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| 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/common_types.h" | |
| 16 #include "webrtc/rtc_base/timeutils.h" | |
| 17 | |
| 18 namespace webrtc { | |
| 19 namespace voetest { | |
| 20 | |
| 21 class LoudestFilter { | |
| 22 public: | |
| 23 /* ForwardThisPacket() | |
| 24 * Decide whether to forward a RTP packet, given its header. | |
| 25 * | |
| 26 * Input: | |
| 27 * rtp_header : Header of the RTP packet of interest. | |
| 28 */ | |
| 29 bool ForwardThisPacket(const webrtc::RTPHeader& rtp_header); | |
| 30 | |
| 31 private: | |
| 32 struct Status { | |
| 33 void Set(int audio_level, int64_t last_time_ms) { | |
| 34 this->audio_level = audio_level; | |
| 35 this->last_time_ms = last_time_ms; | |
| 36 } | |
| 37 int audio_level; | |
| 38 int64_t last_time_ms; | |
| 39 }; | |
| 40 | |
| 41 void RemoveTimeoutStreams(int64_t time_ms); | |
| 42 unsigned int FindQuietestStream(); | |
| 43 | |
| 44 // Keeps the streams being forwarded in pair<SSRC, Status>. | |
| 45 std::map<unsigned int, Status> stream_levels_; | |
| 46 | |
| 47 const int32_t kStreamTimeOutMs = 5000; | |
| 48 const size_t kMaxMixSize = 3; | |
| 49 const int kInvalidAudioLevel = 128; | |
| 50 }; | |
| 51 | |
| 52 } // namespace voetest | |
| 53 } // namespace webrtc | |
| 54 | |
| 55 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_ | |
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