OLD | NEW |
(Empty) | |
| 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| 3 * |
| 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 |
| 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ |
| 10 |
| 11 #ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_ |
| 12 #define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_ |
| 13 |
| 14 #include <map> |
| 15 #include "webrtc/base/timeutils.h" |
| 16 #include "webrtc/common_types.h" |
| 17 |
| 18 namespace voetest { |
| 19 |
| 20 class LoudestFilter { |
| 21 public: |
| 22 /* ForwardThisPacket() |
| 23 * Decide whether to forward a RTP packet, given its header. |
| 24 * |
| 25 * Input: |
| 26 * rtp_header : Header of the RTP packet of interest. |
| 27 */ |
| 28 bool ForwardThisPacket(const webrtc::RTPHeader& rtp_header); |
| 29 |
| 30 private: |
| 31 struct Status { |
| 32 void Set(int audio_level, uint32 last_time_ms) { |
| 33 this->audio_level = audio_level; |
| 34 this->last_time_ms = last_time_ms; |
| 35 } |
| 36 int audio_level; |
| 37 uint32 last_time_ms; |
| 38 }; |
| 39 |
| 40 void RemoveTimeoutStreams(uint32 time_ms); |
| 41 unsigned int FindQuietestStream(); |
| 42 |
| 43 // Keeps the streams being forwarded in pair<SSRC, Status>. |
| 44 std::map<unsigned int, Status> stream_levels_; |
| 45 |
| 46 const int32 kStreamTimeOutMs = 5000; |
| 47 const size_t kMaxMixSize = 3; |
| 48 const int kInvalidAudioLevel = 128; |
| 49 }; |
| 50 |
| 51 |
| 52 } // namespace voetest |
| 53 |
| 54 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_ |
OLD | NEW |