/*
 *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "modules/audio_device/win/core_audio_output_win.h"

#include <memory>

#include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/fine_audio_buffer.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"

using Microsoft::WRL::ComPtr;

namespace webrtc {
namespace webrtc_win {

CoreAudioOutput::CoreAudioOutput(bool automatic_restart)
    : CoreAudioBase(
          CoreAudioBase::Direction::kOutput,
          automatic_restart,
          [this](uint64_t freq) { return OnDataCallback(freq); },
          [this](ErrorType err) { return OnErrorCallback(err); }) {
  RTC_DLOG(INFO) << __FUNCTION__;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  thread_checker_audio_.Detach();
}

CoreAudioOutput::~CoreAudioOutput() {
  RTC_DLOG(INFO) << __FUNCTION__;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  Terminate();
}

int CoreAudioOutput::Init() {
  RTC_DLOG(INFO) << __FUNCTION__;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  return 0;
}

int CoreAudioOutput::Terminate() {
  RTC_DLOG(INFO) << __FUNCTION__;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  StopPlayout();
  return 0;
}

int CoreAudioOutput::NumDevices() const {
  RTC_DCHECK_RUN_ON(&thread_checker_);
  return core_audio_utility::NumberOfActiveDevices(eRender);
}

int CoreAudioOutput::SetDevice(int index) {
  RTC_DLOG(INFO) << __FUNCTION__ << ": " << index;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  return CoreAudioBase::SetDevice(index);
}

int CoreAudioOutput::SetDevice(AudioDeviceModule::WindowsDeviceType device) {
  RTC_DLOG(INFO) << __FUNCTION__ << ": " << device;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  return SetDevice((device == AudioDeviceModule::kDefaultDevice) ? 0 : 1);
}

int CoreAudioOutput::DeviceName(int index,
                                std::string* name,
                                std::string* guid) {
  RTC_DLOG(INFO) << __FUNCTION__ << ": " << index;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  RTC_DCHECK(name);
  return CoreAudioBase::DeviceName(index, name, guid);
}

void CoreAudioOutput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
  RTC_DLOG(INFO) << __FUNCTION__;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  audio_device_buffer_ = audio_buffer;
}

bool CoreAudioOutput::PlayoutIsInitialized() const {
  RTC_DLOG(INFO) << __FUNCTION__;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  return initialized_;
}

int CoreAudioOutput::InitPlayout() {
  RTC_DLOG(INFO) << __FUNCTION__ << ": " << IsRestarting();
  RTC_DCHECK(!initialized_);
  RTC_DCHECK(!Playing());
  RTC_DCHECK(!audio_render_client_);

  // Creates an IAudioClient instance and stores the valid interface pointer in
  // |audio_client3_|, |audio_client2_|, or |audio_client_| depending on
  // platform support. The base class will use optimal output parameters and do
  // an event driven shared mode initialization. The utilized format will be
  // stored in |format_| and can be used for configuration and allocation of
  // audio buffers.
  if (!CoreAudioBase::Init()) {
    return -1;
  }
  RTC_DCHECK(audio_client_);

  // Configure the playout side of the audio device buffer using |format_|
  // after a trivial sanity check of the format structure.
  RTC_DCHECK(audio_device_buffer_);
  WAVEFORMATEX* format = &format_.Format;
  RTC_DCHECK_EQ(format->wFormatTag, WAVE_FORMAT_EXTENSIBLE);
  audio_device_buffer_->SetPlayoutSampleRate(format->nSamplesPerSec);
  audio_device_buffer_->SetPlayoutChannels(format->nChannels);

  // Create a modified audio buffer class which allows us to ask for any number
  // of samples (and not only multiple of 10ms) to match the optimal
  // buffer size per callback used by Core Audio.
  // TODO(henrika): can we share one FineAudioBuffer with the input side?
  fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);

  // Create an IAudioRenderClient for an initialized IAudioClient.
  // The IAudioRenderClient interface enables us to write output data to
  // a rendering endpoint buffer.
  ComPtr<IAudioRenderClient> audio_render_client =
      core_audio_utility::CreateRenderClient(audio_client_.Get());
  if (!audio_render_client.Get()) {
    return -1;
  }

  ComPtr<IAudioClock> audio_clock =
      core_audio_utility::CreateAudioClock(audio_client_.Get());
  if (!audio_clock.Get()) {
    return -1;
  }

