path: root/chromium/media/audio/win/audio_manager_win.cc

// 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/win/audio_manager_win.h"

#include <windows.h>

#include <objbase.h>  // This has to be before initguid.h

#include <initguid.h>
#include <mmsystem.h>
#include <setupapi.h>
#include <stddef.h>

#include <memory>
#include <utility>

#include "base/bind.h"
#include "base/bind_helpers.h"
#include "base/command_line.h"
#include "base/metrics/histogram_functions.h"
#include "base/strings/string_number_conversions.h"
#include "base/win/windows_version.h"
#include "media/audio/audio_device_description.h"
#include "media/audio/audio_io.h"
#include "media/audio/win/audio_device_listener_win.h"
#include "media/audio/win/audio_low_latency_input_win.h"
#include "media/audio/win/audio_low_latency_output_win.h"
#include "media/audio/win/core_audio_util_win.h"
#include "media/audio/win/device_enumeration_win.h"
#include "media/audio/win/waveout_output_win.h"
#include "media/base/audio_parameters.h"
#include "media/base/bind_to_current_loop.h"
#include "media/base/channel_layout.h"
#include "media/base/limits.h"
#include "media/base/media_switches.h"

// The following are defined in various DDK headers, and we (re)define them here
// to avoid adding the DDK as a chrome dependency.
#define DRV_QUERYDEVICEINTERFACE 0x80c
#define DRVM_MAPPER_PREFERRED_GET 0x2015
#define DRV_QUERYDEVICEINTERFACESIZE 0x80d
DEFINE_GUID(AM_KSCATEGORY_AUDIO,
            0x6994ad04,
            0x93ef,
            0x11d0,
            0xa3,
            0xcc,
            0x00,
            0xa0,
            0xc9,
            0x22,
            0x31,
            0x96);

namespace media {

// Maximum number of output streams that can be open simultaneously.
static const int kMaxOutputStreams = 50;

// Up to 8 channels can be passed to the driver.  This should work, given the
// right drivers, but graceful error handling is needed.
static const int kWinMaxChannels = 8;

// Buffer size to use for input and output stream when a proper size can't be
// determined from the system
static const int kFallbackBufferSize = 2048;

static int NumberOfWaveOutBuffers() {
  // Use the user provided buffer count if provided.
  int buffers = 0;
  std::string buffers_str(
      base::CommandLine::ForCurrentProcess()->GetSwitchValueASCII(
          switches::kWaveOutBuffers));
  if (base::StringToInt(buffers_str, &buffers) && buffers > 0) {
    return buffers;
  }

  return 3;
}

static bool IsSupported(HRESULT hr) {
  return hr != S_FALSE && SUCCEEDED(hr);
}

// Records bitstream output support to histograms. Follows information from:
// https://docs.microsoft.com/en-us/windows/desktop/coreaudio/representing-formats-for-iec-61937-transmissions
static void LogBitstreamOutputSupport() {
  auto client = CoreAudioUtil::CreateClient(
      AudioDeviceDescription::kDefaultDeviceId, eRender, eConsole);

  // Happens if no audio output devices are available.
  if (!client)
    return;

  WAVEFORMATEXTENSIBLE wfext;
  memset(&wfext, 0, sizeof(wfext));

  // See link in function comment for where each value comes from.
  wfext.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
  wfext.Format.nChannels = 2;
  wfext.Format.nSamplesPerSec = 192000;
  wfext.Format.nAvgBytesPerSec = 768000;
  wfext.Format.nBlockAlign = 4;
  wfext.Format.wBitsPerSample = 16;
  wfext.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
  wfext.Samples.wValidBitsPerSample = 16;
  wfext.dwChannelMask = KSAUDIO_SPEAKER_7POINT1_SURROUND;

  // Test Dolby Digital+ / Atmos support. For whatever reason Atmos doesn't use
  // the KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL_PLUS_ATMOS SubFormat.
  wfext.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL_PLUS;

  HRESULT hr = client->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE,
                                         &wfext.Format, nullptr);
  base::UmaHistogramBoolean("Media.Audio.Bitstream.EAC3", IsSupported(hr));

  // Test Dolby TrueHD.
  wfext.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_MLP;
  hr = client->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfext.Format,
                                 nullptr);
  base::UmaHistogramBoolean("Media.Audio.Bitstream.TrueHD", IsSupported(hr));

  // Test DTS-HD.
  wfext.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_DTS_HD;
  hr = client->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfext.Format,
                                 nullptr);
  base::UmaHistogramBoolean("Media.Audio.Bitstream.DTS-HD", IsSupported(hr));

