| // Copyright (c) 2012 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_manager_mac.h" |
| |
| #include <algorithm> |
| #include <limits> |
| #include <memory> |
| #include <utility> |
| #include <vector> |
| |
| #include "base/bind.h" |
| #include "base/command_line.h" |
| #include "base/containers/flat_set.h" |
| #include "base/mac/mac_logging.h" |
| #include "base/mac/mac_util.h" |
| #include "base/mac/scoped_cftyperef.h" |
| #include "base/macros.h" |
| #include "base/memory/free_deleter.h" |
| #include "base/power_monitor/power_monitor.h" |
| #include "base/power_monitor/power_observer.h" |
| #include "base/strings/sys_string_conversions.h" |
| #include "base/threading/thread_checker.h" |
| #include "media/audio/audio_device_description.h" |
| #include "media/audio/mac/audio_auhal_mac.h" |
| #include "media/audio/mac/audio_input_mac.h" |
| #include "media/audio/mac/audio_low_latency_input_mac.h" |
| #include "media/audio/mac/core_audio_util_mac.h" |
| #include "media/audio/mac/coreaudio_dispatch_override.h" |
| #include "media/audio/mac/scoped_audio_unit.h" |
| #include "media/base/audio_parameters.h" |
| #include "media/base/audio_timestamp_helper.h" |
| #include "media/base/bind_to_current_loop.h" |
| #include "media/base/channel_layout.h" |
| #include "media/base/limits.h" |
| #include "media/base/mac/audio_latency_mac.h" |
| #include "media/base/media_switches.h" |
| #include "third_party/abseil-cpp/absl/types/optional.h" |
| |
| namespace media { |
| |
| // Maximum number of output streams that can be open simultaneously. |
| static const int kMaxOutputStreams = 50; |
| |
| // Default sample-rate on most Apple hardware. |
| static const int kFallbackSampleRate = 44100; |
| |
| static bool GetDeviceChannels(AudioUnit audio_unit, |
| AUElement element, |
| int* channels); |
| |
| // Helper method to construct AudioObjectPropertyAddress structure given |
| // property selector and scope. The property element is always set to |
| // kAudioObjectPropertyElementMaster. |
| static AudioObjectPropertyAddress GetAudioObjectPropertyAddress( |
| AudioObjectPropertySelector selector, |
| bool is_input) { |
| AudioObjectPropertyScope scope = is_input ? kAudioObjectPropertyScopeInput |
| : kAudioObjectPropertyScopeOutput; |
| AudioObjectPropertyAddress property_address = { |
| selector, scope, kAudioObjectPropertyElementMaster}; |
| return property_address; |
| } |
| |
| static const AudioObjectPropertyAddress kNoiseReductionPropertyAddress = { |
| 'nzca', kAudioDevicePropertyScopeInput, kAudioObjectPropertyElementMaster}; |
| |
| // Get IO buffer size range from HAL given device id and scope. |
| static OSStatus GetIOBufferFrameSizeRange(AudioDeviceID device_id, |
| bool is_input, |
| UInt32* minimum, |
| UInt32* maximum) { |
| DCHECK(AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| AudioObjectPropertyAddress address = GetAudioObjectPropertyAddress( |
| kAudioDevicePropertyBufferFrameSizeRange, is_input); |
| AudioValueRange range = {0, 0}; |
| UInt32 data_size = sizeof(AudioValueRange); |
| OSStatus result = AudioObjectGetPropertyData(device_id, &address, 0, NULL, |
| &data_size, &range); |
| if (result != noErr) { |
| OSSTATUS_DLOG(WARNING, result) |
| << "Failed to query IO buffer size range for device: " << std::hex |
| << device_id; |
| } else { |
| *minimum = range.mMinimum; |
| *maximum = range.mMaximum; |
| } |
| return result; |
| } |
| |
| static bool HasAudioHardware(AudioObjectPropertySelector selector) { |
| DCHECK(AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| AudioDeviceID output_device_id = kAudioObjectUnknown; |
| const AudioObjectPropertyAddress property_address = { |
| selector, |
| kAudioObjectPropertyScopeGlobal, // mScope |
| kAudioObjectPropertyElementMaster // mElement |
| }; |
| UInt32 output_device_id_size = static_cast<UInt32>(sizeof(output_device_id)); |
| OSStatus err = AudioObjectGetPropertyData(kAudioObjectSystemObject, |
| &property_address, |
| 0, // inQualifierDataSize |
| NULL, // inQualifierData |
| &output_device_id_size, |
| &output_device_id); |
| return err == kAudioHardwareNoError && |
| output_device_id != kAudioObjectUnknown; |
| } |
| |
| static std::string GetAudioDeviceNameFromDeviceId(AudioDeviceID device_id, |
| bool is_input) { |
| DCHECK(AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| CFStringRef device_name = nullptr; |
| UInt32 data_size = sizeof(device_name); |
| AudioObjectPropertyAddress property_address = GetAudioObjectPropertyAddress( |
| kAudioDevicePropertyDeviceNameCFString, is_input); |
| OSStatus result = AudioObjectGetPropertyData( |
| device_id, &property_address, 0, nullptr, &data_size, &device_name); |
| std::string device; |
| if (result == noErr) { |
| device = base::SysCFStringRefToUTF8(device_name); |
| CFRelease(device_name); |
| } |
| return device; |
| } |
| |
| // Retrieves information on audio devices, and prepends the default |
| // device to the list if the list is non-empty. |
| static void GetAudioDeviceInfo(bool is_input, |
| media::AudioDeviceNames* device_names) { |
| DCHECK(AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| std::vector<AudioObjectID> device_ids = |
| core_audio_mac::GetAllAudioDeviceIDs(); |
| for (AudioObjectID device_id : device_ids) { |
| const bool is_valid_for_direction = |
| (is_input ? core_audio_mac::IsInputDevice(device_id) |
| : core_audio_mac::IsOutputDevice(device_id)); |
| |
| if (!is_valid_for_direction) |
| continue; |
| |
| absl::optional<std::string> unique_id = |
| core_audio_mac::GetDeviceUniqueID(device_id); |
| if (!unique_id) |
| continue; |
| |
| absl::optional<std::string> label = |
| core_audio_mac::GetDeviceLabel(device_id, is_input); |
| if (!label) |
| continue; |
| |
| // Filter out aggregate devices, e.g. those that get created by using |
| // kAudioUnitSubType_VoiceProcessingIO. |
| if (core_audio_mac::IsPrivateAggregateDevice(device_id)) |
| continue; |
| |
| device_names->emplace_back(std::move(*label), std::move(*unique_id)); |