  // Store valid COM interfaces.
  audio_render_client_ = audio_render_client;
  audio_clock_ = audio_clock;

  initialized_ = true;
  return 0;
}

int CoreAudioOutput::StartPlayout() {
  RTC_DLOG(INFO) << __FUNCTION__ << ": " << IsRestarting();
  RTC_DCHECK(!Playing());
  RTC_DCHECK(fine_audio_buffer_);
  RTC_DCHECK(audio_device_buffer_);
  if (!initialized_) {
    RTC_DLOG(LS_WARNING)
        << "Playout can not start since InitPlayout must succeed first";
  }

  fine_audio_buffer_->ResetPlayout();
  if (!IsRestarting()) {
    audio_device_buffer_->StartPlayout();
  }

  if (!core_audio_utility::FillRenderEndpointBufferWithSilence(
          audio_client_.Get(), audio_render_client_.Get())) {
    RTC_LOG(LS_WARNING) << "Failed to prepare output endpoint with silence";
  }

  num_frames_written_ = endpoint_buffer_size_frames_;

  if (!Start()) {
    return -1;
  }

  is_active_ = true;
  return 0;
}

int CoreAudioOutput::StopPlayout() {
  RTC_DLOG(INFO) << __FUNCTION__ << ": " << IsRestarting();
  if (!initialized_) {
    return 0;
  }

  // Release resources allocated in InitPlayout() and then return if this
  // method is called without any active output audio.
  if (!Playing()) {
    RTC_DLOG(WARNING) << "No output stream is active";
    ReleaseCOMObjects();
    initialized_ = false;
    return 0;
  }

  if (!Stop()) {
    RTC_LOG(LS_ERROR) << "StopPlayout failed";
    return -1;
  }

  if (!IsRestarting()) {
    RTC_DCHECK(audio_device_buffer_);
    audio_device_buffer_->StopPlayout();
  }

  // Release all allocated resources to allow for a restart without
  // intermediate destruction.
  ReleaseCOMObjects();

  initialized_ = false;
  is_active_ = false;
  return 0;
}

bool CoreAudioOutput::Playing() {
  RTC_DLOG(INFO) << __FUNCTION__ << ": " << is_active_;
  return is_active_;
}

// TODO(henrika): finalize support of audio session volume control. As is, we
// are not compatible with the old ADM implementation since it allows accessing
// the volume control with any active audio output stream.
int CoreAudioOutput::VolumeIsAvailable(bool* available) {
  RTC_DLOG(INFO) << __FUNCTION__;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  return IsVolumeControlAvailable(available) ? 0 : -1;
}

// Triggers the restart sequence. Only used for testing purposes to emulate
// a real event where e.g. an active output device is removed.
int CoreAudioOutput::RestartPlayout() {
  RTC_DLOG(INFO) << __FUNCTION__;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  if (!Playing()) {
    return 0;
  }
  if (!Restart()) {
    RTC_LOG(LS_ERROR) << "RestartPlayout failed";
    return -1;
  }
  return 0;
}

bool CoreAudioOutput::Restarting() const {
  RTC_DLOG(INFO) << __FUNCTION__;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  return IsRestarting();
}

int CoreAudioOutput::SetSampleRate(uint32_t sample_rate) {
  RTC_DLOG(INFO) << __FUNCTION__;
  RTC_DCHECK_RUN_ON(&thread_checker_);
  sample_rate_ = sample_rate;
  return 0;
}

void CoreAudioOutput::ReleaseCOMObjects() {
  RTC_DLOG(INFO) << __FUNCTION__;
  CoreAudioBase::ReleaseCOMObjects();
  if (audio_render_client_.Get()) {
    audio_render_client_.Reset();
  }
}

bool CoreAudioOutput::OnErrorCallback(ErrorType error) {
  RTC_DLOG(INFO) << __FUNCTION__ << ": " << as_integer(error);
  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
  if (!initialized_ || !Playing()) {
    return true;
  }

  if (error == CoreAudioBase::ErrorType::kStreamDisconnected) {
    HandleStreamDisconnected();
  } else {
    RTC_DLOG(WARNING) << "Unsupported error type";
  }
  return true;
}

bool CoreAudioOutput::OnDataCallback(uint64_t device_frequency) {
  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
  if (num_data_callbacks_ == 0) {
    RTC_LOG(INFO) << "--- Output audio stream is alive ---";
  }
  // Get the padding value which indicates the amount of valid unread data that
  // the endpoint buffer currently contains.
  UINT32 num_unread_frames = 0;
  _com_error error = audio_client_->GetCurrentPadding(&num_unread_frames);
  if (error.Error() == AUDCLNT_E_DEVICE_INVALIDATED) {
    // Avoid breaking the thread loop implicitly by returning false and return
    // true instead for AUDCLNT_E_DEVICE_INVALIDATED even it is a valid error
    // message. We will use notifications about device changes instead to stop
    // data callbacks and attempt to restart streaming .
    RTC_DLOG(LS_ERROR) << "AUDCLNT_E_DEVICE_INVALIDATED";
    return true;
  }
  if (FAILED(error.Error())) {
    RTC_LOG(LS_ERROR) << "IAudioClient::GetCurrentPadding failed: "
                      << core_audio_utility::ErrorToString(error);
    return false;
  }

