Add support for midstream audio configuration changes.

media/filters/audio_renderer_impl_unittest.cc

Index: media/filters/audio_renderer_impl_unittest.cc
diff --git a/media/filters/audio_renderer_impl_unittest.cc b/media/filters/audio_renderer_impl_unittest.cc
index 965e5cfac6f5e44e95f7edc0be9df2f0c8f0164a..393b103e38fab8f4d3cacb8fd8ece1f553d625dc 100644
--- a/media/filters/audio_renderer_impl_unittest.cc
+++ b/media/filters/audio_renderer_impl_unittest.cc
@@ -11,6 +11,7 @@
 #include "base/stl_util.h"
 #include "base/strings/stringprintf.h"
 #include "media/base/audio_buffer.h"
+#include "media/base/audio_hardware_config.h"
 #include "media/base/audio_timestamp_helper.h"
 #include "media/base/fake_audio_renderer_sink.h"
 #include "media/base/gmock_callback_support.h"
@@ -51,7 +52,8 @@ class AudioRendererImplTest : public ::testing::Test {
  public:
   // Give the decoder some non-garbage media properties.
   AudioRendererImplTest()
-      : needs_stop_(true),
+      : hardware_config_(AudioParameters(), AudioParameters()),
+        needs_stop_(true),
         demuxer_stream_(DemuxerStream::AUDIO),
         decoder_(new MockAudioDecoder()) {
     AudioDecoderConfig audio_config(kCodec,
@@ -73,18 +75,16 @@ class AudioRendererImplTest : public ::testing::Test {
     EXPECT_CALL(*decoder_, Stop(_))
         .WillRepeatedly(Invoke(this, &AudioRendererImplTest::StopDecoder));
 
-    // Set up audio properties.
-    EXPECT_CALL(*decoder_, bits_per_channel())
-        .WillRepeatedly(Return(audio_config.bits_per_channel()));
-    EXPECT_CALL(*decoder_, channel_layout())
-        .WillRepeatedly(Return(audio_config.channel_layout()));
-    EXPECT_CALL(*decoder_, samples_per_second())
-        .WillRepeatedly(Return(audio_config.samples_per_second()));
-
     // Mock out demuxer reads
     EXPECT_CALL(demuxer_stream_, Read(_)).WillRepeatedly(
         RunCallback<0>(DemuxerStream::kOk, DecoderBuffer::CreateEOSBuffer()));
-
+    AudioParameters out_params =
+        AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY,
+                        kChannelLayout,
+                        kSamplesPerSecond,
+                        SampleFormatToBytesPerChannel(kSampleFormat) * 8,
+                        512);
+    hardware_config_.UpdateOutputConfig(out_params);
     ScopedVector<AudioDecoder> decoders;
     decoders.push_back(decoder_);
     sink_ = new FakeAudioRendererSink();
@@ -92,7 +92,8 @@ class AudioRendererImplTest : public ::testing::Test {
         message_loop_.message_loop_proxy(),
         sink_,
         decoders.Pass(),
-        SetDecryptorReadyCB()));
+        SetDecryptorReadyCB(),
+        hardware_config_));
 
     // Stub out time.
     renderer_->set_now_cb_for_testing(base::Bind(
@@ -114,14 +115,14 @@ class AudioRendererImplTest : public ::testing::Test {
   }
 
   void ExpectUnsupportedAudioDecoderConfig() {
-    EXPECT_CALL(*decoder_, bits_per_channel())
-        .WillRepeatedly(Return(3));
-    EXPECT_CALL(*decoder_, channel_layout())
-        .WillRepeatedly(Return(CHANNEL_LAYOUT_UNSUPPORTED));
-    EXPECT_CALL(*decoder_, samples_per_second())
-        .WillRepeatedly(Return(0));
     EXPECT_CALL(*decoder_, Initialize(_, _))
         .WillOnce(RunCallback<1>(PIPELINE_OK));
+    hardware_config_.UpdateOutputConfig(
+        AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY,
+                        CHANNEL_LAYOUT_UNSUPPORTED,
+                        0,
+                        3,
+                        512));
   }
 
   MOCK_METHOD1(OnStatistics, void(const PipelineStatistics&));
@@ -138,8 +139,8 @@ class AudioRendererImplTest : public ::testing::Test {
         .WillOnce(RunCallback<1>(PIPELINE_OK));
     InitializeWithStatus(PIPELINE_OK);
 
-    next_timestamp_.reset(
-        new AudioTimestampHelper(decoder_->samples_per_second()));
+    next_timestamp_.reset(new AudioTimestampHelper(
+        hardware_config_.GetOutputConfig().sample_rate()));
   }
 
   void InitializeWithStatus(PipelineStatus expected) {
@@ -368,7 +369,7 @@ class AudioRendererImplTest : public ::testing::Test {
     do {
       TimeDelta audio_delay = TimeDelta::FromMicroseconds(
           total_frames_read * Time::kMicrosecondsPerSecond /
-          static_cast<float>(decoder_->samples_per_second()));
+          static_cast<float>(hardware_config_.GetOutputConfig().sample_rate()));
 
       frames_read = renderer_->Render(
           bus.get(), audio_delay.InMilliseconds());
@@ -460,6 +461,7 @@ class AudioRendererImplTest : public ::testing::Test {
   base::MessageLoop message_loop_;
   scoped_ptr<AudioRendererImpl> renderer_;
   scoped_refptr<FakeAudioRendererSink> sink_;
+  AudioHardwareConfig hardware_config_;
 
   // Whether or not the test needs the destructor to call Stop() on
   // |renderer_| at destruction.