/*
 *  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.
 */

#ifndef API_AUDIO_AUDIO_FRAME_H_
#define API_AUDIO_AUDIO_FRAME_H_

#include <stdint.h>
#include <stdlib.h>

#include "rtc_base/constructormagic.h"
#include "rtc_base/deprecation.h"
#include "typedefs.h"  // NOLINT(build/include)

namespace webrtc {

/* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
 * allows for adding and subtracting frames while keeping track of the resulting
 * states.
 *
 * Notes
 * - This is a de-facto api, not designed for external use. The AudioFrame class
 *   is in need of overhaul or even replacement, and anyone depending on it
 *   should be prepared for that.
 * - The total number of samples is samples_per_channel_ * num_channels_.
 * - Stereo data is interleaved starting with the left channel.
 */
class AudioFrame {
 public:
  // Using constexpr here causes linker errors unless the variable also has an
  // out-of-class definition, which is impractical in this header-only class.
  // (This makes no sense because it compiles as an enum value, which we most
  // certainly cannot take the address of, just fine.) C++17 introduces inline
  // variables which should allow us to switch to constexpr and keep this a
  // header-only class.
  enum : size_t {
    // Stereo, 32 kHz, 60 ms (2 * 32 * 60)
    kMaxDataSizeSamples = 3840,
    kMaxDataSizeBytes = kMaxDataSizeSamples * sizeof(int16_t),
  };

  enum VADActivity {
    kVadActive = 0,
    kVadPassive = 1,
    kVadUnknown = 2
  };
  enum SpeechType {
    kNormalSpeech = 0,
    kPLC = 1,
    kCNG = 2,
    kPLCCNG = 3,
    kUndefined = 4
  };

  AudioFrame();

  // Resets all members to their default state.
  void Reset();
  // Same as Reset(), but leaves mute state unchanged. Muting a frame requires
  // the buffer to be zeroed on the next call to mutable_data(). Callers
  // intending to write to the buffer immediately after Reset() can instead use
  // ResetWithoutMuting() to skip this wasteful zeroing.
  void ResetWithoutMuting();

  // TODO(solenberg): Remove once downstream users of AudioFrame have updated.
  RTC_DEPRECATED
      void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
                       size_t samples_per_channel, int sample_rate_hz,
                       SpeechType speech_type, VADActivity vad_activity,
                       size_t num_channels = 1) {
    RTC_UNUSED(id);
    UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
                speech_type, vad_activity, num_channels);
  }

  void UpdateFrame(uint32_t timestamp, const int16_t* data,
                   size_t samples_per_channel, int sample_rate_hz,
                   SpeechType speech_type, VADActivity vad_activity,
                   size_t num_channels = 1);

  void CopyFrom(const AudioFrame& src);

  // Sets a wall-time clock timestamp in milliseconds to be used for profiling
  // of time between two points in the audio chain.
  // Example:
  //   t0: UpdateProfileTimeStamp()
  //   t1: ElapsedProfileTimeMs() => t1 - t0 [msec]
  void UpdateProfileTimeStamp();
  // Returns the time difference between now and when UpdateProfileTimeStamp()
  // was last called. Returns -1 if UpdateProfileTimeStamp() has not yet been
  // called.
  int64_t ElapsedProfileTimeMs() const;

  // data() returns a zeroed static buffer if the frame is muted.
  // mutable_frame() always returns a non-static buffer; the first call to
  // mutable_frame() zeros the non-static buffer and marks the frame unmuted.
  const int16_t* data() const;
  int16_t* mutable_data();

  // Prefer to mute frames using AudioFrameOperations::Mute.
  void Mute();
  // Frame is muted by default.
  bool muted() const;

  // These methods are deprecated. Use the functions in
  // webrtc/audio/utility instead. These methods will exists for a
  // short period of time until webrtc clients have updated. See
  // webrtc:6548 for details.
  RTC_DEPRECATED AudioFrame& operator>>=(const int rhs);
  RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs);

  // RTP timestamp of the first sample in the AudioFrame.
  uint32_t timestamp_ = 0;
  // Time since the first frame in milliseconds.
  // -1 represents an uninitialized value.
  int64_t elapsed_time_ms_ = -1;
  // NTP time of the estimated capture time in local timebase in milliseconds.
  // -1 represents an uninitialized value.
  int64_t ntp_time_ms_ = -1;
  size_t samples_per_channel_ = 0;
  int sample_rate_hz_ = 0;
  size_t num_channels_ = 0;
  SpeechType speech_type_ = kUndefined;
  VADActivity vad_activity_ = kVadUnknown;
  // Monotonically increasing timestamp intended for profiling of audio frames.
  // Typically used for measuring elapsed time between two different points in
  // the audio path. No lock is used to save resources and we are thread safe
  // by design. Also, rtc::Optional is not used since it will cause a "complex
  // class/struct needs an explicit out-of-line destructor" build error.
  int64_t profile_timestamp_ms_ = 0;

 private:
  // A permamently zeroed out buffer to represent muted frames. This is a
  // header-only class, so the only way to avoid creating a separate empty
  // buffer per translation unit is to wrap a static in an inline function.
  static const int16_t* empty_data();

  int16_t data_[kMaxDataSizeSamples];
  bool muted_ = true;

  RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
};

}  // namespace webrtc

#endif  // API_AUDIO_AUDIO_FRAME_H_