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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/output_mixer_internal.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/utility/interface/audio_frame_operations.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
namespace voe {
int RemixAndResample(const AudioFrame& src_frame,
PushResampler* resampler,
AudioFrame* dst_frame) {
const int16_t* audio_ptr = src_frame.data_;
int audio_ptr_num_channels = src_frame.num_channels_;
int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
// Downmix before resampling.
if (src_frame.num_channels_ == 2 && dst_frame->num_channels_ == 1) {
AudioFrameOperations::StereoToMono(src_frame.data_,
src_frame.samples_per_channel_,
mono_audio);
audio_ptr = mono_audio;
audio_ptr_num_channels = 1;
}
if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_,
dst_frame->sample_rate_hz_,
audio_ptr_num_channels) == -1) {
dst_frame->CopyFrom(src_frame);
LOG_FERR3(LS_ERROR, InitializeIfNeeded, src_frame.sample_rate_hz_,
dst_frame->sample_rate_hz_, audio_ptr_num_channels);
return -1;
}
const int src_length = src_frame.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) {
dst_frame->CopyFrom(src_frame);
LOG_FERR3(LS_ERROR, Resample, src_length, dst_frame->data_,
AudioFrame::kMaxDataSizeSamples);
return -1;
}
dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
// Upmix after resampling.
if (src_frame.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);
}
return 0;
}
} // namespace voe
} // namespace webrtc
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