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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 "webrtc/voice_engine/utility.h"
#include "webrtc/base/logging.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/voice_engine/voice_engine_defines.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_;
}
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;
int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
// Downmix before resampling.
if (num_channels == 2 && dst_frame->num_channels_ == 1) {
AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
mono_audio);
audio_ptr = mono_audio;
audio_ptr_num_channels = 1;
}
if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
audio_ptr_num_channels) == -1) {
LOG(LS_ERROR) << "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;
assert(false);
}
const size_t src_length = samples_per_channel * audio_ptr_num_channels;
int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
AudioFrame::kMaxDataSizeSamples);
if (out_length == -1) {
LOG(LS_ERROR) << "Resample failed: audio_ptr = " << audio_ptr
<< ", src_length = " << src_length
<< ", dst_frame->data_ = " << dst_frame->data_;
assert(false);
}
dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
// Upmix after resampling.
if (num_channels == 1 && dst_frame->num_channels_ == 2) {
// The audio in dst_frame really is mono at this point; MonoToStereo will
// set this back to stereo.
dst_frame->num_channels_ = 1;
AudioFrameOperations::MonoToStereo(dst_frame);
}
}
void MixWithSat(int16_t target[],
size_t target_channel,
const int16_t source[],
size_t source_channel,
size_t source_len) {
assert(target_channel == 1 || target_channel == 2);
assert(source_channel == 1 || source_channel == 2);
if (target_channel == 2 && source_channel == 1) {
// Convert source from mono to stereo.
int32_t left = 0;
int32_t right = 0;
for (size_t i = 0; i < source_len; ++i) {
left = source[i] + target[i * 2];
right = source[i] + target[i * 2 + 1];
target[i * 2] = WebRtcSpl_SatW32ToW16(left);
target[i * 2 + 1] = WebRtcSpl_SatW32ToW16(right);
}
} else if (target_channel == 1 && source_channel == 2) {
// Convert source from stereo to mono.
int32_t temp = 0;
for (size_t i = 0; i < source_len / 2; ++i) {
temp = ((source[i * 2] + source[i * 2 + 1]) >> 1) + target[i];
target[i] = WebRtcSpl_SatW32ToW16(temp);
}
} else {
int32_t temp = 0;
for (size_t i = 0; i < source_len; ++i) {
temp = source[i] + target[i];
target[i] = WebRtcSpl_SatW32ToW16(temp);
}
}
}
} // namespace voe
} // namespace webrtc
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