| } |
| |
| if (!device_names->empty()) { |
| // Prepend the default device to the list since we always want it to be |
| // on the top of the list for all platforms. There is no duplicate |
| // counting here since the default device has been abstracted out before. |
| device_names->push_front(media::AudioDeviceName::CreateDefault()); |
| } |
| } |
| |
| AudioDeviceID AudioManagerMac::GetAudioDeviceIdByUId( |
| bool is_input, |
| const std::string& device_id) { |
| DCHECK(AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| AudioObjectPropertyAddress property_address = { |
| kAudioHardwarePropertyDevices, |
| kAudioObjectPropertyScopeGlobal, |
| kAudioObjectPropertyElementMaster |
| }; |
| AudioDeviceID audio_device_id = kAudioObjectUnknown; |
| UInt32 device_size = sizeof(audio_device_id); |
| OSStatus result = -1; |
| |
| if (AudioDeviceDescription::IsDefaultDevice(device_id)) { |
| // Default Device. |
| property_address.mSelector = is_input ? |
| kAudioHardwarePropertyDefaultInputDevice : |
| kAudioHardwarePropertyDefaultOutputDevice; |
| |
| result = AudioObjectGetPropertyData(kAudioObjectSystemObject, |
| &property_address, |
| 0, |
| 0, |
| &device_size, |
| &audio_device_id); |
| } else { |
| // Non-default device. |
| base::ScopedCFTypeRef<CFStringRef> uid( |
| base::SysUTF8ToCFStringRef(device_id)); |
| AudioValueTranslation value; |
| value.mInputData = &uid; |
| value.mInputDataSize = sizeof(CFStringRef); |
| value.mOutputData = &audio_device_id; |
| value.mOutputDataSize = device_size; |
| UInt32 translation_size = sizeof(AudioValueTranslation); |
| |
| property_address.mSelector = kAudioHardwarePropertyDeviceForUID; |
| result = AudioObjectGetPropertyData(kAudioObjectSystemObject, |
| &property_address, |
| 0, |
| 0, |
| &translation_size, |
| &value); |
| } |
| |
| if (result) { |
| OSSTATUS_DLOG(WARNING, result) << "Unable to query device " << device_id |
| << " for AudioDeviceID"; |
| } |
| |
| return audio_device_id; |
| } |
| |
| static bool GetDefaultDevice(AudioDeviceID* device, bool input) { |
| DCHECK(AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| CHECK(device); |
| |
| // Obtain the AudioDeviceID of the default input or output AudioDevice. |
| AudioObjectPropertyAddress pa; |
| pa.mSelector = input ? kAudioHardwarePropertyDefaultInputDevice |
| : kAudioHardwarePropertyDefaultOutputDevice; |
| pa.mScope = kAudioObjectPropertyScopeGlobal; |
| pa.mElement = kAudioObjectPropertyElementMaster; |
| |
| UInt32 size = sizeof(*device); |
| OSStatus result = AudioObjectGetPropertyData(kAudioObjectSystemObject, &pa, 0, |
| 0, &size, device); |
| if ((result != kAudioHardwareNoError) || (*device == kAudioDeviceUnknown)) { |
| DLOG(ERROR) << "Error getting default AudioDevice."; |
| return false; |
| } |
| return true; |
| } |
| |
| bool AudioManagerMac::GetDefaultOutputDevice(AudioDeviceID* device) { |
| return GetDefaultDevice(device, false); |
| } |
| |
| // Returns the total number of channels on a device; regardless of what the |
| // device's preferred rendering layout looks like. Should only be used for the |
| // channel count when a device has more than kMaxConcurrentChannels. |
| static bool GetDeviceTotalChannelCount(AudioDeviceID device, |
| AudioObjectPropertyScope scope, |
| int* channels) { |
| DCHECK(AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| CHECK(channels); |
| |
| // Get the stream configuration of the device in an AudioBufferList (with the |
| // buffer pointers set to nullptr) which describes the list of streams and the |
| // number of channels in each stream. |
| AudioObjectPropertyAddress pa = {kAudioDevicePropertyStreamConfiguration, |
| scope, kAudioObjectPropertyElementMaster}; |
| |
| UInt32 size; |
| OSStatus result = AudioObjectGetPropertyDataSize(device, &pa, 0, 0, &size); |
| if (result != noErr || !size) |
| return false; |
| |
| std::unique_ptr<uint8_t[]> list_storage(new uint8_t[size]); |
| AudioBufferList* buffer_list = |
| reinterpret_cast<AudioBufferList*>(list_storage.get()); |
| |
| result = AudioObjectGetPropertyData(device, &pa, 0, 0, &size, buffer_list); |
| if (result != noErr) |
| return false; |
| |
| // Determine number of channels based on the AudioBufferList. |
| // |mNumberBuffers] is the number of interleaved channels in the buffer. |
| // If the number is 1, the buffer is noninterleaved. |
| *channels = 0; |
| for (UInt32 i = 0; i < buffer_list->mNumberBuffers; ++i) |
| *channels += buffer_list->mBuffers[i].mNumberChannels; |
| |
| DVLOG(1) << (scope == kAudioDevicePropertyScopeInput ? "Input" : "Output") |
| << " total channels: " << *channels; |
| return true; |
| } |
| |
| // Returns the channel count from the |audio_unit|'s stream format for input |
| // scope / input element or output scope / output element. |
| static bool GetAudioUnitStreamFormatChannelCount(AudioUnit audio_unit, |
| AUElement element, |
| int* channels) { |
| AudioStreamBasicDescription stream_format; |
| UInt32 size = sizeof(stream_format); |
| OSStatus result = |
| AudioUnitGetProperty(audio_unit, kAudioUnitProperty_StreamFormat, |
| element == AUElement::OUTPUT ? kAudioUnitScope_Output |
| : kAudioUnitScope_Input, |
| element, &stream_format, &size); |
| if (result != noErr) { |
| OSSTATUS_DLOG(ERROR, result) << "Failed to get AudioUnit stream format."; |
| return false; |
| } |
| |
| *channels = stream_format.mChannelsPerFrame; |
| return true; |
| } |
| |
| // Returns the channel layout for |device| as provided by the AudioUnit attached |
| // to that device matching |element|. Returns true if the count could be pulled |
| // from the AudioUnit successfully, false otherwise. |
| static bool GetDeviceChannels(AudioDeviceID device, |
| AUElement element, |
| int* channels) { |
| DCHECK(AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| CHECK(channels); |
| |
| // For input, get the channel count directly from the AudioUnit's stream |
| // format. |
| // TODO(https://crbug.com/796163): Find out if we can use channel layout on |
