/* * 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 MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_ #include #include #include #include #include "absl/types/optional.h" #include "modules/audio_coding/neteq/tools/neteq_input.h" #include "modules/audio_coding/neteq/tools/neteq_test.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; using Delays = std::vector>; void CreateGraphs(Delays* arrival_delay_ms, Delays* corrected_arrival_delay_ms, Delays* playout_delay_ms, Delays* 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; // Creates a python script with file name |script_name|. When executed in // Python, the script will generate graphs with the same timing information // as provided by CreateGraphs. void CreatePythonScript(const std::string& script_name) const; private: struct TimingData { explicit TimingData(int64_t at) : arrival_time_ms(at) {} int64_t arrival_time_ms; absl::optional decode_get_audio_count; absl::optional sync_delay_ms; absl::optional target_delay_ms; absl::optional current_delay_ms; }; std::map data_; std::vector 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 ssrcs_; std::set payload_types_; }; } // namespace test } // namespace webrtc #endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_