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-rw-r--r--media/audio/mac/audio_low_latency_input_mac.cc130
-rw-r--r--media/audio/mac/audio_low_latency_input_mac.h21
-rw-r--r--media/audio/mac/audio_low_latency_input_mac_unittest.cc8
-rw-r--r--media/audio/mac/audio_low_latency_output_mac.cc122
-rw-r--r--media/audio/mac/audio_low_latency_output_mac.h31
5 files changed, 287 insertions, 25 deletions
diff --git a/media/audio/mac/audio_low_latency_input_mac.cc b/media/audio/mac/audio_low_latency_input_mac.cc
index 39bac9b..34dbf745 100644
--- a/media/audio/mac/audio_low_latency_input_mac.cc
+++ b/media/audio/mac/audio_low_latency_input_mac.cc
@@ -33,7 +33,9 @@ AUAudioInputStream::AUAudioInputStream(
: manager_(manager),
sink_(NULL),
audio_unit_(0),
- started_(false) {
+ input_device_id_(kAudioObjectUnknown),
+ started_(false),
+ hardware_latency_frames_(0) {
DCHECK(manager_);
// Set up the desired (output) format specified by the client.
@@ -137,25 +139,34 @@ bool AUAudioInputStream::Open() {
// Set the current device of the AudioOuputUnit to default input device.
- AudioDeviceID input_device;
- UInt32 size = sizeof(input_device);
-
// First, obtain the current input device selected by the user.
- result = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice,
- &size,
- &input_device);
+ AudioObjectPropertyAddress default_intput_device_address = {
+ kAudioHardwarePropertyDefaultInputDevice,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster
+ };
+ AudioDeviceID input_device = kAudioObjectUnknown;
+ UInt32 size = sizeof(input_device);
+ result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
+ &default_intput_device_address,
+ 0,
+ 0,
+ &size,
+ &input_device);
if (result) {
HandleError(result);
return false;
}
+ input_device_id_ = input_device;
+
// Next, set the audio device to be the Audio Unit's current device.
// Note that, devices can only be set to the AUHAL after enabling IO.
result = AudioUnitSetProperty(audio_unit_,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global,
0,
- &input_device,
+ &input_device_id_,
sizeof(input_device));
if (result) {
HandleError(result);
@@ -213,6 +224,10 @@ bool AUAudioInputStream::Open() {
HandleError(result);
return false;
}
+
+ // The hardware latency is fixed and will not change during the call.
+ hardware_latency_frames_ = GetHardwareLatency();
+
return true;
}
@@ -290,26 +305,32 @@ OSStatus AUAudioInputStream::InputProc(void* user_data,
// Deliver recorded data to the consumer as a callback.
return audio_input->Provide(number_of_frames,
- audio_input->audio_buffer_list());
+ audio_input->audio_buffer_list(),
+ time_stamp);
}
OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
- AudioBufferList* io_data) {
+ AudioBufferList* io_data,
+ const AudioTimeStamp* time_stamp) {
+ // Update the capture latency.
+ double capture_latency_frames = GetCaptureLatency(time_stamp);
+
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);
DCHECK(audio_data);
if (!audio_data)
return kAudioUnitErr_InvalidElement;
- // TODO(henrika): improve delay estimation. Using buffer size for now.
- sink_->OnData(this, audio_data, buffer.mDataByteSize, buffer.mDataByteSize);
+ sink_->OnData(this, audio_data, buffer.mDataByteSize, capture_delay_bytes);
return noErr;
}
double AUAudioInputStream::HardwareSampleRate() {
// Determine the default input device's sample-rate.
- AudioDeviceID device_id = kAudioDeviceUnknown;
+ AudioDeviceID device_id = kAudioObjectUnknown;
UInt32 info_size = sizeof(device_id);
AudioObjectPropertyAddress default_input_device_address = {
@@ -348,6 +369,89 @@ double AUAudioInputStream::HardwareSampleRate() {
return nominal_sample_rate;
}
+double AUAudioInputStream::GetHardwareLatency() {
+ if (!audio_unit_ || input_device_id_ == kAudioObjectUnknown) {
+ DLOG(WARNING) << "Audio unit object is NULL or device ID is unknown";
+ return 0.0;
+ }
+
+ // Get audio unit latency.
+ Float64 audio_unit_latency_sec = 0.0;
+ UInt32 size = sizeof(audio_unit_latency_sec);
+ OSStatus result = AudioUnitGetProperty(audio_unit_,
+ kAudioUnitProperty_Latency,
+ kAudioUnitScope_Global,
+ 0,
+ &audio_unit_latency_sec,
+ &size);
+ DLOG_IF(WARNING, result != noErr) << "Could not get audio unit latency.";
+
+ // Get input audio device latency.
