// Copyright 2013 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/base/audio_buffer.h"

#include <cmath>

#include "base/logging.h"
#include "media/base/audio_bus.h"
#include "media/base/limits.h"
#include "media/base/timestamp_constants.h"

namespace media {

static base::TimeDelta CalculateDuration(int frames, double sample_rate) {
  DCHECK_GT(sample_rate, 0);
  return base::TimeDelta::FromMicroseconds(
      frames * base::Time::kMicrosecondsPerSecond / sample_rate);
}

AudioBuffer::AudioBuffer(SampleFormat sample_format,
                         ChannelLayout channel_layout,
                         int channel_count,
                         int sample_rate,
                         int frame_count,
                         bool create_buffer,
                         const uint8_t* const* data,
                         const base::TimeDelta timestamp)
    : sample_format_(sample_format),
      channel_layout_(channel_layout),
      channel_count_(channel_count),
      sample_rate_(sample_rate),
      adjusted_frame_count_(frame_count),
      trim_start_(0),
      end_of_stream_(!create_buffer && data == NULL && frame_count == 0),
      timestamp_(timestamp),
      duration_(end_of_stream_
                    ? base::TimeDelta()
                    : CalculateDuration(adjusted_frame_count_, sample_rate_)),
      data_size_(0) {
  CHECK_GE(channel_count_, 0);
  CHECK_LE(channel_count_, limits::kMaxChannels);
  CHECK_GE(frame_count, 0);
  DCHECK(channel_layout == CHANNEL_LAYOUT_DISCRETE ||
         ChannelLayoutToChannelCount(channel_layout) == channel_count);

  int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format);
  DCHECK_LE(bytes_per_channel, kChannelAlignment);

  // Empty buffer?
  if (!create_buffer)
    return;

  int data_size_per_channel = frame_count * bytes_per_channel;
  if (IsPlanar(sample_format)) {
    // Planar data, so need to allocate buffer for each channel.
    // Determine per channel data size, taking into account alignment.
    int block_size_per_channel =
        (data_size_per_channel + kChannelAlignment - 1) &
        ~(kChannelAlignment - 1);
    DCHECK_GE(block_size_per_channel, data_size_per_channel);

    // Allocate a contiguous buffer for all the channel data.
    data_size_ = channel_count_ * block_size_per_channel;
    data_.reset(static_cast<uint8_t*>(
        base::AlignedAlloc(data_size_, kChannelAlignment)));
    channel_data_.reserve(channel_count_);

    // Copy each channel's data into the appropriate spot.
    for (int i = 0; i < channel_count_; ++i) {
      channel_data_.push_back(data_.get() + i * block_size_per_channel);
      if (data)
        memcpy(channel_data_[i], data[i], data_size_per_channel);
    }
    return;
  }

  // Remaining formats are interleaved data.
  DCHECK(IsInterleaved(sample_format)) << sample_format_;
  // Allocate our own buffer and copy the supplied data into it. Buffer must
  // contain the data for all channels.
  data_size_ = data_size_per_channel * channel_count_;
  data_.reset(
      static_cast<uint8_t*>(base::AlignedAlloc(data_size_, kChannelAlignment)));
  channel_data_.reserve(1);
  channel_data_.push_back(data_.get());
  if (data)
    memcpy(data_.get(), data[0], data_size_);
}

AudioBuffer::~AudioBuffer() {}

// static
scoped_refptr<AudioBuffer> AudioBuffer::CopyFrom(
    SampleFormat sample_format,
    ChannelLayout channel_layout,
    int channel_count,
    int sample_rate,
    int frame_count,
    const uint8_t* const* data,
    const base::TimeDelta timestamp) {
  // If you hit this CHECK you likely have a bug in a demuxer. Go fix it.
  CHECK_GT(frame_count, 0);  // Otherwise looks like an EOF buffer.
  CHECK(data[0]);
  return make_scoped_refptr(new AudioBuffer(sample_format,
                                            channel_layout,
                                            channel_count,
                                            sample_rate,
                                            frame_count,
                                            true,
                                            data,
                                            timestamp));
}

