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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/common_audio/audio_converter.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/common_audio/audio_converter.h')
-rw-r--r-- | third_party/libwebrtc/common_audio/audio_converter.h | 72 |
1 files changed, 72 insertions, 0 deletions
diff --git a/third_party/libwebrtc/common_audio/audio_converter.h b/third_party/libwebrtc/common_audio/audio_converter.h new file mode 100644 index 0000000000..4afbb6d0fd --- /dev/null +++ b/third_party/libwebrtc/common_audio/audio_converter.h @@ -0,0 +1,72 @@ +/* + * Copyright (c) 2014 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 COMMON_AUDIO_AUDIO_CONVERTER_H_ +#define COMMON_AUDIO_AUDIO_CONVERTER_H_ + +#include <stddef.h> + +#include <memory> + +namespace webrtc { + +// Format conversion (remixing and resampling) for audio. Only simple remixing +// conversions are supported: downmix to mono (i.e. `dst_channels` == 1) or +// upmix from mono (i.e. |src_channels == 1|). +// +// The source and destination chunks have the same duration in time; specifying +// the number of frames is equivalent to specifying the sample rates. +class AudioConverter { + public: + // Returns a new AudioConverter, which will use the supplied format for its + // lifetime. Caller is responsible for the memory. + static std::unique_ptr<AudioConverter> Create(size_t src_channels, + size_t src_frames, + size_t dst_channels, + size_t dst_frames); + virtual ~AudioConverter() {} + + AudioConverter(const AudioConverter&) = delete; + AudioConverter& operator=(const AudioConverter&) = delete; + + // Convert `src`, containing `src_size` samples, to `dst`, having a sample + // capacity of `dst_capacity`. Both point to a series of buffers containing + // the samples for each channel. The sizes must correspond to the format + // passed to Create(). + virtual void Convert(const float* const* src, + size_t src_size, + float* const* dst, + size_t dst_capacity) = 0; + + size_t src_channels() const { return src_channels_; } + size_t src_frames() const { return src_frames_; } + size_t dst_channels() const { return dst_channels_; } + size_t dst_frames() const { return dst_frames_; } + + protected: + AudioConverter(); + AudioConverter(size_t src_channels, + size_t src_frames, + size_t dst_channels, + size_t dst_frames); + + // Helper to RTC_CHECK that inputs are correctly sized. + void CheckSizes(size_t src_size, size_t dst_capacity) const; + + private: + const size_t src_channels_; + const size_t src_frames_; + const size_t dst_channels_; + const size_t dst_frames_; +}; + +} // namespace webrtc + +#endif // COMMON_AUDIO_AUDIO_CONVERTER_H_ |