Fix a couple bugs with OSX audio buffer size selection.

1. Device changes may cause an already in process stream creation
to use the wrong buffer size => CHECK() failure.

2. Low latency stream Open() failure causes AudioOutputResampler
to fall back to a high latency buffer size, which will cause a
stream to be created with the wrong buffer size => CHECK() failure.

Fixing 2 has the added advantage of removing an extra useless hop
between stream Open() failure and switching to a fake audio output
device.

BUG=225249
TEST=audio works.


Review URL: https://chromiumcodereview.appspot.com/13390010

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@192946 0039d316-1c4b-4281-b951-d872f2087c98

3 files changed, 42 insertions, 16 deletions

diff --git a/media/audio/audio_output_proxy_unittest.cc b/media/audio/audio_output_proxy_unittest.cc
index cb20bd0..224bbf5 100644
--- a/media/audio/audio_output_proxy_unittest.cc
+++ b/media/audio/audio_output_proxy_unittest.cc
@@ -628,8 +628,15 @@ TEST_F(AudioOutputResamplerTest, LowLatencyOpenFailedFallback) {
 TEST_F(AudioOutputResamplerTest, HighLatencyFallbackFailed) {
   MockAudioOutputStream okay_stream(&manager_, params_);
 
+// Only Windows has a high latency output driver that is not the same as the low
+// latency path.
+#if defined(OS_WIN)
+  static const int kFallbackCount = 2;
+#else
+  static const int kFallbackCount = 1;
+#endif
   EXPECT_CALL(manager(), MakeAudioOutputStream(_))
-      .Times(2)
+      .Times(kFallbackCount)
       .WillRepeatedly(Return(static_cast<AudioOutputStream*>(NULL)));
 
   // To prevent shared memory issues the sample rate and buffer size should
@@ -656,8 +663,15 @@ TEST_F(AudioOutputResamplerTest, HighLatencyFallbackFailed) {
 // stream, and the fake audio output stream and ensure AudioOutputResampler
 // terminates normally.
 TEST_F(AudioOutputResamplerTest, AllFallbackFailed) {
+// Only Windows has a high latency output driver that is not the same as the low
+// latency path.
+#if defined(OS_WIN)
+  static const int kFallbackCount = 3;
+#else
+  static const int kFallbackCount = 2;
+#endif
   EXPECT_CALL(manager(), MakeAudioOutputStream(_))
-      .Times(3)
+      .Times(kFallbackCount)
       .WillRepeatedly(Return(static_cast<AudioOutputStream*>(NULL)));
 
   AudioOutputProxy* proxy = new AudioOutputProxy(resampler_);
diff --git a/media/audio/audio_output_resampler.cc b/media/audio/audio_output_resampler.cc
index 5abea0a..26d83f8 100644
--- a/media/audio/audio_output_resampler.cc
+++ b/media/audio/audio_output_resampler.cc
@@ -124,6 +124,9 @@ static void RecordFallbackStats(const AudioParameters& output_params) {
   }
 }
 
+// Only Windows has a high latency output driver that is not the same as the low
+// latency path.
+#if defined(OS_WIN)
 // Converts low latency based |output_params| into high latency appropriate
 // output parameters in error situations.
 static AudioParameters SetupFallbackParams(
@@ -142,6 +145,7 @@ static AudioParameters SetupFallbackParams(
       input_params.sample_rate(), input_params.bits_per_sample(),
       frames_per_buffer);
 }
+#endif
 
 AudioOutputResampler::AudioOutputResampler(AudioManager* audio_manager,
                                            const AudioParameters& input_params,
@@ -202,18 +206,23 @@ bool AudioOutputResampler::OpenStream() {
     return false;
   }
 
-  DLOG(ERROR) << "Unable to open audio device in low latency mode.  Falling "
-              << "back to high latency audio output.";
-
   // Record UMA statistics about the hardware which triggered the failure so
   // we can debug and triage later.
   RecordFallbackStats(output_params_);
+
+  // Only Windows has a high latency output driver that is not the same as the
+  // low latency path.
+#if defined(OS_WIN)
+  DLOG(ERROR) << "Unable to open audio device in low latency mode.  Falling "
+              << "back to high latency audio output.";
+
   output_params_ = SetupFallbackParams(params_, output_params_);
   Initialize();
   if (dispatcher_->OpenStream()) {
     streams_opened_ = true;
     return true;
   }
+#endif
 
