Fix incorrect AudioConverter setup for discrete channel layouts.

media/base/audio_converter_unittest.cc

 // 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.
 
 // MSVC++ requires this to be set before any other includes to get M_PI.
 #define _USE_MATH_DEFINES
 
 #include <cmath>
 
 #include "base/memory/scoped_ptr.h"
@@ skipping 176 matching lines @@
   scoped_ptr<FakeAudioRenderCallback> expected_callback_;
 
   // Epsilon value with which to perform comparisons between |audio_bus_| and
   // |expected_audio_bus_|.
   double epsilon_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioConverterTest);
 };
 
 // Ensure the buffer delay provided by AudioConverter is accurate.
-TEST(AudioConverterTest, AudioDelay) {
+TEST(AudioConverterTest, AudioDelayAndDiscreteChannelCount) {
   // Choose input and output parameters such that the transform must make
   // multiple calls to fill the buffer.
-  AudioParameters input_parameters = AudioParameters(
-      AudioParameters::AUDIO_PCM_LINEAR, kChannelLayout, kSampleRate,
-      kBitsPerChannel, kLowLatencyBufferSize);
-  AudioParameters output_parameters = AudioParameters(
-      AudioParameters::AUDIO_PCM_LINEAR, kChannelLayout, kSampleRate * 2,
-      kBitsPerChannel, kHighLatencyBufferSize);
+  AudioParameters input_parameters(AudioParameters::AUDIO_PCM_LINEAR,
+                                   CHANNEL_LAYOUT_DISCRETE, 10, kSampleRate,
+                                   kBitsPerChannel, kLowLatencyBufferSize,
+                                   AudioParameters::NO_EFFECTS);
+  AudioParameters output_parameters(AudioParameters::AUDIO_PCM_LINEAR,
+                                    CHANNEL_LAYOUT_DISCRETE, 5, kSampleRate * 2,
+                                    kBitsPerChannel, kHighLatencyBufferSize,
+                                    AudioParameters::NO_EFFECTS);
 
   AudioConverter converter(input_parameters, output_parameters, false);
   FakeAudioRenderCallback callback(0.2);
   scoped_ptr<AudioBus> audio_bus = AudioBus::Create(output_parameters);
   converter.AddInput(&callback);
   converter.Convert(audio_bus.get());
 
   // Calculate the expected buffer delay for given AudioParameters.
   double input_sample_rate = input_parameters.sample_rate();
   int fill_count =
       (output_parameters.frames_per_buffer() * input_sample_rate /
        output_parameters.sample_rate()) / input_parameters.frames_per_buffer();
 
   base::TimeDelta input_frame_duration = base::TimeDelta::FromMicroseconds(
       base::Time::kMicrosecondsPerSecond / input_sample_rate);
 
   int expected_last_delay_milliseconds =
       fill_count * input_parameters.frames_per_buffer() *
       input_frame_duration.InMillisecondsF();
 
   EXPECT_EQ(expected_last_delay_milliseconds,
             callback.last_audio_delay_milliseconds());
+  EXPECT_EQ(input_parameters.channels(), callback.last_channel_count());
 }
 
 TEST_P(AudioConverterTest, ArbitraryOutputRequestSize) {
   // Resize output bus to be half of |output_parameters_|'s frames_per_buffer().
   audio_bus_ = AudioBus::Create(output_parameters_.channels(),
                                 output_parameters_.frames_per_buffer() / 2);
   RunTest(1);
 }
 
 TEST_P(AudioConverterTest, NoInputs) {
@@ skipping 14 matching lines @@
         // No resampling. No channel mixing.
         std::tr1::make_tuple(44100, 44100, CHANNEL_LAYOUT_STEREO, 0.00000048),
 
         // Upsampling. Channel upmixing.
         std::tr1::make_tuple(44100, 48000, CHANNEL_LAYOUT_QUAD, 0.033),
 
         // Downsampling. Channel downmixing.
         std::tr1::make_tuple(48000, 41000, CHANNEL_LAYOUT_MONO, 0.042)));
 
 }  // namespace media