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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/base/optional.h" | 14 #include "webrtc/base/optional.h" |
15 | 15 |
16 namespace webrtc { | 16 namespace webrtc { |
17 | 17 |
| 18 struct AudioEncoderRuntimeConfig { |
| 19 AudioEncoderRuntimeConfig(); |
| 20 AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other); |
| 21 ~AudioEncoderRuntimeConfig(); |
| 22 rtc::Optional<int> bitrate_bps; |
| 23 rtc::Optional<int> frame_length_ms; |
| 24 // 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 rtc::Optional<float> uplink_packet_loss_fraction; |
| 27 rtc::Optional<bool> enable_fec; |
| 28 rtc::Optional<bool> enable_dtx; |
| 29 |
| 30 // 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 // to encode. |
| 33 rtc::Optional<size_t> num_channels; |
| 34 }; |
| 35 |
18 // An AudioNetworkAdaptor optimizes the audio experience by suggesting a | 36 // An AudioNetworkAdaptor optimizes the audio experience by suggesting a |
19 // suitable runtime configuration (bit rate, frame length, FEC, etc.) to the | 37 // suitable runtime configuration (bit rate, frame length, FEC, etc.) to the |
20 // encoder based on network metrics. | 38 // encoder based on network metrics. |
21 class AudioNetworkAdaptor { | 39 class AudioNetworkAdaptor { |
22 public: | 40 public: |
23 struct EncoderRuntimeConfig { | |
24 EncoderRuntimeConfig(); | |
25 EncoderRuntimeConfig(const EncoderRuntimeConfig& other); | |
26 ~EncoderRuntimeConfig(); | |
27 rtc::Optional<int> bitrate_bps; | |
28 rtc::Optional<int> frame_length_ms; | |
29 // Note: This is what we tell the encoder. It doesn't have to reflect | |
30 // the actual NetworkMetrics; it's subject to our decision. | |
31 rtc::Optional<float> uplink_packet_loss_fraction; | |
32 rtc::Optional<bool> enable_fec; | |
33 rtc::Optional<bool> enable_dtx; | |
34 | |
35 // Some encoders can encode fewer channels than the actual input to make | |
36 // better use of the bandwidth. |num_channels| sets the number of channels | |
37 // to encode. | |
38 rtc::Optional<size_t> num_channels; | |
39 }; | |
40 | 41 |
41 virtual ~AudioNetworkAdaptor() = default; | 42 virtual ~AudioNetworkAdaptor() = default; |
42 | 43 |
43 virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0; | 44 virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0; |
44 | 45 |
45 virtual void SetUplinkPacketLossFraction( | 46 virtual void SetUplinkPacketLossFraction( |
46 float uplink_packet_loss_fraction) = 0; | 47 float uplink_packet_loss_fraction) = 0; |
47 | 48 |
48 virtual void SetUplinkRecoverablePacketLossFraction( | 49 virtual void SetUplinkRecoverablePacketLossFraction( |
49 float uplink_recoverable_packet_loss_fraction) = 0; | 50 float uplink_recoverable_packet_loss_fraction) = 0; |
50 | 51 |
51 virtual void SetRtt(int rtt_ms) = 0; | 52 virtual void SetRtt(int rtt_ms) = 0; |
52 | 53 |
53 virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0; | 54 virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0; |
54 | 55 |
55 virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0; | 56 virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0; |
56 | 57 |
57 virtual EncoderRuntimeConfig GetEncoderRuntimeConfig() = 0; | 58 virtual AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() = 0; |
58 | 59 |
59 virtual void StartDebugDump(FILE* file_handle) = 0; | 60 virtual void StartDebugDump(FILE* file_handle) = 0; |
60 | 61 |
61 virtual void StopDebugDump() = 0; | 62 virtual void StopDebugDump() = 0; |
62 }; | 63 }; |
63 | 64 |
64 } // namespace webrtc | 65 } // namespace webrtc |
65 | 66 |
66 #endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWO
RK_ADAPTOR_H_ | 67 #endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWO
RK_ADAPTOR_H_ |
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