From 36d22d82aa202bb199967e9512281e9a53db42c9 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 21:33:14 +0200 Subject: Adding upstream version 115.7.0esr. Signed-off-by: Daniel Baumann --- .../audio/utility/audio_frame_operations.h | 107 +++++++++++++++++++++ 1 file changed, 107 insertions(+) create mode 100644 third_party/libwebrtc/audio/utility/audio_frame_operations.h (limited to 'third_party/libwebrtc/audio/utility/audio_frame_operations.h') diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations.h b/third_party/libwebrtc/audio/utility/audio_frame_operations.h new file mode 100644 index 0000000000..2a5f29f4f5 --- /dev/null +++ b/third_party/libwebrtc/audio/utility/audio_frame_operations.h @@ -0,0 +1,107 @@ +/* + * 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_UTILITY_AUDIO_FRAME_OPERATIONS_H_ +#define AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_ + +#include +#include + +#include "absl/base/attributes.h" +#include "api/audio/audio_frame.h" + +namespace webrtc { + +// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h. +// Change reference parameters to pointers. Consider using a namespace rather +// than a class. +class AudioFrameOperations { + public: + // Add samples in `frame_to_add` with samples in `result_frame` + // putting the results in `results_frame`. The fields + // `vad_activity_` and `speech_type_` of the result frame are + // updated. If `result_frame` is empty (`samples_per_channel_`==0), + // the samples in `frame_to_add` are added to it. The number of + // channels and number of samples per channel must match except when + // `result_frame` is empty. + static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame); + + // `frame.num_channels_` will be updated. This version checks for sufficient + // buffer size and that `num_channels_` is mono. Use UpmixChannels + // instead. TODO(bugs.webrtc.org/8649): remove. + ABSL_DEPRECATED("bugs.webrtc.org/8649") + static int MonoToStereo(AudioFrame* frame); + + // `frame.num_channels_` will be updated. This version checks that + // `num_channels_` is stereo. Use DownmixChannels + // instead. TODO(bugs.webrtc.org/8649): remove. + ABSL_DEPRECATED("bugs.webrtc.org/8649") + static int StereoToMono(AudioFrame* frame); + + // Downmixes 4 channels `src_audio` to stereo `dst_audio`. This is an in-place + // operation, meaning `src_audio` and `dst_audio` may point to the same + // buffer. + static void QuadToStereo(const int16_t* src_audio, + size_t samples_per_channel, + int16_t* dst_audio); + + // `frame.num_channels_` will be updated. This version checks that + // `num_channels_` is 4 channels. + static int QuadToStereo(AudioFrame* frame); + + // Downmixes `src_channels` `src_audio` to `dst_channels` `dst_audio`. + // This is an in-place operation, meaning `src_audio` and `dst_audio` + // may point to the same buffer. Supported channel combinations are + // Stereo to Mono, Quad to Mono, and Quad to Stereo. + static void DownmixChannels(const int16_t* src_audio, + size_t src_channels, + size_t samples_per_channel, + size_t dst_channels, + int16_t* dst_audio); + + // `frame.num_channels_` will be updated. This version checks that + // `num_channels_` and `dst_channels` are valid and performs relevant downmix. + // Supported channel combinations are N channels to Mono, and Quad to Stereo. + static void DownmixChannels(size_t dst_channels, AudioFrame* frame); + + // `frame.num_channels_` will be updated. This version checks that + // `num_channels_` and `dst_channels` are valid and performs relevant + // downmix. Supported channel combinations are Mono to N + // channels. The single channel is replicated. + static void UpmixChannels(size_t target_number_of_channels, + AudioFrame* frame); + + // Swap the left and right channels of `frame`. Fails silently if `frame` is + // not stereo. + static void SwapStereoChannels(AudioFrame* frame); + + // Conditionally zero out contents of `frame` for implementing audio mute: + // `previous_frame_muted` && `current_frame_muted` - Zero out whole frame. + // `previous_frame_muted` && !`current_frame_muted` - Fade-in at frame start. + // !`previous_frame_muted` && `current_frame_muted` - Fade-out at frame end. + // !`previous_frame_muted` && !`current_frame_muted` - Leave frame untouched. + static void Mute(AudioFrame* frame, + bool previous_frame_muted, + bool current_frame_muted); + + // Zero out contents of frame. + static void Mute(AudioFrame* frame); + + // Halve samples in `frame`. + static void ApplyHalfGain(AudioFrame* frame); + + static int Scale(float left, float right, AudioFrame* frame); + + static int ScaleWithSat(float scale, AudioFrame* frame); +}; + +} // namespace webrtc + +#endif // AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_ -- cgit v1.2.3