Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 76 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/normal.h b/third_party/libwebrtc/modules/audio_coding/neteq/normal.h
new file mode 100644
index 0000000000..772293b605
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/normal.h
@@ -0,0 +1,76 @@
+/*
+ *  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 MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
+#define MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
+
+#include <stdint.h>
+#include <string.h>  // Access to size_t.
+
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class AudioMultiVector;
+class BackgroundNoise;
+class DecoderDatabase;
+class Expand;
+
+// This class provides the "Normal" DSP operation, that is performed when
+// there is no data loss, no need to stretch the timing of the signal, and
+// no other "special circumstances" are at hand.
+class Normal {
+ public:
+  Normal(int fs_hz,
+         DecoderDatabase* decoder_database,
+         const BackgroundNoise& background_noise,
+         Expand* expand,
+         StatisticsCalculator* statistics)
+      : fs_hz_(fs_hz),
+        decoder_database_(decoder_database),
+        background_noise_(background_noise),
+        expand_(expand),
+        samples_per_ms_(rtc::CheckedDivExact(fs_hz_, 1000)),
+        default_win_slope_Q14_(
+            rtc::dchecked_cast<uint16_t>((1 << 14) / samples_per_ms_)),
+        statistics_(statistics) {}
+
+  virtual ~Normal() {}
+
+  Normal(const Normal&) = delete;
+  Normal& operator=(const Normal&) = delete;
+
+  // Performs the "Normal" operation. The decoder data is supplied in `input`,
+  // having `length` samples in total for all channels (interleaved). The
+  // result is written to `output`. The number of channels allocated in
+  // `output` defines the number of channels that will be used when
+  // de-interleaving `input`. `last_mode` contains the mode used in the previous
+  // GetAudio call (i.e., not the current one).
+  int Process(const int16_t* input,
+              size_t length,
+              NetEq::Mode last_mode,
+              AudioMultiVector* output);
+
+ private:
+  int fs_hz_;
+  DecoderDatabase* decoder_database_;
+  const BackgroundNoise& background_noise_;
+  Expand* expand_;
+  const size_t samples_per_ms_;
+  const int16_t default_win_slope_Q14_;
+  StatisticsCalculator* const statistics_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_NORMAL_H_