  // Older bitstream formats run at lower sampling rates.
  wfext.Format.nSamplesPerSec = 48000;
  wfext.Format.nAvgBytesPerSec = 192000;

  // Test AC3.
  wfext.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL;
  hr = client->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfext.Format,
                                 nullptr);
  base::UmaHistogramBoolean("Media.Audio.Bitstream.AC3", IsSupported(hr));

  // Test DTS.
  wfext.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_DTS;
  hr = client->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfext.Format,
                                 nullptr);
  base::UmaHistogramBoolean("Media.Audio.Bitstream.DTS", IsSupported(hr));
}

AudioManagerWin::AudioManagerWin(std::unique_ptr<AudioThread> audio_thread,
                                 AudioLogFactory* audio_log_factory)
    : AudioManagerBase(std::move(audio_thread), audio_log_factory) {
  // |CoreAudioUtil::IsSupported()| uses static variables to avoid doing
  // multiple initializations.  This is however not thread safe.
  // So, here we call it explicitly before we kick off the audio thread
  // or do any other work.
  CoreAudioUtil::IsSupported();

  SetMaxOutputStreamsAllowed(kMaxOutputStreams);

  // WARNING: This may be executed on the UI loop, do not add any code here
  // which loads libraries or attempts to call out into the OS.  Instead add
  // such code to the InitializeOnAudioThread() method below.

  // In case we are already on the audio thread (i.e. when running out of
  // process audio), don't post.
  if (GetTaskRunner()->BelongsToCurrentThread()) {
    this->InitializeOnAudioThread();
    return;
  }

  // Task must be posted last to avoid races from handing out "this" to the
  // audio thread. Unretained is safe since we join the audio thread before
  // destructing |this|.
  GetTaskRunner()->PostTask(
      FROM_HERE, base::BindOnce(&AudioManagerWin::InitializeOnAudioThread,
                                base::Unretained(this)));
}

AudioManagerWin::~AudioManagerWin() = default;

void AudioManagerWin::ShutdownOnAudioThread() {
  AudioManagerBase::ShutdownOnAudioThread();

  // Destroy AudioDeviceListenerWin instance on the audio thread because it
  // expects to be constructed and destroyed on the same thread.
  output_device_listener_.reset();
}

bool AudioManagerWin::HasAudioOutputDevices() {
  return (::waveOutGetNumDevs() != 0);
}

bool AudioManagerWin::HasAudioInputDevices() {
  return (::waveInGetNumDevs() != 0);
}

void AudioManagerWin::InitializeOnAudioThread() {
  DCHECK(GetTaskRunner()->BelongsToCurrentThread());

  // Delay metrics recording to avoid any issues at startup.
  GetTaskRunner()->PostDelayedTask(FROM_HERE,
                                   base::BindOnce(&LogBitstreamOutputSupport),
                                   base::TimeDelta::FromSeconds(15));

  // AudioDeviceListenerWin must be initialized on a COM thread.
  output_device_listener_.reset(new AudioDeviceListenerWin(BindToCurrentLoop(
      base::Bind(&AudioManagerWin::NotifyAllOutputDeviceChangeListeners,
                 base::Unretained(this)))));
}

void AudioManagerWin::GetAudioDeviceNamesImpl(bool input,
                                              AudioDeviceNames* device_names) {
  DCHECK(device_names->empty());
  // Enumerate all active audio-endpoint capture devices.
  if (input)
    GetInputDeviceNamesWin(device_names);
  else
    GetOutputDeviceNamesWin(device_names);

  if (!device_names->empty()) {
    device_names->push_front(AudioDeviceName::CreateCommunications());

    // Always add default device parameters as first element.
    device_names->push_front(AudioDeviceName::CreateDefault());
  }
}

void AudioManagerWin::GetAudioInputDeviceNames(AudioDeviceNames* device_names) {
  GetAudioDeviceNamesImpl(true, device_names);
}

void AudioManagerWin::GetAudioOutputDeviceNames(
    AudioDeviceNames* device_names) {
  GetAudioDeviceNamesImpl(false, device_names);
}

