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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.
- */
- #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_
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