/*
 *  Copyright (c) 2012 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.
 */

#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_

#include <stddef.h>
#include <stdint.h>

#include <atomic>

#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_factory.h"
#include "modules/audio_device/include/audio_device_defines.h"
#include "rtc_base/buffer.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/timestamp_aligner.h"

namespace webrtc {

// Delta times between two successive playout callbacks are limited to this
// value before added to an internal array.
const size_t kMaxDeltaTimeInMs = 500;
// TODO(henrika): remove when no longer used by external client.
const size_t kMaxBufferSizeBytes = 3840;  // 10ms in stereo @ 96kHz

class AudioDeviceBuffer {
 public:
  enum LogState {
    LOG_START = 0,
    LOG_STOP,
    LOG_ACTIVE,
  };

  struct Stats {
    void ResetRecStats() {
      rec_callbacks = 0;
      rec_samples = 0;
      max_rec_level = 0;
    }

    void ResetPlayStats() {
      play_callbacks = 0;
      play_samples = 0;
      max_play_level = 0;
    }

    // Total number of recording callbacks where the source provides 10ms audio
    // data each time.
    uint64_t rec_callbacks = 0;

    // Total number of playback callbacks where the sink asks for 10ms audio
    // data each time.
    uint64_t play_callbacks = 0;

    // Total number of recorded audio samples.
    uint64_t rec_samples = 0;

    // Total number of played audio samples.
    uint64_t play_samples = 0;

    // Contains max level (max(abs(x))) of recorded audio packets over the last
    // 10 seconds where a new measurement is done twice per second. The level
    // is reset to zero at each call to LogStats().
    int16_t max_rec_level = 0;

    // Contains max level of recorded audio packets over the last 10 seconds
    // where a new measurement is done twice per second.
    int16_t max_play_level = 0;
  };

  // If `create_detached` is true, the created buffer can be used on another
  // thread compared to the one on which it was created. It's useful for
  // testing.
  explicit AudioDeviceBuffer(TaskQueueFactory* task_queue_factory,
                             bool create_detached = false);
  virtual ~AudioDeviceBuffer();

  int32_t RegisterAudioCallback(AudioTransport* audio_callback);

  void StartPlayout();
  void StartRecording();
  void StopPlayout();
  void StopRecording();

  int32_t SetRecordingSampleRate(uint32_t fsHz);
  int32_t SetPlayoutSampleRate(uint32_t fsHz);
  uint32_t RecordingSampleRate() const;
  uint32_t PlayoutSampleRate() const;

  int32_t SetRecordingChannels(size_t channels);
  int32_t SetPlayoutChannels(size_t channels);
  size_t RecordingChannels() const;
  size_t PlayoutChannels() const;

  // TODO(bugs.webrtc.org/13621) Deprecate this function
  virtual int32_t SetRecordedBuffer(const void* audio_buffer,
                                    size_t samples_per_channel);

  virtual int32_t SetRecordedBuffer(
      const void* audio_buffer,
      size_t samples_per_channel,
      absl::optional<int64_t> capture_timestamp_ns);
  virtual void SetVQEData(int play_delay_ms, int rec_delay_ms);
  virtual int32_t DeliverRecordedData();
  uint32_t NewMicLevel() const;

  virtual int32_t RequestPlayoutData(size_t samples_per_channel);
  virtual int32_t GetPlayoutData(void* audio_buffer);

  int32_t SetTypingStatus(bool typing_status);

 private:
  // Starts/stops periodic logging of audio stats.
  void StartPeriodicLogging();
  void StopPeriodicLogging();

  // Called periodically on the internal thread created by the TaskQueue.
  // Updates some stats but dooes it on the task queue to ensure that access of
  // members is serialized hence avoiding usage of locks.
  // state = LOG_START => members are initialized and the timer starts.
  // state = LOG_STOP => no logs are printed and the timer stops.
  // state = LOG_ACTIVE => logs are printed and the timer is kept alive.
  void LogStats(LogState state);

  // Updates counters in each play/record callback. These counters are later
  // (periodically) read by LogStats() using a lock.
  void UpdateRecStats(int16_t max_abs, size_t samples_per_channel);
  void UpdatePlayStats(int16_t max_abs, size_t samples_per_channel);

  // Clears all members tracking stats for recording and playout.
  // These methods both run on the task queue.
  void ResetRecStats();
  void ResetPlayStats();

  // This object lives on the main (creating) thread and most methods are
  // called on that same thread. When audio has started some methods will be
  // called on either a native audio thread for playout or a native thread for
  // recording. Some members are not annotated since they are "protected by
  // design" and adding e.g. a race checker can cause failures for very few
  // edge cases and it is IMHO not worth the risk to use them in this class.
  // TODO(henrika): see if it is possible to refactor and annotate all members.