| // input element, or confirm that we can't. |
| if (element == AUElement::INPUT) { |
| ScopedAudioUnit au(device, element); |
| if (!au.is_valid()) |
| return false; |
| |
| if (!GetAudioUnitStreamFormatChannelCount(au.audio_unit(), element, |
| channels)) { |
| return false; |
| } |
| |
| DVLOG(1) << "Input channels: " << *channels; |
| return true; |
| } |
| |
| // For output, use the channel layout to determine channel count. |
| DCHECK(element == AUElement::OUTPUT); |
| |
| // If the device has more channels than possible for layouts to express, use |
| // the total count of channels on the device; as of this writing, macOS will |
| // only return up to 8 channels in any layout. To allow WebAudio to work with |
| // > 8 channel devices, we must use the total channel count instead of the |
| // channel count of the preferred layout. |
| int total_channel_count = 0; |
| if (GetDeviceTotalChannelCount(device, |
| element == AUElement::OUTPUT |
| ? kAudioDevicePropertyScopeOutput |
| : kAudioDevicePropertyScopeInput, |
| &total_channel_count) && |
| total_channel_count > kMaxConcurrentChannels) { |
| *channels = total_channel_count; |
| return true; |
| } |
| |
| ScopedAudioUnit au(device, element); |
| if (!au.is_valid()) |
| return false; |
| |
| return GetDeviceChannels(au.audio_unit(), element, channels); |
| } |
| |
| static bool GetDeviceChannels(AudioUnit audio_unit, |
| AUElement element, |
| int* channels) { |
| // Attempt to retrieve the channel layout from the AudioUnit. |
| // |
| // Note: We don't use kAudioDevicePropertyPreferredChannelLayout on the device |
| // because it is not available on all devices. |
| UInt32 size; |
| Boolean writable; |
| OSStatus result = AudioUnitGetPropertyInfo( |
| audio_unit, kAudioUnitProperty_AudioChannelLayout, kAudioUnitScope_Output, |
| element, &size, &writable); |
| if (result != noErr) { |
| OSSTATUS_DLOG(ERROR, result) |
| << "Failed to get property info for AudioUnit channel layout."; |
| } |
| |
| std::unique_ptr<uint8_t[]> layout_storage(new uint8_t[size]); |
| AudioChannelLayout* layout = |
| reinterpret_cast<AudioChannelLayout*>(layout_storage.get()); |
| |
| result = |
| AudioUnitGetProperty(audio_unit, kAudioUnitProperty_AudioChannelLayout, |
| kAudioUnitScope_Output, element, layout, &size); |
| if (result != noErr) { |
| OSSTATUS_LOG(ERROR, result) << "Failed to get AudioUnit channel layout."; |
| return false; |
| } |
| |
| // We don't want to have to know about all channel layout tags, so force OSX |
| // to give us the channel descriptions from the bitmap or tag if necessary. |
| const AudioChannelLayoutTag tag = layout->mChannelLayoutTag; |
| if (tag != kAudioChannelLayoutTag_UseChannelDescriptions) { |
| const bool is_bitmap = tag == kAudioChannelLayoutTag_UseChannelBitmap; |
| const AudioFormatPropertyID fa = |
| is_bitmap ? kAudioFormatProperty_ChannelLayoutForBitmap |
| : kAudioFormatProperty_ChannelLayoutForTag; |
| |
| if (is_bitmap) { |
| result = AudioFormatGetPropertyInfo(fa, sizeof(UInt32), |
| &layout->mChannelBitmap, &size); |
| } else { |
| result = AudioFormatGetPropertyInfo(fa, sizeof(AudioChannelLayoutTag), |
| &tag, &size); |
| } |
| if (result != noErr || !size) { |
| OSSTATUS_DLOG(ERROR, result) |
| << "Failed to get AudioFormat property info, size=" << size; |
| return false; |
| } |
| |
| layout_storage.reset(new uint8_t[size]); |
| layout = reinterpret_cast<AudioChannelLayout*>(layout_storage.get()); |
| if (is_bitmap) { |
| result = AudioFormatGetProperty(fa, sizeof(UInt32), |
| &layout->mChannelBitmap, &size, layout); |
| } else { |
| result = AudioFormatGetProperty(fa, sizeof(AudioChannelLayoutTag), &tag, |
| &size, layout); |
| } |
| if (result != noErr) { |
| OSSTATUS_DLOG(ERROR, result) << "Failed to get AudioFormat property."; |
| return false; |
| } |
| } |
| |
| // There is no channel info for stereo, assume so for mono as well. |
| if (layout->mNumberChannelDescriptions <= 2) { |
| *channels = layout->mNumberChannelDescriptions; |
| } else { |
| *channels = 0; |
| for (UInt32 i = 0; i < layout->mNumberChannelDescriptions; ++i) { |
| if (layout->mChannelDescriptions[i].mChannelLabel != |
| kAudioChannelLabel_Unknown) |
| (*channels)++; |
| } |
| } |
| |
| DVLOG(1) << "Output channels: " << *channels; |
| return true; |
| } |
| |
| class AudioManagerMac::AudioPowerObserver : public base::PowerSuspendObserver { |
| public: |
| AudioPowerObserver() |
| : is_suspending_(false), |
| is_monitoring_(base::PowerMonitor::IsInitialized()), |
| num_resume_notifications_(0) { |
| // The PowerMonitor requires significant setup (a CFRunLoop and preallocated |
| // IO ports) so it's not available under unit tests. See the OSX impl of |
| // base::PowerMonitorDeviceSource for more details. |
| if (!is_monitoring_) |
| return; |
| base::PowerMonitor::AddPowerSuspendObserver(this); |
| } |
| |
| AudioPowerObserver(const AudioPowerObserver&) = delete; |
| AudioPowerObserver& operator=(const AudioPowerObserver&) = delete; |
| |
| ~AudioPowerObserver() override { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| if (!is_monitoring_) |
| return; |
| base::PowerMonitor::RemovePowerSuspendObserver(this); |
| } |
| |
| bool IsSuspending() const { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| return is_suspending_; |
| } |
| |
| size_t num_resume_notifications() const { return num_resume_notifications_; } |
| |
| bool ShouldDeferStreamStart() const { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| // Start() should be deferred if the system is in the middle of a suspend or |
| // has recently started the process of resuming. |
| return is_suspending_ || base::TimeTicks::Now() < earliest_start_time_; |
| } |
| |
| bool IsOnBatteryPower() const { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| return base::PowerMonitor::IsOnBatteryPower(); |
| } |
| |
| private: |
| void OnSuspend() override { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "OnSuspend"; |
