| /* |
| * 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. |
| */ |
| |
| #ifndef AUDIO_REMIX_RESAMPLE_H_ |
| #define AUDIO_REMIX_RESAMPLE_H_ |
| |
| #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 |
| |
| #endif // AUDIO_REMIX_RESAMPLE_H_ |