/*
 *  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.
 */

#include "audio/remix_resample.h"

#include <array>

#include "api/audio/audio_frame.h"
#include "audio/utility/audio_frame_operations.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"

namespace webrtc {
namespace voe {

void RemixAndResample(const AudioFrame& src_frame,
                      PushResampler<int16_t>* resampler,
                      AudioFrame* dst_frame) {
  RemixAndResample(src_frame.data(), src_frame.samples_per_channel_,
                   src_frame.num_channels_, src_frame.sample_rate_hz_,
                   resampler, dst_frame);
  dst_frame->timestamp_ = src_frame.timestamp_;
  dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
  dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
  dst_frame->packet_infos_ = src_frame.packet_infos_;
}

// TODO: b/335805780 - Accept ArrayView.
void RemixAndResample(const int16_t* src_data,
                      size_t samples_per_channel,
                      size_t num_channels,
                      int sample_rate_hz,
                      PushResampler<int16_t>* resampler,
                      AudioFrame* dst_frame) {
  const int16_t* audio_ptr = src_data;
  size_t audio_ptr_num_channels = num_channels;
  std::array<int16_t, AudioFrame::kMaxDataSizeSamples> downmixed_audio;

  // Downmix before resampling.
  if (num_channels > dst_frame->num_channels_) {
    RTC_DCHECK(num_channels == 2 || num_channels == 4)
        << "num_channels: " << num_channels;
    RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2)
        << "dst_frame->num_channels_: " << dst_frame->num_channels_;

    AudioFrameOperations::DownmixChannels(
        InterleavedView<const int16_t>(src_data, samples_per_channel,
                                       num_channels),
        InterleavedView<int16_t>(&downmixed_audio[0], samples_per_channel,
                                 dst_frame->num_channels_));
    audio_ptr = downmixed_audio.data();
    audio_ptr_num_channels = dst_frame->num_channels_;
  }

  if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
                                    audio_ptr_num_channels) == -1) {
    RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = "
                << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
                << dst_frame->sample_rate_hz_
                << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
  }

  // TODO(yujo): for muted input frames, don't resample. Either 1) allow
  // resampler to return output length without doing the resample, so we know
  // how much to zero here; or 2) make resampler accept a hint that the input is
  // zeroed.

  // Ensure the `samples_per_channel_` member is set correctly based on the
  // destination sample rate, number of channels and assumed 10ms buffer size.
  // TODO(tommi): Could we rather assume that this has been done by the caller?
  dst_frame->SetSampleRateAndChannelSize(dst_frame->sample_rate_hz_);

  InterleavedView<const int16_t> src_view(audio_ptr, samples_per_channel,
                                          audio_ptr_num_channels);
  // Stash away the originally requested number of channels. Then provide
  // `dst_frame` as a target buffer with the same number of channels as the
  // source.
  auto original_dst_number_of_channels = dst_frame->num_channels_;
  int out_length = resampler->Resample(
      src_view, dst_frame->mutable_data(dst_frame->samples_per_channel_,
                                        src_view.num_channels()));
  RTC_CHECK_NE(out_length, -1) << "Resample failed: audio_ptr = " << audio_ptr
                               << ", src_length = " << src_view.data().size();

  RTC_DCHECK_EQ(dst_frame->samples_per_channel(),
                out_length / audio_ptr_num_channels);

  // Upmix after resampling.
  if (num_channels == 1 && original_dst_number_of_channels == 2) {
    // The audio in dst_frame really is mono at this point; MonoToStereo will
    // set this back to stereo.
    RTC_DCHECK_EQ(dst_frame->num_channels_, 1);
    AudioFrameOperations::UpmixChannels(2, dst_frame);
  }
}

}  // namespace voe
}  // namespace webrtc