// static
scoped_refptr<AudioBuffer> AudioBuffer::CreateBuffer(
    SampleFormat sample_format,
    ChannelLayout channel_layout,
    int channel_count,
    int sample_rate,
    int frame_count) {
  CHECK_GT(frame_count, 0);  // Otherwise looks like an EOF buffer.
  return make_scoped_refptr(new AudioBuffer(sample_format,
                                            channel_layout,
                                            channel_count,
                                            sample_rate,
                                            frame_count,
                                            true,
                                            NULL,
                                            kNoTimestamp()));
}

// static
scoped_refptr<AudioBuffer> AudioBuffer::CreateEmptyBuffer(
    ChannelLayout channel_layout,
    int channel_count,
    int sample_rate,
    int frame_count,
    const base::TimeDelta timestamp) {
  CHECK_GT(frame_count, 0);  // Otherwise looks like an EOF buffer.
  // Since data == NULL, format doesn't matter.
  return make_scoped_refptr(new AudioBuffer(kSampleFormatF32,
                                            channel_layout,
                                            channel_count,
                                            sample_rate,
                                            frame_count,
                                            false,
                                            NULL,
                                            timestamp));
}

// static
scoped_refptr<AudioBuffer> AudioBuffer::CreateEOSBuffer() {
  return make_scoped_refptr(new AudioBuffer(kUnknownSampleFormat,
                                            CHANNEL_LAYOUT_NONE,
                                            0,
                                            0,
                                            0,
                                            false,
                                            NULL,
                                            kNoTimestamp()));
}

template <typename Target, typename Dest>
static inline Dest ConvertSample(Target value);

// Convert int16_t values in the range [INT16_MIN, INT16_MAX] to [-1.0, 1.0].
template <>
inline float ConvertSample<int16_t, float>(int16_t value) {
  return value * (value < 0 ? -1.0f / std::numeric_limits<int16_t>::min()
                            : 1.0f / std::numeric_limits<int16_t>::max());
}

// Specializations for int32_t
template <>
inline int32_t ConvertSample<int16_t, int32_t>(int16_t value) {
  return static_cast<int32_t>(value) << 16;
}

template <>
inline int32_t ConvertSample<int32_t, int32_t>(int32_t value) {
  return value;
}

template <>
inline int32_t ConvertSample<float, int32_t>(float value) {
  return static_cast<int32_t>(
      value < 0 ? (-value) * std::numeric_limits<int32_t>::min()
                : value * std::numeric_limits<int32_t>::max());
}

// Specializations for int16_t
template <>
inline int16_t ConvertSample<int16_t, int16_t>(int16_t sample) {
  return sample;
}

template <>
inline int16_t ConvertSample<int32_t, int16_t>(int32_t sample) {
  return sample >> 16;
}

template <>
inline int16_t ConvertSample<float, int16_t>(float sample) {
  return static_cast<int16_t>(
      nearbyint(sample < 0 ? (-sample) * std::numeric_limits<int16_t>::min()
                           : sample * std::numeric_limits<int16_t>::max()));
}

void AudioBuffer::ReadFrames(int frames_to_copy,
                             int source_frame_offset,
                             int dest_frame_offset,
                             AudioBus* dest) {
  // Deinterleave each channel (if necessary) and convert to 32bit
  // floating-point with nominal range -1.0 -> +1.0 (if necessary).

