Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

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

1 files changed, 75 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/agc/legacy/digital_agc.h b/third_party/libwebrtc/modules/audio_processing/agc/legacy/digital_agc.h
new file mode 100644
index 0000000000..223c74b9bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/legacy/digital_agc.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2011 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_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
+#define MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+
+typedef struct {
+  int32_t downState[8];
+  int16_t HPstate;
+  int16_t counter;
+  int16_t logRatio;           // log( P(active) / P(inactive) ) (Q10)
+  int16_t meanLongTerm;       // Q10
+  int32_t varianceLongTerm;   // Q8
+  int16_t stdLongTerm;        // Q10
+  int16_t meanShortTerm;      // Q10
+  int32_t varianceShortTerm;  // Q8
+  int16_t stdShortTerm;       // Q10
+} AgcVad;                     // total = 54 bytes
+
+typedef struct {
+  int32_t capacitorSlow;
+  int32_t capacitorFast;
+  int32_t gain;
+  int32_t gainTable[32];
+  int16_t gatePrevious;
+  int16_t agcMode;
+  AgcVad vadNearend;
+  AgcVad vadFarend;
+} DigitalAgc;
+
+int32_t WebRtcAgc_InitDigital(DigitalAgc* digitalAgcInst, int16_t agcMode);
+
+int32_t WebRtcAgc_ComputeDigitalGains(DigitalAgc* digitalAgcInst,
+                                      const int16_t* const* inNear,
+                                      size_t num_bands,
+                                      uint32_t FS,
+                                      int16_t lowLevelSignal,
+                                      int32_t gains[11]);
+
+int32_t WebRtcAgc_ApplyDigitalGains(const int32_t gains[11],
+                                    size_t num_bands,
+                                    uint32_t FS,
+                                    const int16_t* const* in_near,
+                                    int16_t* const* out);
+
+int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* digitalAgcInst,
+                                     const int16_t* inFar,
+                                     size_t nrSamples);
+
+void WebRtcAgc_InitVad(AgcVad* vadInst);
+
+int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst,    // (i) VAD state
+                             const int16_t* in,  // (i) Speech signal
+                             size_t nrSamples);  // (i) number of samples
+
+int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable,         // Q16
+                                     int16_t compressionGaindB,  // Q0 (in dB)
+                                     int16_t targetLevelDbfs,    // Q0 (in dB)
+                                     uint8_t limiterEnable,
+                                     int16_t analogTarget);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_