Prevent AudioLowLatencyInputMac from changing frame sizes.

Otherwise we'll hit a CHECK() later since the shared memory isn't configured for the new frame size.

BUG=154352
TEST=mac build no longer crashes.


Review URL: https://chromiumcodereview.appspot.com/11086009

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@160759 0039d316-1c4b-4281-b951-d872f2087c98

3 files changed, 39 insertions, 16 deletions

diff --git a/media/audio/mac/audio_low_latency_input_mac.cc b/media/audio/mac/audio_low_latency_input_mac.cc
index dc9bd8a..276f3e7 100644
--- a/media/audio/mac/audio_low_latency_input_mac.cc
+++ b/media/audio/mac/audio_low_latency_input_mac.cc
@@ -195,6 +195,11 @@ bool AUAudioInputStream::Open() {
   }
 
   // Set the desired number of frames in the IO buffer (output scope).
+  // WARNING: Setting this value changes the frame size for all audio units in
+  // the current process.  It's imperative that the input and output frame sizes
+  // be the same as audio_util::GetAudioHardwareBufferSize().
+  // TODO(henrika): Due to http://crrev.com/159666 this is currently not true
+  // and should be fixed, a CHECK() should be added at that time.
   result = AudioUnitSetProperty(audio_unit_,
                                 kAudioDevicePropertyBufferFrameSize,
                                 kAudioUnitScope_Output,
@@ -460,6 +465,18 @@ OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
   if (!audio_data)
     return kAudioUnitErr_InvalidElement;
 
+  // Unfortunately AUAudioInputStream and AUAudioOutputStream share the frame
+  // size set by kAudioDevicePropertyBufferFrameSize above on a per process
+  // basis.  What this means is that the |number_of_frames| value may be larger
+  // or smaller than the value set during Configure().  In this case either
+  // audio input or audio output will be broken, so just do nothing.
+  // TODO(henrika): This should never happen so long as we're always using the
+  // hardware sample rate and the input/output streams configure the same frame
+  // size.  This is currently not true.  See http://crbug.com/154352.  Once
+  // fixed, a CHECK() should be added and this wall of text removed.
+  if (number_of_frames != static_cast<UInt32>(number_of_frames_))
+    return noErr;
+
   // Deliver data packet, delay estimation and volume level to the user.
   sink_->OnData(this,
                 audio_data,
diff --git a/media/audio/mac/audio_low_latency_output_mac.cc b/media/audio/mac/audio_low_latency_output_mac.cc
index 0327db0..ea0d04d 100644
--- a/media/audio/mac/audio_low_latency_output_mac.cc
+++ b/media/audio/mac/audio_low_latency_output_mac.cc
@@ -154,6 +154,11 @@ bool AUAudioOutputStream::Configure() {
     return false;
 
   // Set the buffer frame size.
+  // WARNING: Setting this value changes the frame size for all audio units in
+  // the current process.  It's imperative that the input and output frame sizes
+  // be the same as audio_util::GetAudioHardwareBufferSize().
+  // TODO(henrika): Due to http://crrev.com/159666 this is currently not true
+  // and should be fixed, a CHECK() should be added at that time.
   UInt32 buffer_size = number_of_frames_;
   result = AudioUnitSetProperty(
       output_unit_,
@@ -222,10 +227,6 @@ void AUAudioOutputStream::GetVolume(double* volume) {
 OSStatus AUAudioOutputStream::Render(UInt32 number_of_frames,
                                      AudioBufferList* io_data,
                                      const AudioTimeStamp* output_time_stamp) {
-  static const bool kDisableAudioOutputResampler =
-      CommandLine::ForCurrentProcess()->HasSwitch(
-          switches::kDisableAudioOutputResampler);
-
   // Update the playout latency.
   double playout_latency_frames = GetPlayoutLatency(output_time_stamp);
 
@@ -234,18 +235,23 @@ OSStatus AUAudioOutputStream::Render(UInt32 number_of_frames,
   uint32 hardware_pending_bytes = static_cast<uint32>
       ((playout_latency_frames + 0.5) * format_.mBytesPerFrame);
 
-  // If we specify a buffer size which is too low, the OS will ask for more data
-  // to fulfill the hardware request, so resize the AudioBus as appropriate.
-  // This change requires AudioOutputResampler to prevent buffer size mismatches
-  // downstream, so glitch if it's not enabled.
-  if (!kDisableAudioOutputResampler &&
-      static_cast<UInt32>(audio_bus_->frames()) != number_of_frames) {
-    audio_bus_ = AudioBus::Create(audio_bus_->channels(), number_of_frames);
+  // Unfortunately AUAudioInputStream and AUAudioOutputStream share the frame
+  // size set by kAudioDevicePropertyBufferFrameSize above on a per process
+  // basis.  What this means is that the |number_of_frames| value may be larger
+  // or smaller than the value set during Configure().  In this case either
+  // audio input or audio output will be broken, so just output silence.
+  // TODO(henrika): This should never happen so long as we're always using the
+  // hardware sample rate and the input/output streams configure the same frame
+  // size.  This is currently not true.  See http://crbug.com/154352.  Once
+  // fixed, a CHECK() should be added and this wall of text removed.
+  if (number_of_frames != static_cast<UInt32>(audio_bus_->frames())) {
+    memset(audio_data, 0, number_of_frames * format_.mBytesPerFrame);
+    return noErr;
   }
 
-  int frames_filled = std::min(source_->OnMoreData(
-      audio_bus_.get(), AudioBuffersState(0, hardware_pending_bytes)),
-      static_cast<int>(number_of_frames));
+  int frames_filled = source_->OnMoreData(
+      audio_bus_.get(), AudioBuffersState(0, hardware_pending_bytes));
+
   // Note: If this ever changes to output raw float the data must be clipped and
   // sanitized since it may come from an untrusted source such as NaCl.
   audio_bus_->ToInterleaved(
diff --git a/media/base/audio_bus.cc b/media/base/audio_bus.cc
index 9ff6ccd..677a0c7 100644
--- a/media/base/audio_bus.cc
+++ b/media/base/audio_bus.cc
@@ -290,8 +290,8 @@ void AudioBus::ToInterleaved(int frames, int bytes_per_sample,
 }
 
 void AudioBus::CopyTo(AudioBus* dest) const {
-  DCHECK_EQ(channels(), dest->channels());
-  DCHECK_EQ(frames(), dest->frames());
+  CHECK_EQ(channels(), dest->channels());
+  CHECK_EQ(frames(), dest->frames());
 
   // Since we don't know if the other AudioBus is wrapped or not (and we don't
   // want to care), just copy using the public channel() accessors.