| is_suspending_ = true; |
| } |
| |
| void OnResume() override { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "OnResume"; |
| ++num_resume_notifications_; |
| is_suspending_ = false; |
| earliest_start_time_ = |
| base::TimeTicks::Now() + base::Seconds(kStartDelayInSecsForPowerEvents); |
| } |
| |
| bool is_suspending_; |
| const bool is_monitoring_; |
| base::TimeTicks earliest_start_time_; |
| base::ThreadChecker thread_checker_; |
| size_t num_resume_notifications_; |
| }; |
| |
| AudioManagerMac::AudioManagerMac(std::unique_ptr<AudioThread> audio_thread, |
| AudioLogFactory* audio_log_factory) |
| : AudioManagerBase(std::move(audio_thread), audio_log_factory), |
| current_sample_rate_(0), |
| current_output_device_(kAudioDeviceUnknown), |
| in_shutdown_(false), |
| weak_ptr_factory_(this) { |
| SetMaxOutputStreamsAllowed(kMaxOutputStreams); |
| |
| // PostTask since AudioManager creation may be on the startup path and this |
| // may be slow. |
| GetTaskRunner()->PostTask( |
| FROM_HERE, base::BindOnce(&AudioManagerMac::InitializeOnAudioThread, |
| weak_ptr_factory_.GetWeakPtr())); |
| } |
| |
| AudioManagerMac::~AudioManagerMac() = default; |
| |
| void AudioManagerMac::ShutdownOnAudioThread() { |
| // We are now in shutdown mode. This flag disables MaybeChangeBufferSize() |
| // and IncreaseIOBufferSizeIfPossible() which both touches native Core Audio |
| // APIs and they can fail and disrupt tests during shutdown. |
| in_shutdown_ = true; |
| |
| // Even if tasks to close the streams are enqueued, they would not run |
| // leading to CHECKs getting hit in the destructor about open streams. Close |
| // them explicitly here. crbug.com/608049. |
| CloseAllInputStreams(); |
| CHECK(basic_input_streams_.empty()); |
| CHECK(low_latency_input_streams_.empty()); |
| |
| // Deinitialize power observer on audio thread, since it's initialized on the |
| // audio thread. Typically, constructor/destructor and |
| // InitializeOnAudioThread/ShutdownOnAudioThread are all run on the main |
| // thread, but this might not be true in testing. |
| power_observer_.reset(); |
| |
| AudioManagerBase::ShutdownOnAudioThread(); |
| } |
| |
| bool AudioManagerMac::HasAudioOutputDevices() { |
| return HasAudioHardware(kAudioHardwarePropertyDefaultOutputDevice); |
| } |
| |
| bool AudioManagerMac::HasAudioInputDevices() { |
| return HasAudioHardware(kAudioHardwarePropertyDefaultInputDevice); |
| } |
| |
| // static |
| int AudioManagerMac::HardwareSampleRateForDevice(AudioDeviceID device_id) { |
| DCHECK(AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread()); |
| Float64 nominal_sample_rate; |
| UInt32 info_size = sizeof(nominal_sample_rate); |
| |
| static const AudioObjectPropertyAddress kNominalSampleRateAddress = { |
| kAudioDevicePropertyNominalSampleRate, |
| kAudioObjectPropertyScopeGlobal, |
| kAudioObjectPropertyElementMaster |
| }; |
| OSStatus result = AudioObjectGetPropertyData(device_id, |
| &kNominalSampleRateAddress, |
| 0, |
| 0, |
| &info_size, |
| &nominal_sample_rate); |
| if (result != noErr) { |
| OSSTATUS_DLOG(WARNING, result) |
| << "Could not get default sample rate for device: " << device_id; |
| return 0; |
| } |
| |
| return static_cast<int>(nominal_sample_rate); |
| } |
| |
| // static |
| int AudioManagerMac::HardwareSampleRate() { |
| // Determine the default output device's sample-rate. |
| AudioDeviceID device_id = kAudioObjectUnknown; |
| if (!GetDefaultOutputDevice(&device_id)) |
| return kFallbackSampleRate; |
| |
| return HardwareSampleRateForDevice(device_id); |
| } |
| |
| void AudioManagerMac::GetAudioInputDeviceNames( |
| media::AudioDeviceNames* device_names) { |
| DCHECK(device_names->empty()); |
| GetAudioDeviceInfo(true, device_names); |
| } |
| |
| void AudioManagerMac::GetAudioOutputDeviceNames( |
| media::AudioDeviceNames* device_names) { |
| DCHECK(device_names->empty()); |
| GetAudioDeviceInfo(false, device_names); |
| } |
| |
| AudioParameters AudioManagerMac::GetInputStreamParameters( |
| const std::string& device_id) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| AudioDeviceID device = GetAudioDeviceIdByUId(true, device_id); |
| if (device == kAudioObjectUnknown) { |
| DLOG(ERROR) << "Invalid device " << device_id; |
| return AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY, |
| CHANNEL_LAYOUT_STEREO, kFallbackSampleRate, |
| ChooseBufferSize(true, kFallbackSampleRate)); |
| } |
| |
| int channels = 0; |
| ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO; |
| if (GetDeviceChannels(device, AUElement::INPUT, &channels) && channels <= 2) { |
| channel_layout = GuessChannelLayout(channels); |
| } else { |
| DLOG(ERROR) << "Failed to get the device channels, use stereo as default " |
| << "for device " << device_id; |
| } |
| |
| int sample_rate = HardwareSampleRateForDevice(device); |
| if (!sample_rate) |
| sample_rate = kFallbackSampleRate; |
| |
| // Due to the sharing of the input and output buffer sizes, we need to choose |
| // the input buffer size based on the output sample rate. See |
| // http://crbug.com/154352. |
| const int buffer_size = ChooseBufferSize(true, sample_rate); |
| |
| // TODO(grunell): query the native channel layout for the specific device. |
| AudioParameters params( |
| AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, sample_rate, |
| buffer_size, |
| AudioParameters::HardwareCapabilities( |
| GetMinAudioBufferSizeMacOS(limits::kMinAudioBufferSize, sample_rate), |
| limits::kMaxAudioBufferSize)); |
| |
| if (DeviceSupportsAmbientNoiseReduction(device)) { |
| params.set_effects(AudioParameters::NOISE_SUPPRESSION); |
| } |
| |
| // VoiceProcessingIO is only supported on MacOS 10.12 and cannot be used on |
| // aggregate devices, since it creates an aggregate device itself. It also |
| // only runs in mono, but we allow upmixing to stereo since we can't claim a |
| // device works either in stereo without echo cancellation or mono with echo |
| // cancellation. |