AudioParameters AudioManagerWin::GetInputStreamParameters(
    const std::string& device_id) {
  AudioParameters parameters;
  HRESULT hr =
      CoreAudioUtil::GetPreferredAudioParameters(device_id, false, &parameters);

  if (FAILED(hr) || !parameters.IsValid()) {
    LOG(WARNING) << "Unable to get preferred audio params for " << device_id
                 << " 0x" << std::hex << hr;
    // TODO(tommi): We appear to have callers to GetInputStreamParameters that
    // rely on getting valid audio parameters returned for an invalid or
    // unavailable device. We should track down those code paths (it is likely
    // that they actually don't need a real device but depend on the audio
    // code path somehow for a configuration - e.g. tab capture).
    parameters =
        AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
                        CHANNEL_LAYOUT_STEREO, 48000, kFallbackBufferSize);
  }

  int user_buffer_size = GetUserBufferSize();
  if (user_buffer_size)
    parameters.set_frames_per_buffer(user_buffer_size);

  return parameters;
}

std::string AudioManagerWin::GetAssociatedOutputDeviceID(
    const std::string& input_device_id) {
  return CoreAudioUtil::GetMatchingOutputDeviceID(input_device_id);
}

const char* AudioManagerWin::GetName() {
  return "Windows";
}

// Factory for the implementations of AudioOutputStream for AUDIO_PCM_LINEAR
// mode.
// - PCMWaveOutAudioOutputStream: Based on the waveOut API.
AudioOutputStream* AudioManagerWin::MakeLinearOutputStream(
    const AudioParameters& params,
    const LogCallback& log_callback) {
  DCHECK_EQ(AudioParameters::AUDIO_PCM_LINEAR, params.format());
  if (params.channels() > kWinMaxChannels)
    return NULL;

  return new PCMWaveOutAudioOutputStream(this, params, NumberOfWaveOutBuffers(),
                                         WAVE_MAPPER);
}

// Factory for the implementations of AudioOutputStream for
// AUDIO_PCM_LOW_LATENCY mode. Two implementations should suffice most
// windows user's needs.
// - PCMWaveOutAudioOutputStream: Based on the waveOut API.
// - WASAPIAudioOutputStream: Based on Core Audio (WASAPI) API.
AudioOutputStream* AudioManagerWin::MakeLowLatencyOutputStream(
    const AudioParameters& params,
    const std::string& device_id,
    const LogCallback& log_callback) {
  DCHECK_EQ(AudioParameters::AUDIO_PCM_LOW_LATENCY, params.format());
  if (params.channels() > kWinMaxChannels)
    return NULL;

  // Pass an empty string to indicate that we want the default device
  // since we consistently only check for an empty string in
  // WASAPIAudioOutputStream.
  bool communications =
      device_id == AudioDeviceDescription::kCommunicationsDeviceId;
  return new WASAPIAudioOutputStream(
      this,
      communications || device_id == AudioDeviceDescription::kDefaultDeviceId
          ? std::string()
          : device_id,
      params, communications ? eCommunications : eConsole);
}

// Factory for the implementations of AudioInputStream for AUDIO_PCM_LINEAR
// mode.
AudioInputStream* AudioManagerWin::MakeLinearInputStream(
    const AudioParameters& params,
    const std::string& device_id,
    const LogCallback& log_callback) {
  DCHECK_EQ(AudioParameters::AUDIO_PCM_LINEAR, params.format());
  return MakeLowLatencyInputStream(params, device_id, log_callback);
}

// Factory for the implementations of AudioInputStream for
// AUDIO_PCM_LOW_LATENCY mode.
AudioInputStream* AudioManagerWin::MakeLowLatencyInputStream(
    const AudioParameters& params,
    const std::string& device_id,
    const LogCallback& log_callback) {
  // Used for both AUDIO_PCM_LOW_LATENCY and AUDIO_PCM_LINEAR.
  DVLOG(1) << "MakeLowLatencyInputStream: " << device_id;
  return new WASAPIAudioInputStream(this, params, device_id, log_callback);
}

std::string AudioManagerWin::GetDefaultInputDeviceID() {
  return CoreAudioUtil::GetDefaultInputDeviceID();
}

std::string AudioManagerWin::GetDefaultOutputDeviceID() {
  return CoreAudioUtil::GetDefaultOutputDeviceID();
}

std::string AudioManagerWin::GetCommunicationsInputDeviceID() {
  return CoreAudioUtil::GetCommunicationsInputDeviceID();
}

std::string AudioManagerWin::GetCommunicationsOutputDeviceID() {
  return CoreAudioUtil::GetCommunicationsOutputDeviceID();
}

