Reland 501823002: Used native deinterleaved and float point format for the input streams

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 f1dbdf786fb2910b21c043dddad598cb98b54b30..7819dd7827d2d12fdf333d72b8baed15c57a38f2 100644
--- a/media/audio/mac/audio_low_latency_input_mac.cc
+++ b/media/audio/mac/audio_low_latency_input_mac.cc
@@ -10,6 +10,7 @@
 #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/data_buffer.h"
 
@@ -31,6 +32,23 @@ static std::ostream& operator<<(std::ostream& os,
   return os;
 }
 
+static void WrapBufferList(AudioBufferList* buffer_list,
+                           AudioBus* bus,
+                           int frames) {
+  DCHECK(buffer_list);
+  DCHECK(bus);
+  const int channels = bus->channels();
+  const int buffer_list_channels = buffer_list->mNumberBuffers;
+  CHECK_EQ(channels, buffer_list_channels);
+
+  // Copy pointers from AudioBufferList.
+  for (int i = 0; i < channels; ++i)
+    bus->SetChannelData(i, static_cast<float*>(buffer_list->mBuffers[i].mData));
+
+  // Finally set the actual length.
+  bus->set_frames(frames);
+}
+
 // See "Technical Note TN2091 - Device input using the HAL Output Audio Unit"
 // http://developer.apple.com/library/mac/#technotes/tn2091/_index.html
 // for more details and background regarding this implementation.
@@ -46,43 +64,47 @@ AUAudioInputStream::AUAudioInputStream(AudioManagerMac* manager,
       started_(false),
       hardware_latency_frames_(0),
       number_of_channels_in_frame_(0),
-      fifo_(input_params.channels(),
-            number_of_frames_,
-            kNumberOfBlocksBufferInFifo) {
+      audio_wrapper_(AudioBus::CreateWrapper(input_params.channels())) {
   DCHECK(manager_);
 
   // Set up the desired (output) format specified by the client.
   format_.mSampleRate = input_params.sample_rate();
   format_.mFormatID = kAudioFormatLinearPCM;
-  format_.mFormatFlags = kLinearPCMFormatFlagIsPacked |
-                         kLinearPCMFormatFlagIsSignedInteger;
-  format_.mBitsPerChannel = input_params.bits_per_sample();
+  format_.mFormatFlags =
+      kAudioFormatFlagsNativeFloatPacked | kLinearPCMFormatFlagIsNonInterleaved;
+  size_t bytes_per_sample = sizeof(Float32);
+  format_.mBitsPerChannel = bytes_per_sample * 8;
   format_.mChannelsPerFrame = input_params.channels();
-  format_.mFramesPerPacket = 1;  // uncompressed audio
-  format_.mBytesPerPacket = (format_.mBitsPerChannel *
-                             input_params.channels()) / 8;
-  format_.mBytesPerFrame = format_.mBytesPerPacket;
+  format_.mFramesPerPacket = 1;
+  format_.mBytesPerFrame = bytes_per_sample;
+  format_.mBytesPerPacket = format_.mBytesPerFrame * format_.mFramesPerPacket;
   format_.mReserved = 0;
 
   DVLOG(1) << "Desired ouput format: " << format_;
 
-  // Derive size (in bytes) of the buffers that we will render to.
-  UInt32 data_byte_size = number_of_frames_ * format_.mBytesPerFrame;
-  DVLOG(1) << "Size of data buffer in bytes : " << data_byte_size;
+  // Allocate AudioBufferList based on the number of channels.
+  audio_buffer_list_.reset(static_cast<AudioBufferList*>(
+      malloc(sizeof(AudioBufferList) * input_params.channels())));

[DaleCurtis, 2014/08/27 20:22:52]

This will oversize the buffer slightly, but should be fine. Per an Apple email
here's the recommended way:

http://lists.apple.com/archives/coreaudio-api/2005/Nov/msg00252.html

[no longer working on chromium, 2014/08/28 16:12:20]

Thanks, I read this, and decided to go for the existing way because I was not
sure if offsetof() is available.

