| /* |
| * Copyright (c) 2017 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_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_ |
| |
| #include <map> |
| #include <set> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/api/optional.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| class NetEqDelayAnalyzer : public test::NetEqPostInsertPacket, |
| public test::NetEqGetAudioCallback { |
| public: |
| void AfterInsertPacket(const test::NetEqInput::PacketData& packet, |
| NetEq* neteq) override; |
| |
| void BeforeGetAudio(NetEq* neteq) override; |
| |
| void AfterGetAudio(int64_t time_now_ms, |
| const AudioFrame& audio_frame, |
| bool muted, |
| NetEq* neteq) override; |
| |
| void CreateGraphs(std::vector<float>* send_times_s, |
| std::vector<float>* arrival_delay_ms, |
| std::vector<float>* corrected_arrival_delay_ms, |
| std::vector<rtc::Optional<float>>* playout_delay_ms, |
| std::vector<rtc::Optional<float>>* target_delay_ms) const; |
| |
| // Creates a matlab script with file name script_name. When executed in |
| // Matlab, the script will generate graphs with the same timing information |
| // as provided by CreateGraphs. |
| void CreateMatlabScript(const std::string& script_name) const; |
| |
| private: |
| struct TimingData { |
| explicit TimingData(double at) : arrival_time_ms(at) {} |
| double arrival_time_ms; |
| rtc::Optional<int64_t> decode_get_audio_count; |
| rtc::Optional<int64_t> sync_delay_ms; |
| rtc::Optional<int> target_delay_ms; |
| rtc::Optional<int> current_delay_ms; |
| }; |
| std::map<uint32_t, TimingData> data_; |
| std::vector<int64_t> get_audio_time_ms_; |
| size_t get_audio_count_ = 0; |
| size_t last_sync_buffer_ms_ = 0; |
| int last_sample_rate_hz_ = 0; |
| std::set<uint32_t> ssrcs_; |
| std::set<int> payload_types_; |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_ |