Adding input and output delay estimation for mac.

media/audio/mac/audio_low_latency_output_mac.cc

Index: media/audio/mac/audio_low_latency_output_mac.cc
===================================================================
--- media/audio/mac/audio_low_latency_output_mac.cc	(revision 104447)
+++ media/audio/mac/audio_low_latency_output_mac.cc	(working copy)
@@ -47,7 +47,10 @@
     : manager_(manager),
       source_(NULL),
       output_unit_(0),
-      volume_(1) {
+      output_device_id_(kAudioDeviceUnknown),
+      volume_(1),
+      hardware_latency_ms_(0),
+      playout_latency_ms_(0) {
   // We must have a manager.
   DCHECK(manager_);
   // A frame is one sample across all channels. In interleaved audio the per
@@ -72,6 +75,16 @@
 }
 
 bool AUAudioOutputStream::Open() {
+  // Obtain the current input device selected by the user.
+  UInt32 size = sizeof(output_device_id_);
+  OSStatus result = AudioHardwareGetProperty(
+      kAudioHardwarePropertyDefaultOutputDevice,
+      &size,
+      &output_device_id_);
+  DCHECK_EQ(result, 0);
+  if (result)
+    return false;
+
   // Open and initialize the DefaultOutputUnit.
   Component comp;
   ComponentDescription desc;
@@ -84,7 +97,7 @@
   comp = FindNextComponent(0, &desc);
   DCHECK(comp);
 
-  OSStatus result = OpenAComponent(comp, &output_unit_);
+  result = OpenAComponent(comp, &output_unit_);
   DCHECK_EQ(result, 0);
   if (result)
     return false;
@@ -95,6 +108,9 @@
   if (result)
     return false;
 
+  // Update the playout device hardware latency.
+  UpdateHardwareLatency();
+
   return Configure();
 }
 
@@ -185,11 +201,18 @@
 // Note to future hackers of this function: Do not add locks here because this
 // is running on a real-time thread (for low-latency).
 OSStatus AUAudioOutputStream::Render(UInt32 number_of_frames,
-                                     AudioBufferList* io_data) {
+                                     AudioBufferList* io_data,
+                                     const AudioTimeStamp* output_time_stamp) {
+  // Update the playout latency.
+  UpdatePlayoutLatency(output_time_stamp);
+
   AudioBuffer& buffer = io_data->mBuffers[0];
   uint8* audio_data = reinterpret_cast<uint8*>(buffer.mData);
+  uint32 hardware_pending_bytes = (1.0e-3 * playout_latency_ms_ *
+      format_.mSampleRate * format_.mBytesPerFrame);
   uint32 filled = source_->OnMoreData(
-      this, audio_data, buffer.mDataByteSize, AudioBuffersState(0, 0));
+      this, audio_data, buffer.mDataByteSize,
+      AudioBuffersState(buffer.mDataByteSize, hardware_pending_bytes));
 
   // Handle channel order for 5.1 audio.
   if (format_.mChannelsPerFrame == 6) {
@@ -208,7 +231,7 @@
 // DefaultOutputUnit callback
 OSStatus AUAudioOutputStream::InputProc(void* user_data,
                                         AudioUnitRenderActionFlags*,
-                                        const AudioTimeStamp*,
+                                        const AudioTimeStamp* output_time_stamp,
                                         UInt32,
                                         UInt32 number_of_frames,
                                         AudioBufferList* io_data) {
@@ -218,7 +241,7 @@
   if (!audio_output)
     return -1;
 
-  return audio_output->Render(number_of_frames, io_data);
+  return audio_output->Render(number_of_frames, io_data, output_time_stamp);
 }
 
 double AUAudioOutputStream::HardwareSampleRate() {
@@ -261,3 +284,69 @@
 
   return nominal_sample_rate;
 }
+
+void AUAudioOutputStream::UpdateHardwareLatency() {
+  // Get audio unit latency.
+  Float64 audio_unit_latency_s = 0;
+  UInt32 size = sizeof(audio_unit_latency_s);
+  OSStatus result = AudioUnitGetProperty(
+      output_unit_, kAudioUnitProperty_Latency, kAudioUnitScope_Global,
+      0, &audio_unit_latency_s, &size);
+  if (result) {
+    DLOG(WARNING) << "UpdateHardwareLatency: Could not get audio unit latency.";
+    audio_unit_latency_s = 0;
+  }
+
+  // Get audio device latency.
+  UInt32 device_latency_frames = 0;
+  size = sizeof(device_latency_frames);
+  result = AudioDeviceGetProperty(output_device_id_, 0, false,
+                                  kAudioDevicePropertyLatency, &size,
+                                  &device_latency_frames);
+  if (result) {
+    DLOG(WARNING) << "UpdateHardwareLatency: Could not get device latency.";
+    device_latency_frames = 0;
+  }
+
+  // Get the stream latency.
+  UInt32 stream_latency_frames = 0;
+  result = AudioDeviceGetPropertyInfo(output_device_id_, 0, false,
+                                      kAudioDevicePropertyStreams, &size, NULL);
+  if (!result) {
+    scoped_ptr_malloc<AudioStreamID>
+        streams(reinterpret_cast<AudioStreamID*>(malloc(size)));
+    AudioStreamID* stream_ids = streams.get();
+    result = AudioDeviceGetProperty(
+        output_device_id_, 0, false,
+        kAudioDevicePropertyStreams, &size, stream_ids);
+    if (result) {
+      DLOG(WARNING) << "UpdateHardwareLatency: Could not get stream id.";
+      return;
+    }
+
+    result = AudioStreamGetProperty(
+        stream_ids[0], 0, kAudioStreamPropertyLatency,
+        &size, &stream_latency_frames);
+    if (result) {
+      DLOG(WARNING) << "UpdateHardwareLatency: Could not get stream latency.";
+      stream_latency_frames = 0;
+    }
+  } else {
+    DLOG(WARNING) << "UpdateHardwareLatency: Could not get stream id.";
+  }
+
+  hardware_latency_ms_ = static_cast<uint32>(1.0e3 * audio_unit_latency_s +
+      (1.0e3 * (device_latency_frames + stream_latency_frames)
+       / format_.mSampleRate) + 0.5);
+}
+
+void AUAudioOutputStream::UpdatePlayoutLatency(
+    const AudioTimeStamp* output_time_stamp) {
+  // Get the delay between now and when the data is going to hit the hardware.
+  UInt64 output_time_ns = AudioConvertHostTimeToNanos(
+      output_time_stamp->mHostTime);
+  UInt64 now_ns = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+  uint32 delay_ms = static_cast<uint32>(1e-6 * (output_time_ns - now_ns) + 0.5);
+
+  playout_latency_ms_ = delay_ms + hardware_latency_ms_;
+}