  // |dest| must have the same number of channels, and the number of frames
  // specified must be in range.
  DCHECK(!end_of_stream());
  DCHECK_EQ(dest->channels(), channel_count_);
  DCHECK_LE(source_frame_offset + frames_to_copy, adjusted_frame_count_);
  DCHECK_LE(dest_frame_offset + frames_to_copy, dest->frames());

  // Move the start past any frames that have been trimmed.
  source_frame_offset += trim_start_;

  if (!data_) {
    // Special case for an empty buffer.
    dest->ZeroFramesPartial(dest_frame_offset, frames_to_copy);
    return;
  }

  if (sample_format_ == kSampleFormatPlanarF32) {
    // Format is planar float32. Copy the data from each channel as a block.
    for (int ch = 0; ch < channel_count_; ++ch) {
      const float* source_data =
          reinterpret_cast<const float*>(channel_data_[ch]) +
          source_frame_offset;
      memcpy(dest->channel(ch) + dest_frame_offset,
             source_data,
             sizeof(float) * frames_to_copy);
    }
    return;
  }

  if (sample_format_ == kSampleFormatPlanarS16) {
    // Format is planar signed16. Convert each value into float and insert into
    // output channel data.
    for (int ch = 0; ch < channel_count_; ++ch) {
      const int16_t* source_data =
          reinterpret_cast<const int16_t*>(channel_data_[ch]) +
          source_frame_offset;
      float* dest_data = dest->channel(ch) + dest_frame_offset;
      for (int i = 0; i < frames_to_copy; ++i) {
        dest_data[i] = ConvertSample<int16_t, float>(source_data[i]);
      }
    }
    return;
  }

  if (sample_format_ == kSampleFormatF32) {
    // Format is interleaved float32. Copy the data into each channel.
    const float* source_data = reinterpret_cast<const float*>(data_.get()) +
                               source_frame_offset * channel_count_;
    for (int ch = 0; ch < channel_count_; ++ch) {
      float* dest_data = dest->channel(ch) + dest_frame_offset;
      for (int i = 0, offset = ch; i < frames_to_copy;
           ++i, offset += channel_count_) {
        dest_data[i] = source_data[offset];
      }
    }
    return;
  }

  // Remaining formats are integer interleaved data. Use the deinterleaving code
  // in AudioBus to copy the data.
  DCHECK(
      sample_format_ == kSampleFormatU8 || sample_format_ == kSampleFormatS16 ||
      sample_format_ == kSampleFormatS24 || sample_format_ == kSampleFormatS32);
  int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format_);
  int frame_size = channel_count_ * bytes_per_channel;
  const uint8_t* source_data = data_.get() + source_frame_offset * frame_size;
  dest->FromInterleavedPartial(
      source_data, dest_frame_offset, frames_to_copy, bytes_per_channel);
}

template <class Target, typename Dest>
void InterleaveAndConvert(const std::vector<uint8_t*>& channel_data,
                          size_t frames_to_copy,
                          int trim_start,
                          Dest* dest_data) {
  for (size_t ch = 0; ch < channel_data.size(); ++ch) {
    const Target* source_data =
        reinterpret_cast<const Target*>(channel_data[ch]) + trim_start;
    for (size_t i = 0, offset = ch; i < frames_to_copy;
         ++i, offset += channel_data.size()) {
      dest_data[offset] = ConvertSample<Target, Dest>(source_data[i]);
    }
  }
}

template <typename Dest>
void ReadFramesInterleaved(const std::vector<uint8_t*>& channel_data,
                           int channel_count,
                           SampleFormat sample_format,
                           int frames_to_copy,
                           int trim_start,
                           Dest* dest_data) {
  switch (sample_format) {
    case kSampleFormatU8:
      NOTREACHED();
      break;
    case kSampleFormatS16:
      InterleaveAndConvert<int16_t, Dest>(
          channel_data, frames_to_copy * channel_count, trim_start, dest_data);
      break;
    case kSampleFormatS24:
    case kSampleFormatS32:
      InterleaveAndConvert<int32_t, Dest>(
          channel_data, frames_to_copy * channel_count, trim_start, dest_data);
      break;
    case kSampleFormatF32:
      InterleaveAndConvert<float, Dest>(
          channel_data, frames_to_copy * channel_count, trim_start, dest_data);
      break;
    case kSampleFormatPlanarS16:
      InterleaveAndConvert<int16_t, Dest>(channel_data, frames_to_copy,
                                          trim_start, dest_data);
      break;
    case kSampleFormatPlanarF32:
      InterleaveAndConvert<float, Dest>(channel_data, frames_to_copy,
                                        trim_start, dest_data);
      break;
    case kSampleFormatPlanarS32:
      InterleaveAndConvert<int32_t, Dest>(channel_data, frames_to_copy,
                                          trim_start, dest_data);
      break;
    case kUnknownSampleFormat:
      NOTREACHED();
      break;
  }
}

