampleFormatPlanarF32,
                              channel_layout,
+                             channels,
                              kSampleRate,
                              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());
 
@@ skipping 26 matching lines @@
   VerifyResult(bus->channel(0), 10, 61.0f, 1.0f);
 
   // Trim off the last 30 frames.
   buffer->TrimEnd(30);
   EXPECT_EQ(buffer->frame_count(), 0);
   EXPECT_EQ(buffer->timestamp(), start_time + base::TimeDelta::FromSeconds(60));
   EXPECT_EQ(buffer->duration(), base::TimeDelta::FromSeconds(0));
 }
 
 }  // namespace media