  // Main thread on which this object is created.
  SequenceChecker main_thread_checker_;

  Mutex lock_;

  // Task queue used to invoke LogStats() periodically. Tasks are executed on a
  // worker thread but it does not necessarily have to be the same thread for
  // each task.
  rtc::TaskQueue task_queue_;

  // Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
  // and it must outlive this object. It is not possible to change this member
  // while any media is active. It is possible to start media without calling
  // RegisterAudioCallback() but that will lead to ignored audio callbacks in
  // both directions where native audio will be active but no audio samples will
  // be transported.
  AudioTransport* audio_transport_cb_;

  // Sample rate in Hertz. Accessed atomically.
  std::atomic<uint32_t> rec_sample_rate_;
  std::atomic<uint32_t> play_sample_rate_;

  // Number of audio channels. Accessed atomically.
  std::atomic<size_t> rec_channels_;
  std::atomic<size_t> play_channels_;

  // Keeps track of if playout/recording are active or not. A combination
  // of these states are used to determine when to start and stop the timer.
  // Only used on the creating thread and not used to control any media flow.
  bool playing_ RTC_GUARDED_BY(main_thread_checker_);
  bool recording_ RTC_GUARDED_BY(main_thread_checker_);

  // Buffer used for audio samples to be played out. Size can be changed
  // dynamically. The 16-bit samples are interleaved, hence the size is
  // proportional to the number of channels.
  rtc::BufferT<int16_t> play_buffer_;

  // Byte buffer used for recorded audio samples. Size can be changed
  // dynamically.
  rtc::BufferT<int16_t> rec_buffer_;

  // Contains true of a key-press has been detected.
  bool typing_status_;

  // Delay values used by the AEC.
  int play_delay_ms_;
  int rec_delay_ms_;

  // Capture timestamp.
  absl::optional<int64_t> capture_timestamp_ns_;
  // The last time the Timestamp Aligner was used to estimate clock offset
  // between system clock and capture time from audio.
  // This is used to prevent estimating the clock offset too often.
  absl::optional<int64_t> align_offsync_estimation_time_;

  // Counts number of times LogStats() has been called.
  size_t num_stat_reports_ RTC_GUARDED_BY(task_queue_);

  // Time stamp of last timer task (drives logging).
  int64_t last_timer_task_time_ RTC_GUARDED_BY(task_queue_);

  // Counts number of audio callbacks modulo 50 to create a signal when
  // a new storage of audio stats shall be done.
  int16_t rec_stat_count_;
  int16_t play_stat_count_;

  // Time stamps of when playout and recording starts.
  int64_t play_start_time_ RTC_GUARDED_BY(main_thread_checker_);
  int64_t rec_start_time_ RTC_GUARDED_BY(main_thread_checker_);

  // Contains counters for playout and recording statistics.
  Stats stats_ RTC_GUARDED_BY(lock_);

  // Stores current stats at each timer task. Used to calculate differences
  // between two successive timer events.
  Stats last_stats_ RTC_GUARDED_BY(task_queue_);

  // Set to true at construction and modified to false as soon as one audio-
  // level estimate larger than zero is detected.
  bool only_silence_recorded_;

  // Set to true when logging of audio stats is enabled for the first time in
  // StartPeriodicLogging() and set to false by StopPeriodicLogging().
  // Setting this member to false prevents (possiby invalid) log messages from
  // being printed in the LogStats() task.
  bool log_stats_ RTC_GUARDED_BY(task_queue_);

  // Used for converting capture timestaps (received from AudioRecordThread
  // via AudioRecordJni::DataIsRecorded) to RTC clock.
  rtc::TimestampAligner timestamp_aligner_;

// Should *never* be defined in production builds. Only used for testing.
// When defined, the output signal will be replaced by a sinus tone at 440Hz.
#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
  double phase_;
#endif
};

}  // namespace webrtc

#endif  // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_