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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.
*/
/*
* Contains functions often used by different parts of VoiceEngine.
*/
#ifndef WEBRTC_VOICE_ENGINE_UTILITY_H_
#define WEBRTC_VOICE_ENGINE_UTILITY_H_
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
namespace voe {
// Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame|
// to have its sample rate and channels members set to the desired values.
// Updates the |samples_per_channel_| member accordingly.
//
// This version has an AudioFrame |src_frame| as input and sets the output
// |timestamp_|, |elapsed_time_ms_| and |ntp_time_ms_| members equals to the
// input ones.
void RemixAndResample(const AudioFrame& src_frame,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame);
// This version has a pointer to the samples |src_data| as input and receives
// |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as
// parameters.
void RemixAndResample(const int16_t* src_data,
size_t samples_per_channel,
int num_channels,
int sample_rate_hz,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame);
void MixWithSat(int16_t target[],
int target_channel,
const int16_t source[],
int source_channel,
size_t source_len);
} // namespace voe
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_UTILITY_H_
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