+ AudioObjectPropertyAddress property_address = {
+ kAudioDevicePropertyLatency,
+ kAudioDevicePropertyScopeInput,
+ kAudioObjectPropertyElementMaster
+ };
+ UInt32 device_latency_frames = 0;
+ size = sizeof(device_latency_frames);
+ result = AudioObjectGetPropertyData(input_device_id_,
+ &property_address,
+ 0,
+ NULL,
+ &size,
+ &device_latency_frames);
+ DLOG_IF(WARNING, result != noErr) << "Could not get audio device latency.";
+
+ // Get the stream latency.
+ property_address.mSelector = kAudioDevicePropertyStreams;
+ UInt32 stream_latency_frames = 0;
+ size = 0;
+ result = AudioObjectGetPropertyDataSize(input_device_id_,
+ &property_address,
+ 0,
+ NULL,
+ &size);
+ if (!result) {
+ scoped_ptr_malloc<AudioStreamID>
+ streams(reinterpret_cast<AudioStreamID*>(malloc(size)));
+ AudioStreamID* stream_ids = streams.get();
+ result = AudioObjectGetPropertyData(input_device_id_,
+ &property_address,
+ 0,
+ NULL,
+ &size,
+ stream_ids);
+ if (!result) {
+ property_address.mSelector = kAudioStreamPropertyLatency;
+ result = AudioObjectGetPropertyData(stream_ids[0],
+ &property_address,
+ 0,
+ NULL,
+ &size,
+ &stream_latency_frames);
+ }
+ }
+ DLOG_IF(WARNING, result != noErr) << "Could not get audio stream latency.";
+
+ return static_cast<double>((audio_unit_latency_sec *
+ format_.mSampleRate) + device_latency_frames + stream_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 now_ns = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+ 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.
+ return (delay_frames + hardware_latency_frames_);
+}
+
void AUAudioInputStream::HandleError(OSStatus err) {
NOTREACHED() << "error code: " << err;
if (sink_)
diff --git a/media/audio/mac/audio_low_latency_input_mac.h b/media/audio/mac/audio_low_latency_input_mac.h
index 8d1811c..f3b2891 100644
--- a/media/audio/mac/audio_low_latency_input_mac.h
+++ b/media/audio/mac/audio_low_latency_input_mac.h
@@ -27,6 +27,11 @@
// input device and then use the same rate when creating this object.
// Use AUAudioInputStream::HardwareSampleRate() to retrieve the sample rate.
// - Calling Close() also leads to self destruction.
+// - The latency consists of two parts:
+// 1) Hardware latency, which includes Audio Unit latency, audio device
+// latency and audio stream latency;
+// 2) The delay between the actual recording instant and the time when the
+// data packet is provided as a callback.
//
#ifndef MEDIA_AUDIO_MAC_AUDIO_LOW_LATENCY_INPUT_MAC_H_
#define MEDIA_AUDIO_MAC_AUDIO_LOW_LATENCY_INPUT_MAC_H_
@@ -73,7 +78,15 @@ class AUAudioInputStream : public AudioInputStream {
AudioBufferList* io_data);
// Pushes recorded data to consumer of the input audio stream.
- OSStatus Provide(UInt32 number_of_frames, AudioBufferList* io_data);
+ OSStatus Provide(UInt32 number_of_frames, AudioBufferList* io_data,
+ const AudioTimeStamp* time_stamp);
+
+ // Gets the fixed capture hardware latency and store it during initialization.
+ // Returns 0 if not available.
+ double GetHardwareLatency();
+
+ // Gets the current capture delay value.
+ double GetCaptureLatency(const AudioTimeStamp* input_time_stamp);
// Issues the OnError() callback to the |sink_|.
void HandleError(OSStatus err);
@@ -96,6 +109,9 @@ class AUAudioInputStream : public AudioInputStream {
// The AUHAL also enables selection of non default devices.
AudioUnit audio_unit_;
+ // The UID refers to the current input audio device.
+ AudioDeviceID input_device_id_;
+
// Provides a mechanism for encapsulating one or more buffers of audio data.
AudioBufferList audio_buffer_list_;
@@ -106,6 +122,9 @@ class AUAudioInputStream : public AudioInputStream {
// True after successfull Start(), false after successful Stop().
bool started_;
+ // Fixed capture hardware latency in frames.