| if (base::mac::IsAtLeastOS10_12() && |
| (params.channel_layout() == CHANNEL_LAYOUT_MONO || |
| params.channel_layout() == CHANNEL_LAYOUT_STEREO) && |
| core_audio_mac::GetDeviceTransportType(device) != |
| kAudioDeviceTransportTypeAggregate) { |
| params.set_effects(params.effects() | |
| AudioParameters::EXPERIMENTAL_ECHO_CANCELLER); |
| } |
| |
| return params; |
| } |
| |
| std::string AudioManagerMac::GetAssociatedOutputDeviceID( |
| const std::string& input_device_unique_id) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| AudioObjectID input_device_id = |
| GetAudioDeviceIdByUId(true, input_device_unique_id); |
| if (input_device_id == kAudioObjectUnknown) |
| return std::string(); |
| |
| std::vector<AudioObjectID> related_device_ids = |
| core_audio_mac::GetRelatedDeviceIDs(input_device_id); |
| |
| // Defined as a set as device IDs might be duplicated in |
| // GetRelatedDeviceIDs(). |
| base::flat_set<AudioObjectID> related_output_device_ids; |
| for (AudioObjectID device_id : related_device_ids) { |
| if (core_audio_mac::GetNumStreams(device_id, false /* is_input */) > 0) |
| related_output_device_ids.insert(device_id); |
| } |
| |
| // Return the device ID if there is only one associated device. |
| // When there are multiple associated devices, we currently do not have a way |
| // to detect if a device (e.g. a digital output device) is actually connected |
| // to an endpoint, so we cannot randomly pick a device. |
| if (related_output_device_ids.size() == 1) { |
| absl::optional<std::string> related_unique_id = |
| core_audio_mac::GetDeviceUniqueID(*related_output_device_ids.begin()); |
| if (related_unique_id) |
| return std::move(*related_unique_id); |
| } |
| |
| return std::string(); |
| } |
| |
| const char* AudioManagerMac::GetName() { |
| return "Mac"; |
| } |
| |
| AudioOutputStream* AudioManagerMac::MakeLinearOutputStream( |
| const AudioParameters& params, |
| const LogCallback& log_callback) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| return MakeLowLatencyOutputStream(params, std::string(), log_callback); |
| } |
| |
| AudioOutputStream* AudioManagerMac::MakeLowLatencyOutputStream( |
| const AudioParameters& params, |
| const std::string& device_id, |
| const LogCallback& log_callback) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| bool device_listener_first_init = false; |
| // Lazily create the audio device listener on the first stream creation, |
| // even if getting an audio device fails. Otherwise, if we have 0 audio |
| // devices, the listener will never be initialized, and new valid devices |
| // will never be detected. |
| if (!output_device_listener_) { |
| // NOTE: Use BindToCurrentLoop() to ensure the callback is always PostTask'd |
| // even if OSX calls us on the right thread. Some CoreAudio drivers will |
| // fire the callbacks during stream creation, leading to re-entrancy issues |
| // otherwise. See http://crbug.com/349604 |
| output_device_listener_ = std::make_unique<AudioDeviceListenerMac>( |
| BindToCurrentLoop(base::BindRepeating( |
| &AudioManagerMac::HandleDeviceChanges, base::Unretained(this)))); |
| device_listener_first_init = true; |
| } |
| |
| AudioDeviceID device = GetAudioDeviceIdByUId(false, device_id); |
| if (device == kAudioObjectUnknown) { |
| DLOG(ERROR) << "Failed to open output device: " << device_id; |
| return NULL; |
| } |
| |
| // Only set the device and sample rate if we just initialized the device |
| // listener. |
| if (device_listener_first_init) { |
| // Only set the current output device for the default device. |
| if (AudioDeviceDescription::IsDefaultDevice(device_id)) |
| current_output_device_ = device; |
| // Just use the current sample rate since we don't allow non-native sample |
| // rates on OSX. |
| current_sample_rate_ = params.sample_rate(); |
| } |
| |
| AUHALStream* stream = new AUHALStream(this, params, device, log_callback); |
| output_streams_.push_back(stream); |
| return stream; |
| } |
| |
| std::string AudioManagerMac::GetDefaultOutputDeviceID() { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| return GetDefaultDeviceID(false /* is_input */); |
| } |
| |
| std::string AudioManagerMac::GetDefaultInputDeviceID() { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| return GetDefaultDeviceID(true /* is_input */); |
| } |
| |
| std::string AudioManagerMac::GetDefaultDeviceID(bool is_input) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| AudioDeviceID device_id = kAudioObjectUnknown; |
| if (!GetDefaultDevice(&device_id, is_input)) |
| return std::string(); |
| |
| const AudioObjectPropertyAddress property_address = { |
| kAudioDevicePropertyDeviceUID, |
| kAudioObjectPropertyScopeGlobal, |
| kAudioObjectPropertyElementMaster |
| }; |
| CFStringRef device_uid = NULL; |
| UInt32 size = sizeof(device_uid); |
| OSStatus status = AudioObjectGetPropertyData(device_id, |
| &property_address, |
| 0, |
| NULL, |
| &size, |
| &device_uid); |
| if (status != kAudioHardwareNoError || !device_uid) |
| return std::string(); |
| |
| std::string ret(base::SysCFStringRefToUTF8(device_uid)); |
| CFRelease(device_uid); |
| |
| return ret; |
| } |
| |
| AudioInputStream* AudioManagerMac::MakeLinearInputStream( |
| const AudioParameters& params, |
| const std::string& device_id, |
| const LogCallback& log_callback) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| DCHECK_EQ(AudioParameters::AUDIO_PCM_LINEAR, params.format()); |
| AudioInputStream* stream = new PCMQueueInAudioInputStream(this, params); |
| basic_input_streams_.push_back(stream); |
| return stream; |
| } |
| |
| AudioInputStream* AudioManagerMac::MakeLowLatencyInputStream( |
| const AudioParameters& params, |
| const std::string& device_id, |
| const LogCallback& log_callback) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| DCHECK_EQ(AudioParameters::AUDIO_PCM_LOW_LATENCY, params.format()); |
| // Gets the AudioDeviceID that refers to the AudioInputDevice with the device |
