1 files changed, 43 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h
new file mode 100644
index 0000000000..92c5c354a7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h
@@ -0,0 +1,43 @@
+/*
+ *  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 MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
+
+#include <stddef.h>
+
+#include "rtc_base/ignore_wundef.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#include "third_party/opus/src/include/opus.h"
+#include "third_party/opus/src/include/opus_multistream.h"
+RTC_POP_IGNORING_WUNDEF()
+
+struct WebRtcOpusEncInst {
+  OpusEncoder* encoder;
+  OpusMSEncoder* multistream_encoder;
+  size_t channels;
+  int in_dtx_mode;
+  bool avoid_noise_pumping_during_dtx;
+  int sample_rate_hz;
+  float smooth_energy_non_active_frames;
+};
+
+struct WebRtcOpusDecInst {
+  OpusDecoder* decoder;
+  OpusMSDecoder* multistream_decoder;
+  int prev_decoded_samples;
+  bool plc_use_prev_decoded_samples;
+  size_t channels;
+  int in_dtx_mode;
+  int sample_rate_hz;
+};
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_