+  audio_buffer_list_->mNumberBuffers = input_params.channels();
 
   // Allocate AudioBuffers to be used as storage for the received audio.
   // The AudioBufferList structure works as a placeholder for the
   // AudioBuffer structure, which holds a pointer to the actual data buffer.
-  audio_data_buffer_.reset(new uint8[data_byte_size]);
-  audio_buffer_list_.mNumberBuffers = 1;
-
-  AudioBuffer* audio_buffer = audio_buffer_list_.mBuffers;
-  audio_buffer->mNumberChannels = input_params.channels();
-  audio_buffer->mDataByteSize = data_byte_size;
-  audio_buffer->mData = audio_data_buffer_.get();
+  UInt32 data_byte_size = number_of_frames_ * format_.mBytesPerFrame;
+  audio_data_buffer_.reset(static_cast<uint8*>(base::AlignedAlloc(
+      data_byte_size, AudioBus::kChannelAlignment)));
+  CHECK_LE(static_cast<int>(data_byte_size * input_params.channels()),
+           media::AudioBus::CalculateMemorySize(input_params));
+  AudioBuffer* audio_buffer = audio_buffer_list_->mBuffers;
+  for (UInt32 i = 0; i < audio_buffer_list_->mNumberBuffers; ++i) {
+    audio_buffer[i].mNumberChannels = 1;
+    audio_buffer[i].mDataByteSize = data_byte_size;
+    audio_buffer[i].mData = audio_data_buffer_.get() + i * data_byte_size;
+  }
 }
 
-AUAudioInputStream::~AUAudioInputStream() {}
+AUAudioInputStream::~AUAudioInputStream() {
+}
 
 // Obtain and open the AUHAL AudioOutputUnit for recording.
 bool AUAudioInputStream::Open() {
@@ -165,23 +187,6 @@ bool AUAudioInputStream::Open() {
     return false;
   }
 
-  // Register the input procedure for the AUHAL.
-  // This procedure will be called when the AUHAL has received new data
-  // from the input device.
-  AURenderCallbackStruct callback;
-  callback.inputProc = InputProc;
-  callback.inputProcRefCon = this;
-  result = AudioUnitSetProperty(audio_unit_,
-                                kAudioOutputUnitProperty_SetInputCallback,
-                                kAudioUnitScope_Global,
-                                0,
-                                &callback,
-                                sizeof(callback));
-  if (result) {
-    HandleError(result);
-    return false;
-  }
-
   // Set up the the desired (output) format.
   // For obtaining input from a device, the device format is always expressed
   // on the output scope of the AUHAL's Element 1.
@@ -229,6 +234,23 @@ bool AUAudioInputStream::Open() {
     }
   }
 
+  // Register the input procedure for the AUHAL.
+  // This procedure will be called when the AUHAL has received new data
+  // from the input device.
+  AURenderCallbackStruct callback;
+  callback.inputProc = InputProc;
+  callback.inputProcRefCon = this;
+  result = AudioUnitSetProperty(audio_unit_,
+                                kAudioOutputUnitProperty_SetInputCallback,
+                                kAudioUnitScope_Global,
+                                0,
+                                &callback,
+                                sizeof(callback));
+  if (result) {
+    HandleError(result);
+    return false;
+  }
+
   // Finally, initialize the audio unit and ensure that it is ready to render.
   // Allocates memory according to the maximum number of audio frames
   // it can produce in response to a single render call.
@@ -342,9 +364,9 @@ void AUAudioInputStream::SetVolume(double volume) {
 
   Float32 volume_float32 = static_cast<Float32>(volume);
   AudioObjectPropertyAddress property_address = {
-    kAudioDevicePropertyVolumeScalar,
-    kAudioDevicePropertyScopeInput,
-    kAudioObjectPropertyElementMaster
+      kAudioDevicePropertyVolumeScalar,
+      kAudioDevicePropertyScopeInput,
+      kAudioObjectPropertyElementMaster
   };
 
   // Try to set the volume for master volume channel.
@@ -390,15 +412,15 @@ void AUAudioInputStream::SetVolume(double volume) {
 
 double AUAudioInputStream::GetVolume() {
   // Verify that we have a valid device.
-  if (input_device_id_ == kAudioObjectUnknown){
+  if (input_device_id_ == kAudioObjectUnknown) {
     NOTREACHED() << "Device ID is unknown";
     return 0.0;
   }
 
