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/*
* Copyright (c) 2019 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 "modules/audio_coding/acm2/acm_remixing.h"
#include "rtc_base/checks.h"
namespace webrtc {
void DownMixFrame(const AudioFrame& input, rtc::ArrayView<int16_t> output) {
RTC_DCHECK_EQ(input.num_channels_, 2);
RTC_DCHECK_EQ(output.size(), input.samples_per_channel_);
if (input.muted()) {
std::fill(output.begin(), output.begin() + input.samples_per_channel_, 0);
} else {
const int16_t* const input_data = input.data();
for (size_t n = 0; n < input.samples_per_channel_; ++n) {
output[n] = rtc::dchecked_cast<int16_t>(
(int32_t{input_data[2 * n]} + int32_t{input_data[2 * n + 1]}) >> 1);
}
}
}
void ReMixFrame(const AudioFrame& input,
size_t num_output_channels,
std::vector<int16_t>* output) {
const size_t output_size = num_output_channels * input.samples_per_channel_;
RTC_DCHECK(!(input.num_channels_ == 0 && num_output_channels > 0 &&
input.samples_per_channel_ > 0));
if (output->size() != output_size) {
output->resize(output_size);
}
// For muted frames, fill the frame with zeros.
if (input.muted()) {
std::fill(output->begin(), output->end(), 0);
return;
}
// Ensure that the special case of zero input channels is handled correctly
// (zero samples per channel is already handled correctly in the code below).
if (input.num_channels_ == 0) {
return;
}
const int16_t* const input_data = input.data();
size_t out_index = 0;
// When upmixing is needed and the input is mono copy the left channel
// into the left and right channels, and set any remaining channels to zero.
if (input.num_channels_ == 1 && input.num_channels_ < num_output_channels) {
for (size_t k = 0; k < input.samples_per_channel_; ++k) {
(*output)[out_index++] = input_data[k];
(*output)[out_index++] = input_data[k];
for (size_t j = 2; j < num_output_channels; ++j) {
(*output)[out_index++] = 0;
}
RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
}
RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
return;
}
size_t in_index = 0;
// When upmixing is needed and the output is surround, copy the available
// channels directly, and set the remaining channels to zero.
if (input.num_channels_ < num_output_channels) {
for (size_t k = 0; k < input.samples_per_channel_; ++k) {
for (size_t j = 0; j < input.num_channels_; ++j) {
(*output)[out_index++] = input_data[in_index++];
}
for (size_t j = input.num_channels_; j < num_output_channels; ++j) {
(*output)[out_index++] = 0;
}
RTC_DCHECK_EQ(in_index, (k + 1) * input.num_channels_);
RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
}
RTC_DCHECK_EQ(in_index, input.samples_per_channel_ * input.num_channels_);
RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
return;
}
// When downmixing is needed, and the input is stereo, average the channels.
if (input.num_channels_ == 2) {
for (size_t n = 0; n < input.samples_per_channel_; ++n) {
(*output)[n] = rtc::dchecked_cast<int16_t>(
(int32_t{input_data[2 * n]} + int32_t{input_data[2 * n + 1]}) >> 1);
}
return;
}
// When downmixing is needed, and the input is multichannel, drop the surplus
// channels.
const size_t num_channels_to_drop = input.num_channels_ - num_output_channels;
for (size_t k = 0; k < input.samples_per_channel_; ++k) {
for (size_t j = 0; j < num_output_channels; ++j) {
(*output)[out_index++] = input_data[in_index++];
}
in_index += num_channels_to_drop;
}
}
} // namespace webrtc
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