OLD | NEW |
1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_
ADAPTOR_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_
ADAPTOR_H_ |
12 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_
ADAPTOR_H_ | 12 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_
ADAPTOR_H_ |
13 | 13 |
| 14 #include "webrtc/api/audio_codecs/audio_encoder.h" |
14 #include "webrtc/api/optional.h" | 15 #include "webrtc/api/optional.h" |
15 | 16 |
16 namespace webrtc { | 17 namespace webrtc { |
17 | 18 |
18 struct AudioEncoderRuntimeConfig { | 19 struct AudioEncoderRuntimeConfig { |
19 AudioEncoderRuntimeConfig(); | 20 AudioEncoderRuntimeConfig(); |
20 AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other); | 21 AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other); |
21 ~AudioEncoderRuntimeConfig(); | 22 ~AudioEncoderRuntimeConfig(); |
22 rtc::Optional<int> bitrate_bps; | 23 rtc::Optional<int> bitrate_bps; |
23 rtc::Optional<int> frame_length_ms; | 24 rtc::Optional<int> frame_length_ms; |
24 // Note: This is what we tell the encoder. It doesn't have to reflect | 25 // Note: This is what we tell the encoder. It doesn't have to reflect |
25 // the actual NetworkMetrics; it's subject to our decision. | 26 // the actual NetworkMetrics; it's subject to our decision. |
26 rtc::Optional<float> uplink_packet_loss_fraction; | 27 rtc::Optional<float> uplink_packet_loss_fraction; |
27 rtc::Optional<bool> enable_fec; | 28 rtc::Optional<bool> enable_fec; |
28 rtc::Optional<bool> enable_dtx; | 29 rtc::Optional<bool> enable_dtx; |
29 | 30 |
30 // Some encoders can encode fewer channels than the actual input to make | 31 // Some encoders can encode fewer channels than the actual input to make |
31 // better use of the bandwidth. |num_channels| sets the number of channels | 32 // better use of the bandwidth. |num_channels| sets the number of channels |
32 // to encode. | 33 // to encode. |
33 rtc::Optional<size_t> num_channels; | 34 rtc::Optional<size_t> num_channels; |
34 }; | 35 }; |
35 | 36 |
36 // An AudioNetworkAdaptor optimizes the audio experience by suggesting a | 37 // An AudioNetworkAdaptor optimizes the audio experience by suggesting a |
37 // suitable runtime configuration (bit rate, frame length, FEC, etc.) to the | 38 // suitable runtime configuration (bit rate, frame length, FEC, etc.) to the |
38 // encoder based on network metrics. | 39 // encoder based on network metrics. |
39 class AudioNetworkAdaptor { | 40 class AudioNetworkAdaptor { |
40 public: | 41 public: |
41 | |
42 virtual ~AudioNetworkAdaptor() = default; | 42 virtual ~AudioNetworkAdaptor() = default; |
43 | 43 |
44 virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0; | 44 virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0; |
45 | 45 |
46 virtual void SetUplinkPacketLossFraction( | 46 virtual void SetUplinkPacketLossFraction( |
47 float uplink_packet_loss_fraction) = 0; | 47 float uplink_packet_loss_fraction) = 0; |
48 | 48 |
49 virtual void SetUplinkRecoverablePacketLossFraction( | 49 virtual void SetUplinkRecoverablePacketLossFraction( |
50 float uplink_recoverable_packet_loss_fraction) = 0; | 50 float uplink_recoverable_packet_loss_fraction) = 0; |
51 | 51 |
52 virtual void SetRtt(int rtt_ms) = 0; | 52 virtual void SetRtt(int rtt_ms) = 0; |
53 | 53 |
54 virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0; | 54 virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0; |
55 | 55 |
56 virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0; | 56 virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0; |
57 | 57 |
58 virtual AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() = 0; | 58 virtual AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() = 0; |
59 | 59 |
60 virtual void StartDebugDump(FILE* file_handle) = 0; | 60 virtual void StartDebugDump(FILE* file_handle) = 0; |
61 | 61 |
62 virtual void StopDebugDump() = 0; | 62 virtual void StopDebugDump() = 0; |
| 63 |
| 64 virtual ANAStats GetStats() const = 0; |
63 }; | 65 }; |
64 | 66 |
65 } // namespace webrtc | 67 } // namespace webrtc |
66 | 68 |
67 #endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWO
RK_ADAPTOR_H_ | 69 #endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWO
RK_ADAPTOR_H_ |
OLD | NEW |