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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/audio/remix_resample.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/remix_resample.h')
-rw-r--r-- | third_party/libwebrtc/audio/remix_resample.h | 44 |
1 files changed, 44 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/remix_resample.h b/third_party/libwebrtc/audio/remix_resample.h new file mode 100644 index 0000000000..bd8da76c6a --- /dev/null +++ b/third_party/libwebrtc/audio/remix_resample.h @@ -0,0 +1,44 @@ +/* + * 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_ |