+ double hardware_latency_frames_;
+
DISALLOW_COPY_AND_ASSIGN(AUAudioInputStream);
};
diff --git a/media/audio/mac/audio_low_latency_input_mac_unittest.cc b/media/audio/mac/audio_low_latency_input_mac_unittest.cc
index ea492a4..3d94ad6 100644
--- a/media/audio/mac/audio_low_latency_input_mac_unittest.cc
+++ b/media/audio/mac/audio_low_latency_input_mac_unittest.cc
@@ -13,8 +13,10 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
+using ::testing::_;
using ::testing::AnyNumber;
using ::testing::Between;
+using ::testing::Ge;
using ::testing::NotNull;
class MockAudioInputCallback : public AudioInputStream::AudioInputCallback {
@@ -215,7 +217,8 @@ TEST(MacAudioInputTest, AUAudioInputStreamVerifyMonoRecording) {
// startup sequence takes some time, it is reasonable to expect 5-10
// callbacks in this time period. All should contain valid packets of
// the same size.
- EXPECT_CALL(sink, OnData(ais, NotNull(), bytes_per_packet, bytes_per_packet))
+ EXPECT_CALL(sink, OnData(ais, NotNull(), bytes_per_packet,
+ Ge(bytes_per_packet)))
.Times(Between(5, 10));
ais->Start(&sink);
@@ -248,7 +251,8 @@ TEST(MacAudioInputTest, AUAudioInputStreamVerifyStereoRecording) {
// startup sequence takes some time, it is reasonable to expect 5-10
// callbacks in this time period. All should contain valid packets of
// the same size.
- EXPECT_CALL(sink, OnData(ais, NotNull(), bytes_per_packet, bytes_per_packet))
+ EXPECT_CALL(sink, OnData(ais, NotNull(), bytes_per_packet,
+ Ge(bytes_per_packet)))
.Times(Between(5, 10));
ais->Start(&sink);
diff --git a/media/audio/mac/audio_low_latency_output_mac.cc b/media/audio/mac/audio_low_latency_output_mac.cc
index eb8eb1c..259e240 100644
--- a/media/audio/mac/audio_low_latency_output_mac.cc
+++ b/media/audio/mac/audio_low_latency_output_mac.cc
@@ -47,7 +47,9 @@ AUAudioOutputStream::AUAudioOutputStream(
: manager_(manager),
source_(NULL),
output_unit_(0),
- volume_(1) {
+ output_device_id_(kAudioObjectUnknown),
+ 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
@@ -72,6 +74,23 @@ AUAudioOutputStream::~AUAudioOutputStream() {
}
bool AUAudioOutputStream::Open() {
+ // Obtain the current input device selected by the user.
+ UInt32 size = sizeof(output_device_id_);
+ AudioObjectPropertyAddress default_output_device_address = {
+ kAudioHardwarePropertyDefaultOutputDevice,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster
+ };
+ OSStatus result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
+ &default_output_device_address,
+ 0,
+ 0,
+ &size,
+ &output_device_id_);
+ DCHECK_EQ(result, 0);
+ if (result)
+ return false;
+
// Open and initialize the DefaultOutputUnit.
Component comp;
ComponentDescription desc;
@@ -84,7 +103,7 @@ bool AUAudioOutputStream::Open() {
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 +114,8 @@ bool AUAudioOutputStream::Open() {
if (result)
return false;
+ hardware_latency_frames_ = GetHardwareLatency();
+
return Configure();
}
@@ -185,11 +206,18 @@ void AUAudioOutputStream::GetVolume(double* volume) {
// 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 + 0.5) * format_.mBytesPerFrame);
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) {
@@ -208,7 +236,7 @@ OSStatus AUAudioOutputStream::Render(UInt32 number_of_frames,
// DefaultOutputUnit callback
OSStatus AUAudioOutputStream::InputProc(void* user_data,
AudioUnitRenderActionFlags*,
- const AudioTimeStamp*,
+ const AudioTimeStamp* output_time_stamp,
UInt32,
UInt32 number_of_frames,
AudioBufferList* io_data) {
@@ -218,12 +246,12 @@ OSStatus AUAudioOutputStream::InputProc(void* user_data,
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;
+ AudioDeviceID device_id = kAudioObjectUnknown;
UInt32 info_size = sizeof(device_id);
AudioObjectPropertyAddress default_output_device_address = {
@@ -261,3 +289,83 @@ double AUAudioOutputStream::HardwareSampleRate() {
return nominal_sample_rate;
}
+
+double AUAudioOutputStream::GetHardwareLatency() {
+ if (!output_unit_ || output_device_id_ == kAudioObjectUnknown) {
+ DLOG(WARNING) << "Audio unit object is NULL or device ID is unknown";
+ return 0.0;
+ }
+
+ // 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);
+ DLOG_IF(WARNING, result != noErr) << "Could not get audio unit latency.";
+
+ // Get output audio device latency.