  // Contains how much new data we can write to the buffer without the risk of
  // overwriting previously written data that the audio engine has not yet read
  // from the buffer. I.e., it is the maximum buffer size we can request when
  // calling IAudioRenderClient::GetBuffer().
  UINT32 num_requested_frames =
      endpoint_buffer_size_frames_ - num_unread_frames;
  if (num_requested_frames == 0) {
    RTC_DLOG(LS_WARNING)
        << "Audio thread is signaled but no new audio samples are needed";
    return true;
  }

  // Request all available space in the rendering endpoint buffer into which the
  // client can later write an audio packet.
  uint8_t* audio_data;
  error = audio_render_client_->GetBuffer(num_requested_frames, &audio_data);
  if (FAILED(error.Error())) {
    RTC_LOG(LS_ERROR) << "IAudioRenderClient::GetBuffer failed: "
                      << core_audio_utility::ErrorToString(error);
    return false;
  }

  // Update output delay estimate but only about once per second to save
  // resources. The estimate is usually stable.
  if (num_data_callbacks_ % 100 == 0) {
    // TODO(henrika): note that FineAudioBuffer adds latency as well.
    latency_ms_ = EstimateOutputLatencyMillis(device_frequency);
    if (num_data_callbacks_ % 500 == 0) {
      RTC_DLOG(INFO) << "latency: " << latency_ms_;
    }
  }

  // Get audio data from WebRTC and write it to the allocated buffer in
  // |audio_data|. The playout latency is not updated for each callback.
  fine_audio_buffer_->GetPlayoutData(
      rtc::MakeArrayView(reinterpret_cast<int16_t*>(audio_data),
                         num_requested_frames * format_.Format.nChannels),
      latency_ms_);

  // Release the buffer space acquired in IAudioRenderClient::GetBuffer.
  error = audio_render_client_->ReleaseBuffer(num_requested_frames, 0);
  if (FAILED(error.Error())) {
    RTC_LOG(LS_ERROR) << "IAudioRenderClient::ReleaseBuffer failed: "
                      << core_audio_utility::ErrorToString(error);
    return false;
  }

  num_frames_written_ += num_requested_frames;
  ++num_data_callbacks_;

  return true;
}

// TODO(henrika): IAudioClock2::GetDevicePosition could perhaps be used here
// instead. Tried it once, but it crashed for capture devices.
int CoreAudioOutput::EstimateOutputLatencyMillis(uint64_t device_frequency) {
  UINT64 position = 0;
  UINT64 qpc_position = 0;
  int delay_ms = 0;
  // Get the device position through output parameter |position|. This is the
  // stream position of the sample that is currently playing through the
  // speakers.
  _com_error error = audio_clock_->GetPosition(&position, &qpc_position);
  if (error.Error() == S_OK) {
    // Number of frames already played out through the speaker.
    const uint64_t num_played_out_frames =
        format_.Format.nSamplesPerSec * position / device_frequency;

    // Number of frames that have been written to the buffer but not yet
    // played out corresponding to the estimated latency measured in number
    // of audio frames.
    const uint64_t delay_frames = num_frames_written_ - num_played_out_frames;

    // Convert latency in number of frames into milliseconds.
    webrtc::TimeDelta delay =
        webrtc::TimeDelta::us(delay_frames * rtc::kNumMicrosecsPerSec /
                              format_.Format.nSamplesPerSec);
    delay_ms = delay.ms();
  }
  return delay_ms;
}

// Called from OnErrorCallback() when error type is kStreamDisconnected.
// Note that this method is called on the audio thread and the internal restart
// sequence is also executed on that same thread. The audio thread is therefore
// not stopped during restart. Such a scheme also makes the restart process less
// complex.
// Note that, none of the called methods are thread checked since they can also
// be called on the main thread. Thread checkers are instead added on one layer
// above (in audio_device_module.cc) which ensures that the public API is thread
// safe.
// TODO(henrika): add more details.
bool CoreAudioOutput::HandleStreamDisconnected() {
  RTC_DLOG(INFO) << "<<<--- " << __FUNCTION__;
  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
  RTC_DCHECK(automatic_restart());

  if (StopPlayout() != 0) {
    return false;
  }

  if (!SwitchDeviceIfNeeded()) {
    return false;
  }

  if (InitPlayout() != 0) {
    return false;
  }
  if (StartPlayout() != 0) {
    return false;
  }

  RTC_DLOG(INFO) << __FUNCTION__ << " --->>>";
  return true;
}

}  // namespace webrtc_win

}  // namespace webrtc