| /* |
| * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_ |
| #define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_ |
| |
| #include <map> |
| #include "webrtc/common_types.h" |
| #include "webrtc/rtc_base/timeutils.h" |
| |
| namespace webrtc { |
| namespace voetest { |
| |
| class LoudestFilter { |
| public: |
| /* ForwardThisPacket() |
| * Decide whether to forward a RTP packet, given its header. |
| * |
| * Input: |
| * rtp_header : Header of the RTP packet of interest. |
| */ |
| bool ForwardThisPacket(const webrtc::RTPHeader& rtp_header); |
| |
| private: |
| struct Status { |
| void Set(int audio_level, int64_t last_time_ms) { |
| this->audio_level = audio_level; |
| this->last_time_ms = last_time_ms; |
| } |
| int audio_level; |
| int64_t last_time_ms; |
| }; |
| |
| void RemoveTimeoutStreams(int64_t time_ms); |
| unsigned int FindQuietestStream(); |
| |
| // Keeps the streams being forwarded in pair<SSRC, Status>. |
| std::map<unsigned int, Status> stream_levels_; |
| |
| const int32_t kStreamTimeOutMs = 5000; |
| const size_t kMaxMixSize = 3; |
| const int kInvalidAudioLevel = 128; |
| }; |
| |
| } // namespace voetest |
| } // namespace webrtc |
| |
| #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_ |