   DLOG(ERROR) << "Unable to open audio device in high latency mode.  Falling "
               << "back to fake audio output.";
diff --git a/media/audio/mac/audio_low_latency_output_mac.cc b/media/audio/mac/audio_low_latency_output_mac.cc
index 592a6a1..afa480a 100644
--- a/media/audio/mac/audio_low_latency_output_mac.cc
+++ b/media/audio/mac/audio_low_latency_output_mac.cc
@@ -78,12 +78,8 @@ AUAudioOutputStream::AUAudioOutputStream(
   DVLOG(1) << "Desired ouput format: " << format_;
 
   // Calculate the number of sample frames per callback.
-  number_of_frames_ = params.GetBytesPerBuffer() / format_.mBytesPerPacket;
+  number_of_frames_ = params.frames_per_buffer();
   DVLOG(1) << "Number of frames per callback: " << number_of_frames_;
-  const AudioParameters parameters =
-      manager_->GetDefaultOutputStreamParameters();
-  CHECK_EQ(number_of_frames_,
-           static_cast<size_t>(parameters.frames_per_buffer()));
 }
 
 AUAudioOutputStream::~AUAudioOutputStream() {
@@ -99,7 +95,7 @@ bool AUAudioOutputStream::Open() {
                                                &size,
                                                &output_device_id_);
   if (result != noErr || output_device_id_ == kAudioObjectUnknown) {
-    OSSTATUS_DLOG(WARNING, result)
+    OSSTATUS_DLOG(ERROR, result)
         << "Could not get default audio output device.";
     return false;
   }
@@ -119,13 +115,13 @@ bool AUAudioOutputStream::Open() {
 
   result = AudioComponentInstanceNew(comp, &output_unit_);
   if (result != noErr) {
-    OSSTATUS_DLOG(WARNING, result) << "AudioComponentInstanceNew() failed.";
+    OSSTATUS_DLOG(ERROR, result) << "AudioComponentInstanceNew() failed.";
     return false;
   }
 
   result = AudioUnitInitialize(output_unit_);
   if (result != noErr) {
-    OSSTATUS_DLOG(WARNING, result) << "AudioUnitInitialize() failed.";
+    OSSTATUS_DLOG(ERROR, result) << "AudioUnitInitialize() failed.";
     return false;
   }
 
@@ -147,7 +143,7 @@ bool AUAudioOutputStream::Configure() {
       &input,
       sizeof(input));
   if (result != noErr) {
-    OSSTATUS_DLOG(WARNING, result)
+    OSSTATUS_DLOG(ERROR, result)
       << "AudioUnitSetProperty(kAudioUnitProperty_SetRenderCallback) failed.";
     return false;
   }
@@ -161,7 +157,7 @@ bool AUAudioOutputStream::Configure() {
       &format_,
       sizeof(format_));
   if (result != noErr) {
-    OSSTATUS_DLOG(WARNING, result)
+    OSSTATUS_DLOG(ERROR, result)
         << "AudioUnitSetProperty(kAudioUnitProperty_StreamFormat) failed.";
     return false;
   }
@@ -172,6 +168,13 @@ bool AUAudioOutputStream::Configure() {
   // be the same as the frames_per_buffer() returned by
   // GetDefaultOutputStreamParameters.
   // See http://crbug.com/154352 for details.
+  const AudioParameters hw_params =
+      manager_->GetDefaultOutputStreamParameters();
+  if (number_of_frames_ != static_cast<size_t>(hw_params.frames_per_buffer())) {
+    DLOG(ERROR) << "Audio buffer size does not match hardware buffer size.";
+    return false;
+  }
+
   UInt32 buffer_size = number_of_frames_;
   result = AudioUnitSetProperty(
       output_unit_,
@@ -181,7 +184,7 @@ bool AUAudioOutputStream::Configure() {
       &buffer_size,
       sizeof(buffer_size));
   if (result != noErr) {
-    OSSTATUS_DLOG(WARNING, result)
+    OSSTATUS_DLOG(ERROR, result)
         << "AudioUnitSetProperty(kAudioDevicePropertyBufferFrameSize) failed.";
     return false;
   }