blob: a45270b39a61b4117d1742380ffb5c3eb225f697 [file] [log] [blame]
* 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 "api/audio/audio_frame.h"
#include "common_audio/resampler/include/push_resampler.h"
namespace webrtc {
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,
size_t num_channels,
int sample_rate_hz,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame);
} // namespace voe
} // namespace webrtc