| // unique id. This AudioDeviceID is used to set the device for Audio Unit. |
| AudioDeviceID audio_device_id = GetAudioDeviceIdByUId(true, device_id); |
| if (audio_device_id == kAudioObjectUnknown) { |
| return nullptr; |
| } |
| |
| VoiceProcessingMode voice_processing_mode = |
| (params.effects() & AudioParameters::ECHO_CANCELLER) |
| ? VoiceProcessingMode::kEnabled |
| : VoiceProcessingMode::kDisabled; |
| |
| auto* stream = new AUAudioInputStream(this, params, audio_device_id, |
| log_callback, voice_processing_mode); |
| low_latency_input_streams_.push_back(stream); |
| return stream; |
| } |
| |
| AudioParameters AudioManagerMac::GetPreferredOutputStreamParameters( |
| const std::string& output_device_id, |
| const AudioParameters& input_params) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| const AudioDeviceID device = GetAudioDeviceIdByUId(false, output_device_id); |
| if (device == kAudioObjectUnknown) { |
| DLOG(ERROR) << "Invalid output device " << output_device_id; |
| return input_params.IsValid() |
| ? input_params |
| : AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY, |
| CHANNEL_LAYOUT_STEREO, kFallbackSampleRate, |
| ChooseBufferSize(false, kFallbackSampleRate)); |
| } |
| |
| const bool has_valid_input_params = input_params.IsValid(); |
| const int hardware_sample_rate = HardwareSampleRateForDevice(device); |
| |
| // Allow pass through buffer sizes. If concurrent input and output streams |
| // exist, they will use the smallest buffer size amongst them. As such, each |
| // stream must be able to FIFO requests appropriately when this happens. |
| int buffer_size; |
| if (has_valid_input_params) { |
| // Ensure the latency asked for is maintained, even if the sample rate is |
| // changed here. |
| const int scaled_buffer_size = input_params.frames_per_buffer() * |
| hardware_sample_rate / |
| input_params.sample_rate(); |
| // If passed in via the input_params we allow buffer sizes to go as |
| // low as the the kMinAudioBufferSize, ignoring what |
| // ChooseBufferSize() normally returns. |
| buffer_size = |
| std::min(static_cast<int>(limits::kMaxAudioBufferSize), |
| std::max(scaled_buffer_size, |
| static_cast<int>(limits::kMinAudioBufferSize))); |
| } else { |
| buffer_size = ChooseBufferSize(false, hardware_sample_rate); |
| } |
| |
| int hardware_channels; |
| if (!GetDeviceChannels(device, AUElement::OUTPUT, &hardware_channels)) |
| hardware_channels = 2; |
| |
| // Use the input channel count and channel layout if possible. Let OSX take |
| // care of remapping the channels; this lets user specified channel layouts |
| // work correctly. |
| int output_channels = input_params.channels(); |
| ChannelLayout channel_layout = input_params.channel_layout(); |
| if (!has_valid_input_params || output_channels > hardware_channels) { |
| output_channels = hardware_channels; |
| channel_layout = GuessChannelLayout(output_channels); |
| if (channel_layout == CHANNEL_LAYOUT_UNSUPPORTED) |
| channel_layout = CHANNEL_LAYOUT_DISCRETE; |
| } |
| |
| AudioParameters params( |
| AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, |
| hardware_sample_rate, buffer_size, |
| AudioParameters::HardwareCapabilities( |
| GetMinAudioBufferSizeMacOS(limits::kMinAudioBufferSize, |
| hardware_sample_rate), |
| limits::kMaxAudioBufferSize)); |
| params.set_channels_for_discrete(output_channels); |
| return params; |
| } |
| |
| void AudioManagerMac::InitializeOnAudioThread() { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| InitializeCoreAudioDispatchOverride(); |
| power_observer_ = std::make_unique<AudioPowerObserver>(); |
| } |
| |
| void AudioManagerMac::HandleDeviceChanges() { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| const int new_sample_rate = HardwareSampleRate(); |
| AudioDeviceID new_output_device; |
| GetDefaultOutputDevice(&new_output_device); |
| |
| if (current_sample_rate_ == new_sample_rate && |
| current_output_device_ == new_output_device) { |
| return; |
| } |
| |
| current_sample_rate_ = new_sample_rate; |
| current_output_device_ = new_output_device; |
| NotifyAllOutputDeviceChangeListeners(); |
| } |
| |
| int AudioManagerMac::ChooseBufferSize(bool is_input, int sample_rate) { |
| // kMinAudioBufferSize is too small for the output side because |
| // CoreAudio can get into under-run if the renderer fails delivering data |
| // to the browser within the allowed time by the OS. The workaround is to |
| // use 256 samples as the default output buffer size for sample rates |
| // smaller than 96KHz. |
| // TODO(xians): Remove this workaround after WebAudio supports user defined |
| // buffer size. See https://github.com/WebAudio/web-audio-api/issues/348 |
| // for details. |
| int buffer_size = |
| is_input ? limits::kMinAudioBufferSize : 2 * limits::kMinAudioBufferSize; |
| const int user_buffer_size = GetUserBufferSize(); |
| buffer_size = user_buffer_size |
| ? user_buffer_size |
| : GetMinAudioBufferSizeMacOS(buffer_size, sample_rate); |
| return buffer_size; |
| } |
| |
| bool AudioManagerMac::IsSuspending() const { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| return power_observer_->IsSuspending(); |
| } |
| |
| bool AudioManagerMac::ShouldDeferStreamStart() const { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| return power_observer_->ShouldDeferStreamStart(); |
| } |
| |
| bool AudioManagerMac::IsOnBatteryPower() const { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| return power_observer_->IsOnBatteryPower(); |
| } |
| |
| size_t AudioManagerMac::GetNumberOfResumeNotifications() const { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| return power_observer_->num_resume_notifications(); |
| } |
| |
| bool AudioManagerMac::MaybeChangeBufferSize(AudioDeviceID device_id, |
| AudioUnit audio_unit, |
| AudioUnitElement element, |
| size_t desired_buffer_size, |
| bool* size_was_changed, |
| size_t* io_buffer_frame_size) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| if (in_shutdown_) { |
