Index: webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h |
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..4b432763e2534b2e779242e88812fee5a3fbf3af |
--- /dev/null |
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h |
@@ -0,0 +1,56 @@ |
+/* |
+ * Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_ |
+#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_ |
+ |
+#include "webrtc/base/optional.h" |
+ |
+namespace webrtc { |
+ |
+// An AudioNetworkAdaptor optimizes the audio experience by suggesting a |
+// suitable runtime configuration (bit rate, frame length, FEC, etc.) to the |
+// encoder based on network metrics. |
+class AudioNetworkAdaptor { |
+ public: |
+ struct EncoderRuntimeConfig { |
+ EncoderRuntimeConfig(); |
+ EncoderRuntimeConfig(const EncoderRuntimeConfig& other); |
+ ~EncoderRuntimeConfig(); |
+ rtc::Optional<int> bitrate_bps; |
+ rtc::Optional<int> frame_length_ms; |
+ rtc::Optional<float> uplink_packet_loss_fraction; |
+ rtc::Optional<bool> enable_fec; |
+ rtc::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. |
+ rtc::Optional<size_t> num_channels; |
+ }; |
+ |
+ virtual ~AudioNetworkAdaptor() = default; |
+ |
+ virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0; |
+ |
+ virtual void SetUplinkPacketLossFraction( |
+ float uplink_packet_loss_fraction) = 0; |
+ |
+ virtual void SetReceiverFrameLengthRange(int min_frame_length_ms, |
+ int max_frame_length_ms) = 0; |
+ |
+ virtual EncoderRuntimeConfig GetEncoderRuntimeConfig() = 0; |
+ |
+ virtual void StartDebugDump(FILE* file_handle) = 0; |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_ |