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32c3398e1a
Also adding more checks around parameter valuesm since the `sample_rate_hz` argument is technically not needed, but will be removed in a follow-up CL. Bug: chromium:335805780 Change-Id: Ia7e50658f8a686ab71980f9c59cce5f097b0af40 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353340 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42581}
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 "api/audio/audio_view.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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// Note: The RemixAndResample methods assume 10ms buffer sizes.
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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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// TODO(tommi): The `sample_rate_hz` argument can probably be removed since it's
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// always related to `src_data.samples_per_frame()'.
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void RemixAndResample(InterleavedView<const int16_t> src_data,
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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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