void AudioBuffer::ReadFramesInterleavedS32(int frames_to_copy,
                                           int32_t* dest_data) {
  DCHECK_LE(frames_to_copy, adjusted_frame_count_);
  ReadFramesInterleaved<int32_t>(channel_data_, channel_count_, sample_format_,
                                 frames_to_copy, trim_start_, dest_data);
}

void AudioBuffer::ReadFramesInterleavedS16(int frames_to_copy,
                                           int16_t* dest_data) {
  DCHECK_LE(frames_to_copy, adjusted_frame_count_);
  ReadFramesInterleaved<int16_t>(channel_data_, channel_count_, sample_format_,
                                 frames_to_copy, trim_start_, dest_data);
}

void AudioBuffer::TrimStart(int frames_to_trim) {
  CHECK_GE(frames_to_trim, 0);
  CHECK_LE(frames_to_trim, adjusted_frame_count_);

  // Adjust the number of frames in this buffer and where the start really is.
  adjusted_frame_count_ -= frames_to_trim;
  trim_start_ += frames_to_trim;

  // Adjust timestamp_ and duration_ to reflect the smaller number of frames.
  const base::TimeDelta old_duration = duration_;
  duration_ = CalculateDuration(adjusted_frame_count_, sample_rate_);
  timestamp_ += old_duration - duration_;
}

void AudioBuffer::TrimEnd(int frames_to_trim) {
  CHECK_GE(frames_to_trim, 0);
  CHECK_LE(frames_to_trim, adjusted_frame_count_);

  // Adjust the number of frames and duration for this buffer.
  adjusted_frame_count_ -= frames_to_trim;
  duration_ = CalculateDuration(adjusted_frame_count_, sample_rate_);
}

void AudioBuffer::TrimRange(int start, int end) {
  CHECK_GE(start, 0);
  CHECK_LE(end, adjusted_frame_count_);

  const int frames_to_trim = end - start;
  CHECK_GE(frames_to_trim, 0);
  CHECK_LE(frames_to_trim, adjusted_frame_count_);

  const int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format_);
  const int frames_to_copy = adjusted_frame_count_ - end;
  if (frames_to_copy > 0) {
    switch (sample_format_) {
      case kSampleFormatPlanarS16:
      case kSampleFormatPlanarF32:
      case kSampleFormatPlanarS32:
        // Planar data must be shifted per channel.
        for (int ch = 0; ch < channel_count_; ++ch) {
          memmove(channel_data_[ch] + (trim_start_ + start) * bytes_per_channel,
                  channel_data_[ch] + (trim_start_ + end) * bytes_per_channel,
                  bytes_per_channel * frames_to_copy);
        }
        break;
      case kSampleFormatU8:
      case kSampleFormatS16:
      case kSampleFormatS24:
      case kSampleFormatS32:
      case kSampleFormatF32: {
        // Interleaved data can be shifted all at once.
        const int frame_size = channel_count_ * bytes_per_channel;
        memmove(channel_data_[0] + (trim_start_ + start) * frame_size,
                channel_data_[0] + (trim_start_ + end) * frame_size,
                frame_size * frames_to_copy);
        break;
      }
      case kUnknownSampleFormat:
        NOTREACHED() << "Invalid sample format!";
    }
  } else {
    CHECK_EQ(frames_to_copy, 0);
  }

  // Trim the leftover data off the end of the buffer and update duration.
  TrimEnd(frames_to_trim);
}

}  // namespace media