1 files changed, 51 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
new file mode 100644
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--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2017 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_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
+
+#include <stddef.h>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+struct AudioEncoderRuntimeConfig {
+  AudioEncoderRuntimeConfig();
+  AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other);
+  ~AudioEncoderRuntimeConfig();
+  AudioEncoderRuntimeConfig& operator=(const AudioEncoderRuntimeConfig& other);
+  bool operator==(const AudioEncoderRuntimeConfig& other) const;
+  absl::optional<int> bitrate_bps;
+  absl::optional<int> frame_length_ms;
+  // Note: This is what we tell the encoder. It doesn't have to reflect
+  // the actual NetworkMetrics; it's subject to our decision.
+  absl::optional<float> uplink_packet_loss_fraction;
+  absl::optional<bool> enable_fec;
+  absl::optional<bool> enable_dtx;
+
+  // Some encoders can encode fewer channels than the actual input to make
+  // better use of the bandwidth. `num_channels` sets the number of channels
+  // to encode.
+  absl::optional<size_t> num_channels;
+
+  // This is true if the last frame length change was an increase, and otherwise
+  // false.
+  // The value of this boolean is used to apply a different offset to the
+  // per-packet overhead that is reported by the BWE. The exact offset value
+  // is most important right after a frame length change, because the frame
+  // length change affects the overhead. In the steady state, the exact value is
+  // not important because the BWE will compensate.
+  bool last_fl_change_increase = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_