/*
 *  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 AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
#define AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_

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

#include "absl/base/attributes.h"
#include "api/array_view.h"
#include "api/audio/audio_frame.h"

namespace webrtc {

// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
// Change reference parameters to pointers. Consider using a namespace rather
// than a class.
class AudioFrameOperations {
 public:
  // Add samples in `frame_to_add` with samples in `result_frame`
  // putting the results in `results_frame`.  The fields
  // `vad_activity_` and `speech_type_` of the result frame are
  // updated. If `result_frame` is empty (`samples_per_channel_`==0),
  // the samples in `frame_to_add` are added to it.  The number of
  // channels and number of samples per channel must match except when
  // `result_frame` is empty.
  static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);

  // Downmixes 4 channels `src_audio` to stereo `dst_audio`. This is an in-place
  // operation, meaning `src_audio` and `dst_audio` may point to the same
  // buffer.
  static void QuadToStereo(rtc::ArrayView<const int16_t> src_audio,
                           size_t samples_per_channel,
                           rtc::ArrayView<int16_t> dst_audio);

  // `frame.num_channels_` will be updated. This version checks that
  // `num_channels_` is 4 channels.
  static int QuadToStereo(AudioFrame* frame);

  // Downmixes `src_channels` `src_audio` to `dst_channels` `dst_audio`.
  // This is an in-place operation, meaning `src_audio` and `dst_audio`
  // may point to the same buffer. Supported channel combinations are
  // Stereo to Mono, Quad to Mono, and Quad to Stereo.
  static void DownmixChannels(rtc::ArrayView<const int16_t> src_audio,
                              size_t src_channels,
                              size_t samples_per_channel,
                              size_t dst_channels,
                              rtc::ArrayView<int16_t> dst_audio);

  // `frame.num_channels_` will be updated. This version checks that
  // `num_channels_` and `dst_channels` are valid and performs relevant downmix.
  // Supported channel combinations are N channels to Mono, and Quad to Stereo.
  static void DownmixChannels(size_t dst_channels, AudioFrame* frame);

  // `frame.num_channels_` will be updated. This version checks that
  // `num_channels_` and `dst_channels` are valid and performs relevant
  // downmix. Supported channel combinations are Mono to N
  // channels. The single channel is replicated.
  static void UpmixChannels(size_t target_number_of_channels,
                            AudioFrame* frame);

  // Swap the left and right channels of `frame`. Fails silently if `frame` is
  // not stereo.
  static void SwapStereoChannels(AudioFrame* frame);

  // Conditionally zero out contents of `frame` for implementing audio mute:
  //  `previous_frame_muted` &&  `current_frame_muted` - Zero out whole frame.
  //  `previous_frame_muted` && !`current_frame_muted` - Fade-in at frame start.
  // !`previous_frame_muted` &&  `current_frame_muted` - Fade-out at frame end.
  // !`previous_frame_muted` && !`current_frame_muted` - Leave frame untouched.
  static void Mute(AudioFrame* frame,
                   bool previous_frame_muted,
                   bool current_frame_muted);

  // Zero out contents of frame.
  static void Mute(AudioFrame* frame);

  // Halve samples in `frame`.
  static void ApplyHalfGain(AudioFrame* frame);

  static int Scale(float left, float right, AudioFrame* frame);

  static int ScaleWithSat(float scale, AudioFrame* frame);
};

}  // namespace webrtc

#endif  // AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_