AudioParameters AudioManagerWin::GetPreferredOutputStreamParameters(
    const std::string& output_device_id,
    const AudioParameters& input_params) {
  const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess();
  ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO;
  int sample_rate = 48000;
  int buffer_size = kFallbackBufferSize;
  int effects = AudioParameters::NO_EFFECTS;
  int min_buffer_size = 0;
  int max_buffer_size = 0;

  // TODO(henrika): Remove kEnableExclusiveAudio and related code. It doesn't
  // look like it's used.
  if (cmd_line->HasSwitch(switches::kEnableExclusiveAudio)) {
    // TODO(rtoy): tune these values for best possible WebAudio
    // performance. WebRTC works well at 48kHz and a buffer size of 480
    // samples will be used for this case. Note that exclusive mode is
    // experimental. This sample rate will be combined with a buffer size of
    // 256 samples, which corresponds to an output delay of ~5.33ms.
    sample_rate = 48000;
    buffer_size = 256;
    if (input_params.IsValid())
      channel_layout = input_params.channel_layout();
  } else {
    AudioParameters params;
    HRESULT hr = CoreAudioUtil::GetPreferredAudioParameters(
        output_device_id.empty() ? GetDefaultOutputDeviceID()
                                 : output_device_id,
        true, &params);
    if (FAILED(hr)) {
      // This can happen when CoreAudio isn't supported or available
      // (e.g. certain installations of Windows Server 2008 R2).
      // Instead of returning the input_params, we'll return invalid
      // AudioParameters to make sure that an attempt to create this output
      // stream, won't succeed. This behavior is also consistent with
      // GetInputStreamParameters.
      DLOG(ERROR) << "GetPreferredAudioParameters failed: " << std::hex << hr;
      return AudioParameters();
    }

    DCHECK(params.IsValid());

    buffer_size = params.frames_per_buffer();
    channel_layout = params.channel_layout();
    sample_rate = params.sample_rate();
    effects = params.effects();

    AudioParameters::HardwareCapabilities hardware_capabilities =
        params.hardware_capabilities().value_or(
            AudioParameters::HardwareCapabilities());
    min_buffer_size = hardware_capabilities.min_frames_per_buffer;
    max_buffer_size = hardware_capabilities.max_frames_per_buffer;
  }

  if (input_params.IsValid()) {
    // If the user has enabled checking supported channel layouts or we don't
    // have a valid channel layout yet, try to use the input layout.  See bugs
    // http://crbug.com/259165 and http://crbug.com/311906 for more details.
    if (cmd_line->HasSwitch(switches::kTrySupportedChannelLayouts) ||
        channel_layout == CHANNEL_LAYOUT_UNSUPPORTED) {
      // Check if it is possible to open up at the specified input channel
      // layout but avoid checking if the specified layout is the same as the
      // hardware (preferred) layout. We do this extra check to avoid the
      // CoreAudioUtil::IsChannelLayoutSupported() overhead in most cases.
      if (input_params.channel_layout() != channel_layout) {
        // TODO(henrika): Internally, IsChannelLayoutSupported does many of the
        // operations that have already been done such as opening up a client
        // and fetching the WAVEFORMATPCMEX format.  Ideally we should only do
        // that once.  Then here, we can check the layout from the data we
        // already hold.
        if (CoreAudioUtil::IsChannelLayoutSupported(
                output_device_id, eRender, eConsole,
                input_params.channel_layout())) {
          // Open up using the same channel layout as the source if it is
          // supported by the hardware.
          channel_layout = input_params.channel_layout();
          DVLOG(1) << "Hardware channel layout is not used; using same layout"
                   << " as the source instead (" << channel_layout << ")";
        }
      }
    }

    effects |= input_params.effects();

    // Allow non-default buffer sizes if we have a valid min and max.
    if (min_buffer_size > 0 && max_buffer_size > 0) {
      buffer_size =
          std::min(max_buffer_size,
                   std::max(input_params.frames_per_buffer(), min_buffer_size));
    }
  }

  int user_buffer_size = GetUserBufferSize();
  if (user_buffer_size)
    buffer_size = user_buffer_size;

  AudioParameters params(
      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, sample_rate,
      buffer_size,
      AudioParameters::HardwareCapabilities(min_buffer_size, max_buffer_size));
  params.set_effects(effects);
  DCHECK(params.IsValid());
  return params;
}

// static
std::unique_ptr<AudioManager> CreateAudioManager(
    std::unique_ptr<AudioThread> audio_thread,
    AudioLogFactory* audio_log_factory) {
  return std::make_unique<AudioManagerWin>(std::move(audio_thread),
                                           audio_log_factory);
}

}  // namespace media