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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/audio/remix_resample.h
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/remix_resample.h')
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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_