blob: bd8da76c6a1e008d348679a9b8e8b546feaa6d46 [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