Switch the input code to use AudioBlockFifo

media/audio/mac/audio_low_latency_input_mac.cc

Index: media/audio/mac/audio_low_latency_input_mac.cc
diff --git a/media/audio/mac/audio_low_latency_input_mac.cc b/media/audio/mac/audio_low_latency_input_mac.cc
index f0f6629185fb63fca4ebe97004231f47b18e31b6..19a7530da90e3ad8ded48a751e4aa0a0b448a006 100644
--- a/media/audio/mac/audio_low_latency_input_mac.cc
+++ b/media/audio/mac/audio_low_latency_input_mac.cc
@@ -10,12 +10,15 @@
 #include "base/logging.h"
 #include "base/mac/mac_logging.h"
 #include "media/audio/mac/audio_manager_mac.h"
+#include "media/base/audio_block_fifo.h"
 #include "media/base/audio_bus.h"
-#include "media/base/audio_fifo.h"
 #include "media/base/data_buffer.h"
 
 namespace media {
 
+// Number of blocks of buffers used in the |fifo_|.
+const int kNumberOfBlocksBufferInFifo = 2;
+
 static std::ostream& operator<<(std::ostream& os,
                                 const AudioStreamBasicDescription& format) {
   os << "sample rate       : " << format.mSampleRate << std::endl
@@ -44,8 +47,8 @@ AUAudioInputStream::AUAudioInputStream(AudioManagerMac* manager,
       started_(false),
       hardware_latency_frames_(0),
       number_of_channels_in_frame_(0),
-      audio_bus_(media::AudioBus::Create(input_params)),
-      audio_wrapper_(media::AudioBus::Create(input_params)) {
+      fifo_(new AudioBlockFifo(input_params.channels(), number_of_frames_,

[DaleCurtis, 2014/07/18 18:21:54]

No need for dynamic allocation.

[no longer working on chromium, 2014/07/21 15:13:30]

Done.

+                               kNumberOfBlocksBufferInFifo)) {
   DCHECK(manager_);
 
   // Set up the desired (output) format specified by the client.
@@ -494,43 +497,18 @@ OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
   if (!audio_data)
     return kAudioUnitErr_InvalidElement;
 
-  if (number_of_frames != number_of_frames_) {
-    // Create a FIFO on the fly to handle any discrepancies in callback rates.
-    if (!fifo_) {
-      VLOG(1) << "Audio frame size changed from " << number_of_frames_ << " to "
-              << number_of_frames << "; adding FIFO to compensate.";
-      fifo_.reset(new AudioFifo(
-          format_.mChannelsPerFrame, number_of_frames_ + number_of_frames));
-    }
-
-    if (audio_wrapper_->frames() != static_cast<int>(number_of_frames)) {
-      audio_wrapper_ = media::AudioBus::Create(format_.mChannelsPerFrame,
-                                               number_of_frames);
-    }
-  }
-
-  // Copy captured (and interleaved) data into deinterleaved audio bus.
-  audio_wrapper_->FromInterleaved(
-      audio_data, audio_wrapper_->frames(), format_.mBitsPerChannel / 8);
-
-  // When FIFO does not kick in, data will be directly passed to the callback.
-  if (!fifo_) {
-    CHECK_EQ(audio_wrapper_->frames(), static_cast<int>(number_of_frames_));
-    sink_->OnData(
-        this, audio_wrapper_.get(), capture_delay_bytes, normalized_volume);
-    return noErr;
-  }
+  // Copy captured (and interleaved) data into FIFO.
+  fifo_->Push(audio_data, number_of_frames, format_.mBitsPerChannel / 8);
 
-  // Compensate the audio delay caused by the FIFO.
-  capture_delay_bytes += fifo_->frames() * format_.mBytesPerFrame;
-  fifo_->Push(audio_wrapper_.get());
-  if (fifo_->frames() >= static_cast<int>(number_of_frames_)) {
-    // Consume the audio from the FIFO.
-    fifo_->Consume(audio_bus_.get(), 0, audio_bus_->frames());
-    DCHECK(fifo_->frames() < static_cast<int>(number_of_frames_));
+  // Consume and deliver the data when the FIFO has a block of available data.
+  while (fifo_->available_blocks()) {
+    const AudioBus* audio_bus = fifo_->Consume();
+    CHECK_EQ(audio_bus->frames(), static_cast<int>(number_of_frames_));

[DaleCurtis, 2014/07/18 18:21:54]

DCHECK? shouldn't ever be wrong since you're no longer reallocating.

[no longer working on chromium, 2014/07/21 15:13:30]

Done. DCHECK is helping to make sure we deliver what the clients request.

 
+    // Compensate the audio delay caused by the FIFO.
+    capture_delay_bytes += fifo_->available_frames() * format_.mBytesPerFrame;
     sink_->OnData(
-        this, audio_bus_.get(), capture_delay_bytes, normalized_volume);
+        this, audio_bus, capture_delay_bytes, normalized_volume);
   }
 
   return noErr;