   AudioObjectPropertyAddress property_address = {
-    kAudioDevicePropertyVolumeScalar,
-    kAudioDevicePropertyScopeInput,
-    kAudioObjectPropertyElementMaster
+      kAudioDevicePropertyVolumeScalar,
+      kAudioDevicePropertyScopeInput,
+      kAudioObjectPropertyElementMaster
   };
 
   if (AudioObjectHasProperty(input_device_id_, &property_address)) {
@@ -406,12 +428,8 @@ double AUAudioInputStream::GetVolume() {
     // master channel.
     Float32 volume_float32 = 0.0;
     UInt32 size = sizeof(volume_float32);
-    OSStatus result = AudioObjectGetPropertyData(input_device_id_,
-                                                 &property_address,
-                                                 0,
-                                                 NULL,
-                                                 &size,
-                                                 &volume_float32);
+    OSStatus result = AudioObjectGetPropertyData(
+        input_device_id_, &property_address, 0, NULL, &size, &volume_float32);
     if (result == noErr)
       return static_cast<double>(volume_float32);
   } else {
@@ -472,9 +490,8 @@ OSStatus AUAudioInputStream::InputProc(void* user_data,
     return result;
 
   // Deliver recorded data to the consumer as a callback.
-  return audio_input->Provide(number_of_frames,
-                              audio_input->audio_buffer_list(),
-                              time_stamp);
+  return audio_input->Provide(
+      number_of_frames, audio_input->audio_buffer_list(), time_stamp);
 }
 
 OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
@@ -491,22 +508,42 @@ OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
 
   AudioBuffer& buffer = io_data->mBuffers[0];
   uint8* audio_data = reinterpret_cast<uint8*>(buffer.mData);
-  uint32 capture_delay_bytes = static_cast<uint32>
-      ((capture_latency_frames + 0.5) * format_.mBytesPerFrame);
+  uint32 capture_delay_bytes = static_cast<uint32>(
+      (capture_latency_frames + 0.5) * format_.mBytesPerFrame);
   DCHECK(audio_data);
   if (!audio_data)
     return kAudioUnitErr_InvalidElement;
 
-  // Copy captured (and interleaved) data into FIFO.
-  fifo_.Push(audio_data, number_of_frames, format_.mBitsPerChannel / 8);
+  // Wrap the output AudioBufferList to |audio_wrapper_|.
+  WrapBufferList(io_data, audio_wrapper_.get(), number_of_frames);
+
+  // If the stream parameters change for any reason, we need to insert a FIFO
+  // since the OnMoreData() pipeline can't handle frame size changes.
+  if (number_of_frames != number_of_frames_) {
+    // Create a FIFO on the fly to handle any discrepancies in callback rates.
+    if (!fifo_) {
+      fifo_.reset(new AudioBlockFifo(audio_wrapper_->channels(),
+                                     number_of_frames_,
+                                     kNumberOfBlocksBufferInFifo));
+    }
+  }
 
+  // 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;
+  }
+
+  // Compensate the audio delay caused by the FIFO.
+  capture_delay_bytes += fifo_->GetAvailableFrames() * format_.mBytesPerFrame;
+
+  fifo_->Push(audio_wrapper_.get());
   // Consume and deliver the data when the FIFO has a block of available data.
-  while (fifo_.available_blocks()) {
-    const AudioBus* audio_bus = fifo_.Consume();
+  while (fifo_->available_blocks()) {
+    const AudioBus* audio_bus = fifo_->Consume();
     DCHECK_EQ(audio_bus->frames(), static_cast<int>(number_of_frames_));
-
-    // Compensate the audio delay caused by the FIFO.
-    capture_delay_bytes += fifo_.GetAvailableFrames() * format_.mBytesPerFrame;
     sink_->OnData(this, audio_bus, capture_delay_bytes, normalized_volume);
   }
 
@@ -519,9 +556,9 @@ int AUAudioInputStream::HardwareSampleRate() {
   UInt32 info_size = sizeof(device_id);
 
   AudioObjectPropertyAddress default_input_device_address = {
-    kAudioHardwarePropertyDefaultInputDevice,
-    kAudioObjectPropertyScopeGlobal,
-    kAudioObjectPropertyElementMaster
+      kAudioHardwarePropertyDefaultInputDevice,
+      kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster
   };
   OSStatus result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
                                                &default_input_device_address,
@@ -536,10 +573,8 @@ int AUAudioInputStream::HardwareSampleRate() {
   info_size = sizeof(nominal_sample_rate);
 
   AudioObjectPropertyAddress nominal_sample_rate_address = {
-    kAudioDevicePropertyNominalSampleRate,
-    kAudioObjectPropertyScopeGlobal,
-    kAudioObjectPropertyElementMaster
-  };
+      kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
   result = AudioObjectGetPropertyData(device_id,
                                       &nominal_sample_rate_address,
                                       0,
@@ -572,9 +607,9 @@ double AUAudioInputStream::GetHardwareLatency() {
 
