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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 <string.h>
#include "api/audio/audio_frame.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace acm2 {
ACMResampler::ACMResampler() {}
ACMResampler::~ACMResampler() {}
int ACMResampler::Resample10Msec(const int16_t* in_audio,
int in_freq_hz,
int out_freq_hz,
size_t num_audio_channels,
size_t out_capacity_samples,
int16_t* out_audio) {
InterleavedView<const int16_t> src(
in_audio, SampleRateToDefaultChannelSize(in_freq_hz), num_audio_channels);
InterleavedView<int16_t> dst(out_audio,
SampleRateToDefaultChannelSize(out_freq_hz),
num_audio_channels);
RTC_DCHECK_GE(out_capacity_samples, dst.size());
if (in_freq_hz == out_freq_hz) {
if (out_capacity_samples < src.data().size()) {
RTC_DCHECK_NOTREACHED();
return -1;
}
CopySamples(dst, src);
RTC_DCHECK_EQ(dst.samples_per_channel(), src.samples_per_channel());
return static_cast<int>(dst.samples_per_channel());
}
int out_length = resampler_.Resample(src, dst);
if (out_length == -1) {
RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << src.data().size()
<< ", " << out_audio << ", " << out_capacity_samples
<< ") failed.";
return -1;
}
RTC_DCHECK_EQ(out_length, dst.size());
RTC_DCHECK_EQ(out_length / num_audio_channels, dst.samples_per_channel());
return static_cast<int>(dst.samples_per_channel());
}
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);
// Update if resampling is required.
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.
int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
int samples_per_channel_int = resampler_.Resample10Msec(
last_audio_buffer_.data(), current_sample_rate_hz,
desired_sample_rate_hz, audio_frame->num_channels_,
AudioFrame::kMaxDataSizeSamples, temp_output);
if (samples_per_channel_int < 0) {
RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - "
"Resampling last_audio_buffer_ failed.";
return false;
}
}
// TODO(bugs.webrtc.org/3923) Glitches in the output may appear if the output
// rate from NetEq changes.
if (need_resampling) {
// TODO(yujo): handle this more efficiently for muted frames.
int samples_per_channel_int = resampler_.Resample10Msec(
audio_frame->data(), current_sample_rate_hz, desired_sample_rate_hz,
audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
audio_frame->mutable_data());
if (samples_per_channel_int < 0) {
RTC_LOG(LS_ERROR)
<< "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
return false;
}
audio_frame->samples_per_channel_ =
static_cast<size_t>(samples_per_channel_int);
audio_frame->sample_rate_hz_ = desired_sample_rate_hz;
RTC_DCHECK_EQ(audio_frame->sample_rate_hz_,
dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
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.
// TODO: b/335805780 - Use CopySamples().
memcpy(last_audio_buffer_.data(), audio_frame->data(),
sizeof(int16_t) * audio_frame->samples_per_channel_ *
audio_frame->num_channels_);
return true;
}
} // namespace acm2
} // namespace webrtc
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