Adding upstream version 124.0.1.upstream/124.0.1

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

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_