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/*
* 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 "modules/audio_coding/acm2/acm_resampler.h"
#include <array>
#include <cstdint>
#include "api/audio/audio_frame.h"
#include "api/audio/audio_view.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace acm2 {
ResamplerHelper::ResamplerHelper() {
ClearSamples(last_audio_buffer_);
}
bool ResamplerHelper::MaybeResample(int desired_sample_rate_hz,
AudioFrame* audio_frame) {
const int current_sample_rate_hz = audio_frame->sample_rate_hz_;
RTC_DCHECK_NE(current_sample_rate_hz, 0);
RTC_DCHECK_GT(desired_sample_rate_hz, 0);
// Update if resampling is required.
// TODO(tommi): `desired_sample_rate_hz` should never be -1.
// Remove the first check.
const bool need_resampling =
(desired_sample_rate_hz != -1) &&
(current_sample_rate_hz != desired_sample_rate_hz);
if (need_resampling && !resampled_last_output_frame_) {
// Prime the resampler with the last frame.
InterleavedView<const int16_t> src(last_audio_buffer_.data(),
audio_frame->samples_per_channel(),
audio_frame->num_channels());
std::array<int16_t, AudioFrame::kMaxDataSizeSamples> temp_output;
InterleavedView<int16_t> dst(
temp_output.data(),
SampleRateToDefaultChannelSize(desired_sample_rate_hz),
audio_frame->num_channels_);
resampler_.Resample(src, dst);
}
// TODO(bugs.webrtc.org/3923) Glitches in the output may appear if the output
// rate from NetEq changes.
if (need_resampling) {
// Grab the source view of the current layout before changing properties.
InterleavedView<const int16_t> src = audio_frame->data_view();
audio_frame->SetSampleRateAndChannelSize(desired_sample_rate_hz);
InterleavedView<int16_t> dst = audio_frame->mutable_data(
audio_frame->samples_per_channel(), audio_frame->num_channels());
// TODO(tommi): Don't resample muted audio frames.
resampler_.Resample(src, dst);
resampled_last_output_frame_ = true;
} else {
resampled_last_output_frame_ = false;
// We might end up here ONLY if codec is changed.
}
// Store current audio in `last_audio_buffer_` for next time.
InterleavedView<int16_t> dst(last_audio_buffer_.data(),
audio_frame->samples_per_channel(),
audio_frame->num_channels());
CopySamples(dst, audio_frame->data_view());
return true;
}
} // namespace acm2
} // namespace webrtc
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