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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/test/audioproc_float.h | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/test/audioproc_float.h | 71 |
1 files changed, 71 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/test/audioproc_float.h b/third_party/libwebrtc/api/test/audioproc_float.h new file mode 100644 index 0000000000..1ef1c9828f --- /dev/null +++ b/third_party/libwebrtc/api/test/audioproc_float.h @@ -0,0 +1,71 @@ +/* + * 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_ |