| DVLOG(1) << "Disabled since we are shutting down"; |
| return false; |
| } |
| const bool is_input = (element == 1); |
| DVLOG(1) << "MaybeChangeBufferSize(id=0x" << std::hex << device_id |
| << ", is_input=" << is_input << ", desired_buffer_size=" << std::dec |
| << desired_buffer_size << ")"; |
| |
| *size_was_changed = false; |
| *io_buffer_frame_size = 0; |
| |
| // Log the device name (and id) for debugging purposes. |
| std::string device_name = GetAudioDeviceNameFromDeviceId(device_id, is_input); |
| DVLOG(1) << "name: " << device_name << " (ID: 0x" << std::hex << device_id |
| << ")"; |
| |
| // Get the current size of the I/O buffer for the specified device. The |
| // property is read on a global scope, hence using element 0. The default IO |
| // buffer size on Mac OSX for OS X 10.9 and later is 512 audio frames. |
| UInt32 buffer_size = 0; |
| UInt32 property_size = sizeof(buffer_size); |
| OSStatus result = AudioUnitGetProperty( |
| audio_unit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, |
| 0, &buffer_size, &property_size); |
| if (result != noErr) { |
| OSSTATUS_DLOG(ERROR, result) |
| << "AudioUnitGetProperty(kAudioDevicePropertyBufferFrameSize) failed."; |
| return false; |
| } |
| // Store the currently used (not changed yet) I/O buffer frame size. |
| *io_buffer_frame_size = buffer_size; |
| |
| DVLOG(1) << "current IO buffer size: " << buffer_size; |
| DVLOG(1) << "#output streams: " << output_streams_.size(); |
| DVLOG(1) << "#input streams: " << low_latency_input_streams_.size(); |
| |
| // Check if a buffer size change is required. If the caller asks for a |
| // reduced size (|desired_buffer_size| < |buffer_size|), the new lower size |
| // will be set. For larger buffer sizes, we have to perform some checks to |
| // see if the size can actually be changed. If there is any other active |
| // streams on the same device, either input or output, a larger size than |
| // their requested buffer size can't be set. The reason is that an existing |
| // stream can't handle buffer size larger than its requested buffer size. |
| // See http://crbug.com/428706 for a reason why. |
| |
| if (buffer_size == desired_buffer_size) |
| return true; |
| |
| if (desired_buffer_size > buffer_size) { |
| // Do NOT set the buffer size if there is another output stream using |
| // the same device with a smaller requested buffer size. |
| // Note, for the caller stream, its requested_buffer_size() will be the same |
| // as |desired_buffer_size|, so it won't return true due to comparing with |
| // itself. |
| for (auto* stream : output_streams_) { |
| if (stream->device_id() == device_id && |
| stream->requested_buffer_size() < desired_buffer_size) { |
| return true; |
| } |
| } |
| |
| // Do NOT set the buffer size if there is another input stream using |
| // the same device with a smaller buffer size. |
| for (auto* stream : low_latency_input_streams_) { |
| if (stream->device_id() == device_id && |
| stream->requested_buffer_size() < desired_buffer_size) { |
| return true; |
| } |
| } |
| } |
| |
| // In this scope we know that the IO buffer size should be modified. But |
| // first, verify that |desired_buffer_size| is within the valid range and |
| // modify the desired buffer size if it is outside this range. |
| // Note that, we have found that AudioUnitSetProperty(PropertyBufferFrameSize) |
| // does in fact do this limitation internally and report noErr even if the |
| // user tries to set an invalid size. As an example, asking for a size of |
| // 4410 will on most devices be limited to 4096 without any further notice. |
| UInt32 minimum = buffer_size; |
| UInt32 maximum = buffer_size; |
| result = GetIOBufferFrameSizeRange(device_id, is_input, &minimum, &maximum); |
| if (result != noErr) { |
| // OS error is logged in GetIOBufferFrameSizeRange(). |
| return false; |
| } |
| DVLOG(1) << "valid IO buffer size range: [" << minimum << ", " << maximum |
| << "]"; |
| buffer_size = desired_buffer_size; |
| if (buffer_size < minimum) |
| buffer_size = minimum; |
| else if (buffer_size > maximum) |
| buffer_size = maximum; |
| DVLOG(1) << "validated desired buffer size: " << buffer_size; |
| |
| // Set new (and valid) I/O buffer size for the specified device. The property |
| // is set on a global scope, hence using element 0. |
| result = AudioUnitSetProperty(audio_unit, kAudioDevicePropertyBufferFrameSize, |
| kAudioUnitScope_Global, 0, &buffer_size, |
| sizeof(buffer_size)); |
| OSSTATUS_DLOG_IF(ERROR, result != noErr, result) |
| << "AudioUnitSetProperty(kAudioDevicePropertyBufferFrameSize) failed. " |
| << "Size:: " << buffer_size; |
| *size_was_changed = (result == noErr); |
| DVLOG_IF(1, result == noErr) << "IO buffer size changed to: " << buffer_size; |
| // Store the currently used (after a change) I/O buffer frame size. |
| *io_buffer_frame_size = buffer_size; |
| return result == noErr; |
| } |
| |
| // static |
| base::TimeDelta AudioManagerMac::GetHardwareLatency( |
| AudioUnit audio_unit, |
| AudioDeviceID device_id, |
| AudioObjectPropertyScope scope, |
| int sample_rate) { |
| if (!audio_unit || device_id == kAudioObjectUnknown) { |
| DLOG(WARNING) << "Audio unit object is NULL or device ID is unknown"; |
| return base::TimeDelta(); |
| } |
| |
| // 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); |
| OSSTATUS_DLOG_IF(WARNING, result != noErr, result) |
| << "Could not get audio unit latency"; |
| |
| // Get audio device latency. |
| AudioObjectPropertyAddress property_address = { |
| kAudioDevicePropertyLatency, scope, kAudioObjectPropertyElementMaster}; |
| UInt32 device_latency_frames = 0; |
| size = sizeof(device_latency_frames); |
| result = AudioObjectGetPropertyData(device_id, &property_address, 0, nullptr, |
| &size, &device_latency_frames); |
| OSSTATUS_DLOG_IF(WARNING, result != noErr, result) |
| << "Could not get audio device latency."; |
| |
| // Retrieve stream ids and take the stream latency from the first stream. |
