Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 107 insertions, 0 deletions

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 <stddef.h>
+#include <stdint.h>
+
+#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_