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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
commit6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch)
treea68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/test/audioproc_float.h
parentInitial commit. (diff)
downloadthunderbird-upstream.tar.xz
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Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
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+/*
+ * Copyright (c) 2018 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 API_TEST_AUDIOPROC_FLOAT_H_
+#define API_TEST_AUDIOPROC_FLOAT_H_
+
+#include <memory>
+#include <vector>
+
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+namespace test {
+
+// This is an interface for the audio processing simulation utility. This
+// utility can be used to simulate the audioprocessing module using a recording
+// (either an AEC dump or wav files), and generate the output as a wav file.
+// Any audio_processing object specified in the input is used for the
+// simulation. The optional `audio_processing` object provides the
+// AudioProcessing instance that is used during the simulation. Note that when
+// the audio_processing object is specified all functionality that relies on
+// using the AudioProcessingBuilder is deactivated, since the AudioProcessing
+// object is already created and the builder is not used in the simulation. It
+// is needed to pass the command line flags as `argc` and `argv`, so these can
+// be interpreted properly by the utility. To see a list of all supported
+// command line flags, run the executable with the '--help' flag.
+int AudioprocFloat(rtc::scoped_refptr<AudioProcessing> audio_processing,
+ int argc,
+ char* argv[]);
+
+// This is an interface for the audio processing simulation utility. This
+// utility can be used to simulate the audioprocessing module using a recording
+// (either an AEC dump or wav files), and generate the output as a wav file.
+// The `ap_builder` object will be used to create the AudioProcessing instance
+// that is used during the simulation. The `ap_builder` supports setting of
+// injectable components, which will be passed on to the created AudioProcessing
+// instance. It is needed to pass the command line flags as `argc` and `argv`,
+// so these can be interpreted properly by the utility.
+// To get a fully-working audioproc_f utility, all that is needed is to write a
+// main function, create an AudioProcessingBuilder, optionally set custom
+// processing components on it, and pass the builder together with the command
+// line arguments into this function.
+// To see a list of all supported command line flags, run the executable with
+// the '--help' flag.
+int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
+ int argc,
+ char* argv[]);
+
+// Interface for the audio processing simulation utility, which is similar to
+// the one above, but which adds the option of receiving the input as a string
+// and returning the output as an array. The first three arguments fulfill the
+// same purpose as above. Pass the `input_aecdump` to provide the content of an
+// AEC dump file as a string. After the simulation is completed,
+// `processed_capture_samples` will contain the the samples processed on the
+// capture side.
+int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
+ int argc,
+ char* argv[],
+ absl::string_view input_aecdump,
+ std::vector<float>* processed_capture_samples);
+} // namespace test
+} // namespace webrtc
+
+#endif // API_TEST_AUDIOPROC_FLOAT_H_