| // There may be multiple streams with different latencies, but since we're |
| // likely using this delay information for a/v sync we must choose one of |
| // them; Apple recommends just taking the first entry. |
| // |
| // TODO(dalecurtis): Refactor all these "get data size" + "get data" calls |
| // into a common utility function that just returns a std::unique_ptr. |
| UInt32 stream_latency_frames = 0; |
| property_address.mSelector = kAudioDevicePropertyStreams; |
| result = AudioObjectGetPropertyDataSize(device_id, &property_address, 0, |
| nullptr, &size); |
| if (result == noErr && size >= sizeof(AudioStreamID)) { |
| std::unique_ptr<uint8_t[]> stream_id_storage(new uint8_t[size]); |
| AudioStreamID* stream_ids = |
| reinterpret_cast<AudioStreamID*>(stream_id_storage.get()); |
| result = AudioObjectGetPropertyData(device_id, &property_address, 0, |
| nullptr, &size, stream_ids); |
| if (result == noErr) { |
| property_address.mSelector = kAudioStreamPropertyLatency; |
| size = sizeof(stream_latency_frames); |
| result = |
| AudioObjectGetPropertyData(stream_ids[0], &property_address, 0, |
| nullptr, &size, &stream_latency_frames); |
| OSSTATUS_DLOG_IF(WARNING, result != noErr, result) |
| << "Could not get stream latency for stream #0."; |
| } else { |
| OSSTATUS_DLOG(WARNING, result) |
| << "Could not get audio device stream ids."; |
| } |
| } else { |
| OSSTATUS_DLOG_IF(WARNING, result != noErr, result) |
| << "Could not get audio device stream ids size."; |
| } |
| |
| return base::Seconds(audio_unit_latency_sec) + |
| AudioTimestampHelper::FramesToTime( |
| device_latency_frames + stream_latency_frames, sample_rate); |
| } |
| |
| bool AudioManagerMac::DeviceSupportsAmbientNoiseReduction( |
| AudioDeviceID device_id) { |
| return AudioObjectHasProperty(device_id, &kNoiseReductionPropertyAddress); |
| } |
| |
| bool AudioManagerMac::SuppressNoiseReduction(AudioDeviceID device_id) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| DCHECK(DeviceSupportsAmbientNoiseReduction(device_id)); |
| NoiseReductionState& state = device_noise_reduction_states_[device_id]; |
| if (state.suppression_count == 0) { |
| UInt32 initially_enabled = 0; |
| UInt32 size = sizeof(initially_enabled); |
| OSStatus result = |
| AudioObjectGetPropertyData(device_id, &kNoiseReductionPropertyAddress, |
| 0, nullptr, &size, &initially_enabled); |
| if (result != noErr) |
| return false; |
| |
| if (initially_enabled) { |
| const UInt32 disable = 0; |
| OSStatus result = |
| AudioObjectSetPropertyData(device_id, &kNoiseReductionPropertyAddress, |
| 0, nullptr, sizeof(disable), &disable); |
| if (result != noErr) { |
| OSSTATUS_DLOG(WARNING, result) |
| << "Failed to disable ambient noise reduction for device: " |
| << std::hex << device_id; |
| } |
| state.initial_state = NoiseReductionState::ENABLED; |
| } else { |
| state.initial_state = NoiseReductionState::DISABLED; |
| } |
| } |
| |
| // Only increase the counter if suppression succeeded or is already active. |
| ++state.suppression_count; |
| return true; |
| } |
| |
| void AudioManagerMac::UnsuppressNoiseReduction(AudioDeviceID device_id) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| NoiseReductionState& state = device_noise_reduction_states_[device_id]; |
| DCHECK_NE(state.suppression_count, 0); |
| --state.suppression_count; |
| if (state.suppression_count == 0) { |
| if (state.initial_state == NoiseReductionState::ENABLED) { |
| const UInt32 enable = 1; |
| OSStatus result = |
| AudioObjectSetPropertyData(device_id, &kNoiseReductionPropertyAddress, |
| 0, nullptr, sizeof(enable), &enable); |
| if (result != noErr) { |
| OSSTATUS_DLOG(WARNING, result) |
| << "Failed to re-enable ambient noise reduction for device: " |
| << std::hex << device_id; |
| } |
| } |
| } |
| } |
| |
| bool AudioManagerMac::AudioDeviceIsUsedForInput(AudioDeviceID device_id) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| if (!basic_input_streams_.empty()) { |
| // For Audio Queues and in the default case (Mac OS X), the audio comes |
| // from the system’s default audio input device as set by a user in System |
| // Preferences. |
| AudioDeviceID default_id; |
| GetDefaultDevice(&default_id, true); |
| if (default_id == device_id) |
| return true; |
| } |
| |
| // Each low latency streams has its own device ID. |
| for (auto* stream : low_latency_input_streams_) { |
| if (stream->device_id() == device_id) |
| return true; |
| } |
| return false; |
| } |
| |
| void AudioManagerMac::ReleaseOutputStream(AudioOutputStream* stream) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| output_streams_.remove(static_cast<AUHALStream*>(stream)); |
| AudioManagerBase::ReleaseOutputStream(stream); |
| } |
| |
| void AudioManagerMac::ReleaseOutputStreamUsingRealDevice( |
| AudioOutputStream* stream, |
| AudioDeviceID device_id) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| DVLOG(1) << "Closing output stream with id=0x" << std::hex << device_id; |
| DVLOG(1) << "requested_buffer_size: " |
| << static_cast<AUHALStream*>(stream)->requested_buffer_size(); |
| |
| // Start by closing down the specified output stream. |
| output_streams_.remove(static_cast<AUHALStream*>(stream)); |
| AudioManagerBase::ReleaseOutputStream(stream); |
| } |
| |
| void AudioManagerMac::ReleaseInputStream(AudioInputStream* stream) { |
| DCHECK(GetTaskRunner()->BelongsToCurrentThread()); |
| auto stream_it = std::find(basic_input_streams_.begin(), |
| basic_input_streams_.end(), |
| stream); |
| if (stream_it == basic_input_streams_.end()) |
| low_latency_input_streams_.remove(static_cast<AUAudioInputStream*>(stream)); |
| else |
| basic_input_streams_.erase(stream_it); |
| |
| AudioManagerBase::ReleaseInputStream(stream); |
| } |
| |
| std::unique_ptr<AudioManager> CreateAudioManager( |
| std::unique_ptr<AudioThread> audio_thread, |
| AudioLogFactory* audio_log_factory) { |
| return std::make_unique<AudioManagerMac>(std::move(audio_thread), |
| audio_log_factory); |
| } |
| |
| } // namespace media |
