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| 1 /* | 1 /* |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 |
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| 49 RtcpStatistics rtcp_stats; | 49 RtcpStatistics rtcp_stats; |
| 50 }; | 50 }; |
| 51 | 51 |
| 52 struct Stats { | 52 struct Stats { |
| 53 std::string ToString(int64_t time_ms) const; | 53 std::string ToString(int64_t time_ms) const; |
| 54 std::string encoder_implementation_name = "unknown"; | 54 std::string encoder_implementation_name = "unknown"; |
| 55 int input_frame_rate = 0; | 55 int input_frame_rate = 0; |
| 56 int encode_frame_rate = 0; | 56 int encode_frame_rate = 0; |
| 57 int avg_encode_time_ms = 0; | 57 int avg_encode_time_ms = 0; |
| 58 int encode_usage_percent = 0; | 58 int encode_usage_percent = 0; |
| 59 uint32_t frames_encoded = 0; |
| 59 // Bitrate the encoder is currently configured to use due to bandwidth | 60 // Bitrate the encoder is currently configured to use due to bandwidth |
| 60 // limitations. | 61 // limitations. |
| 61 int target_media_bitrate_bps = 0; | 62 int target_media_bitrate_bps = 0; |
| 62 // Bitrate the encoder is actually producing. | 63 // Bitrate the encoder is actually producing. |
| 63 int media_bitrate_bps = 0; | 64 int media_bitrate_bps = 0; |
| 64 // Media bitrate this VideoSendStream is configured to prefer if there are | 65 // Media bitrate this VideoSendStream is configured to prefer if there are |
| 65 // no bandwidth limitations. | 66 // no bandwidth limitations. |
| 66 int preferred_media_bitrate_bps = 0; | 67 int preferred_media_bitrate_bps = 0; |
| 67 bool suspended = false; | 68 bool suspended = false; |
| 68 bool bw_limited_resolution = false; | 69 bool bw_limited_resolution = false; |
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| 217 EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0); | 218 EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0); |
| 218 } | 219 } |
| 219 | 220 |
| 220 protected: | 221 protected: |
| 221 virtual ~VideoSendStream() {} | 222 virtual ~VideoSendStream() {} |
| 222 }; | 223 }; |
| 223 | 224 |
| 224 } // namespace webrtc | 225 } // namespace webrtc |
| 225 | 226 |
| 226 #endif // WEBRTC_VIDEO_SEND_STREAM_H_ | 227 #endif // WEBRTC_VIDEO_SEND_STREAM_H_ |
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