/* * 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.
*/
// Note: The RemixAndResample methods assume 10ms buffer sizes.
// 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);
// TODO(tommi): The `sample_rate_hz` argument can probably be removed since it's // always related to `src_data.samples_per_frame()'. void RemixAndResample(InterleavedView<const int16_t> src_data, int sample_rate_hz,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame);
} // namespace voe
} // namespace webrtc
#endif// AUDIO_REMIX_RESAMPLE_H_
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