Adding input and output delay estimation for mac.

media/audio/mac/audio_low_latency_output_mac.cc

 // Copyright (c) 2011 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/audio/mac/audio_low_latency_output_mac.h"
 
 #include <CoreServices/CoreServices.h>
 
 #include "base/basictypes.h"
 #include "base/logging.h"
@@ skipping 29 matching lines @@
 // 4) At some point some thread will call Stop(), which we handle by directly
 // stopping the default output Audio Unit.
 // 6) The same thread that called stop will call Close() where we cleanup
 // and notify the audio manager, which likely will destroy this object.
 
 AUAudioOutputStream::AUAudioOutputStream(
     AudioManagerMac* manager, const AudioParameters& params)
     : manager_(manager),
       source_(NULL),
       output_unit_(0),
-      volume_(1) {
+      output_device_id_(kAudioDeviceUnknown),
+      volume_(1),
+      hardware_latency_frames_(0) {
   // We must have a manager.
   DCHECK(manager_);
   // A frame is one sample across all channels. In interleaved audio the per
   // frame fields identify the set of n |channels|. In uncompressed audio, a
   // packet is always one frame.
   format_.mSampleRate = params.sample_rate;
   format_.mFormatID = kAudioFormatLinearPCM;
   format_.mFormatFlags = kLinearPCMFormatFlagIsPacked |
                          kLinearPCMFormatFlagIsSignedInteger;
   format_.mBitsPerChannel = params.bits_per_sample;
   format_.mChannelsPerFrame = params.channels;
   format_.mFramesPerPacket = 1;
   format_.mBytesPerPacket = (format_.mBitsPerChannel * params.channels) / 8;
   format_.mBytesPerFrame = format_.mBytesPerPacket;
   format_.mReserved = 0;
 
   // Calculate the number of sample frames per callback.
   number_of_frames_ = params.GetPacketSize() / format_.mBytesPerPacket;
 }
 
 AUAudioOutputStream::~AUAudioOutputStream() {
 }
 
 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;
 
   desc.componentType = kAudioUnitType_Output;
   desc.componentSubType = kAudioUnitSubType_DefaultOutput;
   desc.componentManufacturer = kAudioUnitManufacturer_Apple;
   desc.componentFlags = 0;
   desc.componentFlagsMask = 0;
   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;
 
   result = AudioUnitInitialize(output_unit_);
 
   DCHECK_EQ(result, 0);
   if (result)
     return false;
 
+  // Gets the playout device hardware latency and stores the value.
+  hardware_latency_frames_ = GetHardwareLatency();
+
   return Configure();
 }
 
 bool AUAudioOutputStream::Configure() {
   // Set the render callback.
   AURenderCallbackStruct input;
   input.inputProc = InputProc;
   input.inputProcRefCon = this;
   OSStatus result = AudioUnitSetProperty(
       output_unit_,
@@ skipping 70 matching lines @@
 void AUAudioOutputStream::GetVolume(double* volume) {
   if (!output_unit_)
     return;
   *volume = volume_;
 }
 
 // Pulls on our provider to get rendered audio stream.
 // 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.
+  double playout_latency_frames = GetPlayoutLatency(output_time_stamp);
+
   AudioBuffer& buffer = io_data->mBuffers[0];
   uint8* audio_data = reinterpret_cast<uint8*>(buffer.mData);
+  uint32 hardware_pending_bytes = static_cast<uint32>
+      (playout_latency_frames * format_.mBytesPerFrame + 0.5);
   uint32 filled = source_->OnMoreData(
-      this, audio_data, buffer.mDataByteSize, AudioBuffersState(0, 0));
+      this, audio_data, buffer.mDataByteSize,
+      AudioBuffersState(0, hardware_pending_bytes));
 
   // Handle channel order for 5.1 audio.
   if (format_.mChannelsPerFrame == 6) {
     if (format_.mBitsPerChannel == 8) {
       SwizzleCoreAudioLayout5_1(reinterpret_cast<uint8*>(audio_data), filled);
     } else if (format_.mBitsPerChannel == 16) {
       SwizzleCoreAudioLayout5_1(reinterpret_cast<int16*>(audio_data), filled);
     } else if (format_.mBitsPerChannel == 32) {
       SwizzleCoreAudioLayout5_1(reinterpret_cast<int32*>(audio_data), filled);
     }
   }
 
   return noErr;
 }
 
 // DefaultOutputUnit callback
 OSStatus AUAudioOutputStream::InputProc(void* user_data,
                                         AudioUnitRenderActionFlags*,
-                                        const AudioTimeStamp*,
+                                        const AudioTimeStamp* output_time_stamp,
                                         UInt32,
                                         UInt32 number_of_frames,
                                         AudioBufferList* io_data) {
   AUAudioOutputStream* audio_output =
       static_cast<AUAudioOutputStream*>(user_data);
   DCHECK(audio_output);
   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() {
   // Determine the default output device's sample-rate.
   AudioDeviceID device_id = kAudioDeviceUnknown;
   UInt32 info_size = sizeof(device_id);
 
   AudioObjectPropertyAddress default_output_device_address = {
       kAudioHardwarePropertyDefaultOutputDevice,
       kAudioObjectPropertyScopeGlobal,
@@ skipping 22 matching lines @@
                                       0,
                                       0,
                                       &info_size,
                                       &nominal_sample_rate);
   DCHECK_EQ(result, 0);
   if (result)
     return 0.0;  // error
 
   return nominal_sample_rate;
 }
+
+double AUAudioOutputStream::GetHardwareLatency() {
+  // Get audio unit latency.
+  Float64 audio_unit_latency_sec = 0.0;
+  UInt32 size = sizeof(audio_unit_latency_sec);
+  OSStatus result = AudioUnitGetProperty(output_unit_,
+                                         kAudioUnitProperty_Latency,
+                                         kAudioUnitScope_Global,
+                                         0,
+                                         &audio_unit_latency_sec,
+                                         &size);
+  if (result)

[scherkus (not reviewing), 2011/10/19 18:23:57]

ditto on using {} and returning 0

[no longer working on chromium, 2011/10/20 11:25:51]

Done.

+    DLOG(WARNING) << "GetHardwareLatency: Could not get audio unit latency.";
+
+  // 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) << "GetHardwareLatency: Could not get device latency.";
+
+  // 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)
+      result = AudioStreamGetProperty(stream_ids[0],
+                                      0,
+                                      kAudioStreamPropertyLatency,
+                                      &size,
+                                      &stream_latency_frames);
+  }
+  // Log the warning if it fails to get stream latency.
+  if (result)
+    DLOG(WARNING) << "GetHardwareLatency: Could not get stream latency.";
+
+  // Returns the hardware latency value in freams.
+  return static_cast<double>(audio_unit_latency_sec *
+      format_.mSampleRate + device_latency_frames + stream_latency_frames);
+}
+
+double AUAudioOutputStream::GetPlayoutLatency(
+    const AudioTimeStamp* output_time_stamp) {
+  // Get the delay between now and when the data is going to reach the hardware.
+  UInt64 output_time_ns = AudioConvertHostTimeToNanos(
+      output_time_stamp->mHostTime);
+  UInt64 now_ns = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+  double delay_frames = static_cast<double>
+      (1e-9 * (output_time_ns - now_ns) * format_.mSampleRate);
+
+  return (delay_frames + hardware_latency_frames_);
+}