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Upstream commit: https://webrtc.googlesource.com/src/+/b0ea637ec28a73d2a323ea03026ab191f232e7f6 Use backticks not vertical bars to denote variables in comments for /audio Bug: webrtc:12338 Change-Id: Ief89269aa39d0cb6749a1c6cc995ce8830ca327f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226942 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34564}
44 lines
1.6 KiB
C++
44 lines
1.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef AUDIO_REMIX_RESAMPLE_H_
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#define AUDIO_REMIX_RESAMPLE_H_
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#include "api/audio/audio_frame.h"
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#include "common_audio/resampler/include/push_resampler.h"
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namespace webrtc {
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namespace voe {
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// Upmix or downmix and resample the audio to `dst_frame`. Expects `dst_frame`
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// to have its sample rate and channels members set to the desired values.
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// Updates the `samples_per_channel_` member accordingly.
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//
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// This version has an AudioFrame `src_frame` as input and sets the output
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// `timestamp_`, `elapsed_time_ms_` and `ntp_time_ms_` members equals to the
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// input ones.
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void RemixAndResample(const AudioFrame& src_frame,
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PushResampler<int16_t>* resampler,
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AudioFrame* dst_frame);
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// This version has a pointer to the samples `src_data` as input and receives
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// `samples_per_channel`, `num_channels` and `sample_rate_hz` of the data as
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// parameters.
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void RemixAndResample(const int16_t* src_data,
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size_t samples_per_channel,
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size_t num_channels,
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int sample_rate_hz,
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PushResampler<int16_t>* resampler,
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AudioFrame* dst_frame);
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} // namespace voe
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} // namespace webrtc
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#endif // AUDIO_REMIX_RESAMPLE_H_
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