+ AudioObjectPropertyAddress property_address = {
+ kAudioDevicePropertyLatency,
+ kAudioDevicePropertyScopeOutput,
+ kAudioObjectPropertyElementMaster
+ };
+ UInt32 device_latency_frames = 0;
+ size = sizeof(device_latency_frames);
+ result = AudioObjectGetPropertyData(output_device_id_,
+ &property_address,
+ 0,
+ NULL,
+ &size,
+ &device_latency_frames);
+ DLOG_IF(WARNING, result != noErr) << "Could not get audio device latency.";
+
+ // Get the stream latency.
+ property_address.mSelector = kAudioDevicePropertyStreams;
+ UInt32 stream_latency_frames = 0;
+ result = AudioObjectGetPropertyDataSize(output_device_id_,
+ &property_address,
+ 0,
+ NULL,
+ &size);
+ if (!result) {
+ scoped_ptr_malloc<AudioStreamID>
+ streams(reinterpret_cast<AudioStreamID*>(malloc(size)));
+ AudioStreamID* stream_ids = streams.get();
+ result = AudioObjectGetPropertyData(output_device_id_,
+ &property_address,
+ 0,
+ NULL,
+ &size,
+ stream_ids);
+ if (!result) {
+ property_address.mSelector = kAudioStreamPropertyLatency;
+ result = AudioObjectGetPropertyData(stream_ids[0],
+ &property_address,
+ 0,
+ NULL,
+ &size,
+ &stream_latency_frames);
+ }
+ }
+ DLOG_IF(WARNING, result != noErr) << "Could not get audio stream latency.";
+
+ 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 the moment getting the callback and the scheduled
+ // time stamp that tells when the data is going to be played out.
+ 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_);
+}
diff --git a/media/audio/mac/audio_low_latency_output_mac.h b/media/audio/mac/audio_low_latency_output_mac.h
index 724d60b..cab4832 100644
--- a/media/audio/mac/audio_low_latency_output_mac.h
+++ b/media/audio/mac/audio_low_latency_output_mac.h
@@ -1,12 +1,25 @@
// 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.
-
+//
+// Implementation notes:
+//
+// - It is recommended to first acquire the native sample rate of the default
+// output device and then use the same rate when creating this object.
+// Use AUAudioOutputStream::HardwareSampleRate() to retrieve the sample rate.
+// - Calling Close() also leads to self destruction.
+// - The latency consists of two parts:
+// 1) Hardware latency, which includes Audio Unit latency, audio device
+// latency and audio stream latency;
+// 2) The delay between the moment getting the callback and the scheduled time
+// stamp that tells when the data is going to be played out.
+//
#ifndef MEDIA_AUDIO_MAC_AUDIO_LOW_LATENCY_OUTPUT_MAC_H_
#define MEDIA_AUDIO_MAC_AUDIO_LOW_LATENCY_OUTPUT_MAC_H_
#include <AudioUnit/AudioUnit.h>
+#include "base/memory/scoped_ptr.h"
#include "media/audio/audio_io.h"
#include "media/audio/audio_parameters.h"
@@ -44,11 +57,19 @@ class AUAudioOutputStream : public AudioOutputStream {
UInt32 number_of_frames,
AudioBufferList* io_data);
- OSStatus Render(UInt32 number_of_frames, AudioBufferList* io_data);
+ OSStatus Render(UInt32 number_of_frames, AudioBufferList* io_data,
+ const AudioTimeStamp* output_time_stamp);
// Sets up the stream format for the default output Audio Unit.
bool Configure();
+ // Gets the fixed playout device hardware latency and stores it. Returns 0
+ // if not available.
+ double GetHardwareLatency();
+
+ // Gets the current playout latency value.
+ double GetPlayoutLatency(const AudioTimeStamp* output_time_stamp);
+
// Our creator, the audio manager needs to be notified when we close.
AudioManagerMac* manager_;
@@ -63,9 +84,15 @@ class AUAudioOutputStream : public AudioOutputStream {
// The default output Audio Unit which talks to the audio hardware.
AudioUnit output_unit_;
+ // The UID refers to the current output audio device.
+ AudioDeviceID output_device_id_;
+
// Volume level from 0 to 1.
float volume_;
+ // Fixed playout hardware latency in frames.
+ double hardware_latency_frames_;
+
DISALLOW_COPY_AND_ASSIGN(AUAudioOutputStream);
};
generated by cgit v1.1 at 2025-01-31 06:41:52 +0000