   // Get input audio device latency.
   AudioObjectPropertyAddress property_address = {
-    kAudioDevicePropertyLatency,
-    kAudioDevicePropertyScopeInput,
-    kAudioObjectPropertyElementMaster
+      kAudioDevicePropertyLatency,
+      kAudioDevicePropertyScopeInput,
+      kAudioObjectPropertyElementMaster
   };
   UInt32 device_latency_frames = 0;
   size = sizeof(device_latency_frames);
@@ -586,19 +621,19 @@ double AUAudioInputStream::GetHardwareLatency() {
                                       &device_latency_frames);
   DLOG_IF(WARNING, result != noErr) << "Could not get audio device latency.";
 
-  return static_cast<double>((audio_unit_latency_sec *
-      format_.mSampleRate) + device_latency_frames);
+  return static_cast<double>((audio_unit_latency_sec * format_.mSampleRate) +
+                             device_latency_frames);
 }
 
 double AUAudioInputStream::GetCaptureLatency(
     const AudioTimeStamp* input_time_stamp) {
   // Get the delay between between the actual recording instant and the time
   // when the data packet is provided as a callback.
-  UInt64 capture_time_ns = AudioConvertHostTimeToNanos(
-      input_time_stamp->mHostTime);
+  UInt64 capture_time_ns =
+      AudioConvertHostTimeToNanos(input_time_stamp->mHostTime);
   UInt64 now_ns = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
-  double delay_frames = static_cast<double>
-      (1e-9 * (now_ns - capture_time_ns) * format_.mSampleRate);
+  double delay_frames = static_cast<double>(1e-9 * (now_ns - capture_time_ns) *
+                                            format_.mSampleRate);
 
   // Total latency is composed by the dynamic latency and the fixed
   // hardware latency.
@@ -608,18 +643,14 @@ double AUAudioInputStream::GetCaptureLatency(
 int AUAudioInputStream::GetNumberOfChannelsFromStream() {
   // Get the stream format, to be able to read the number of channels.
   AudioObjectPropertyAddress property_address = {
-    kAudioDevicePropertyStreamFormat,
-    kAudioDevicePropertyScopeInput,
-    kAudioObjectPropertyElementMaster
+      kAudioDevicePropertyStreamFormat,
+      kAudioDevicePropertyScopeInput,
+      kAudioObjectPropertyElementMaster
   };
   AudioStreamBasicDescription stream_format;
   UInt32 size = sizeof(stream_format);
-  OSStatus result = AudioObjectGetPropertyData(input_device_id_,
-                                               &property_address,
-                                               0,
-                                               NULL,
-                                               &size,
-                                               &stream_format);
+  OSStatus result = AudioObjectGetPropertyData(
+      input_device_id_, &property_address, 0, NULL, &size, &stream_format);
   if (result != noErr) {
     DLOG(WARNING) << "Could not get stream format";
     return 0;
@@ -629,8 +660,8 @@ int AUAudioInputStream::GetNumberOfChannelsFromStream() {
 }
 
 void AUAudioInputStream::HandleError(OSStatus err) {
-  NOTREACHED() << "error " << GetMacOSStatusErrorString(err)
-               << " (" << err << ")";
+  NOTREACHED() << "error " << GetMacOSStatusErrorString(err) << " (" << err
+               << ")";
   if (sink_)
     sink_->OnError(this);
 }
@@ -638,13 +669,12 @@ void AUAudioInputStream::HandleError(OSStatus err) {
 bool AUAudioInputStream::IsVolumeSettableOnChannel(int channel) {
   Boolean is_settable = false;
   AudioObjectPropertyAddress property_address = {
-    kAudioDevicePropertyVolumeScalar,
-    kAudioDevicePropertyScopeInput,
-    static_cast<UInt32>(channel)
+      kAudioDevicePropertyVolumeScalar,
+      kAudioDevicePropertyScopeInput,
+      static_cast<UInt32>(channel)
   };
-  OSStatus result = AudioObjectIsPropertySettable(input_device_id_,
-                                                  &property_address,
-                                                  &is_settable);
+  OSStatus result = AudioObjectIsPropertySettable(
+      input_device_id_, &property_address, &is_settable);
   return (result == noErr) ? is_settable : false;
 }