neteq_test_factory.h

/*

 *  Copyright (c) 2018 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_TEST_FACTORY_H_

#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_

#include <memory>

#include <optional>

#include <string>

#include "absl/strings/string_view.h"

#include "modules/audio_coding/neteq/tools/neteq_test.h"

namespace webrtc {

namespace test {

class SsrcSwitchDetector;

class NetEqStatsGetter;

class NetEqStatsPlotter;

// Note that the NetEqTestFactory needs to be alive when the NetEqTest object is

// used for a simulation.

class NetEqTestFactory {

 public:

  NetEqTestFactory();

  ~NetEqTestFactory();

  struct Config {

    Config();

    Config(const Config& other);

    ~Config();

    // RTP payload type for PCM-u.

    static constexpr int default_pcmu() { return 0; }

    int pcmu = default_pcmu();

    // RTP payload type for PCM-a.

    static constexpr int default_pcma() { return 8; }

    int pcma = default_pcma();

    // RTP payload type for iLBC.

    static constexpr int default_ilbc() { return 102; }

    int ilbc = default_ilbc();

    // RTP payload type for iSAC.

    static constexpr int default_isac() { return 103; }

    int isac = default_isac();

    // RTP payload type for iSAC-swb (32 kHz).

    static constexpr int default_isac_swb() { return 104; }

    int isac_swb = default_isac_swb();

    // RTP payload type for Opus.

    static constexpr int default_opus() { return 111; }

    int opus = default_opus();

    // RTP payload type for PCM16b-nb (8 kHz).

    static constexpr int default_pcm16b() { return 93; }

    int pcm16b = default_pcm16b();

    // RTP payload type for PCM16b-wb (16 kHz).

    static constexpr int default_pcm16b_wb() { return 94; }

    int pcm16b_wb = default_pcm16b_wb();

    // RTP payload type for PCM16b-swb32 (32 kHz).

    static constexpr int default_pcm16b_swb32() { return 95; }

    int pcm16b_swb32 = default_pcm16b_swb32();

    // RTP payload type for PCM16b-swb48 (48 kHz).

    static constexpr int default_pcm16b_swb48() { return 96; }

    int pcm16b_swb48 = default_pcm16b_swb48();

    // RTP payload type for G.722.

    static constexpr int default_g722() { return 9; }

    int g722 = default_g722();

    // RTP payload type for AVT/DTMF (8 kHz).

    static constexpr int default_avt() { return 106; }

    int avt = default_avt();

    // RTP payload type for AVT/DTMF (16 kHz).

    static constexpr int default_avt_16() { return 114; }

    int avt_16 = default_avt_16();

    // RTP payload type for AVT/DTMF (32 kHz).

    static constexpr int default_avt_32() { return 115; }

    int avt_32 = default_avt_32();

    // RTP payload type for AVT/DTMF (48 kHz).

    static constexpr int default_avt_48() { return 116; }

    int avt_48 = default_avt_48();

    // RTP payload type for redundant audio (RED).

    static constexpr int default_red() { return 117; }

    int red = default_red();

    // RTP payload type for comfort noise (8 kHz).

    static constexpr int default_cn_nb() { return 13; }

    int cn_nb = default_cn_nb();

    // RTP payload type for comfort noise (16 kHz).

    static constexpr int default_cn_wb() { return 98; }

    int cn_wb = default_cn_wb();

    // RTP payload type for comfort noise (32 kHz).

    static constexpr int default_cn_swb32() { return 99; }

    int cn_swb32 = default_cn_swb32();

    // RTP payload type for comfort noise (48 kHz).

    static constexpr int default_cn_swb48() { return 100; }

    int cn_swb48 = default_cn_swb48();

    // A PCM file that will be used to populate dummy RTP packets.

    std::string replacement_audio_file;

    // Only use packets with this SSRC.

    std::optional<uint32_t> ssrc_filter;

    // Extension ID for audio level (RFC 6464).

    static constexpr int default_audio_level() { return 1; }

    int audio_level = default_audio_level();

    // Extension ID for absolute sender time.

    static constexpr int default_abs_send_time() { return 3; }

    int abs_send_time = default_abs_send_time();

    // Extension ID for transport sequence number.

    static constexpr int default_transport_seq_no() { return 5; }

    int transport_seq_no = default_transport_seq_no();

    // Extension ID for video content type.

    static constexpr int default_video_content_type() { return 7; }

    int video_content_type = default_video_content_type();

    // Extension ID for video timing.

    static constexpr int default_video_timing() { return 8; }

    int video_timing = default_video_timing();

    // Generate a matlab script for plotting the delay profile.

    bool matlabplot = false;

    // Generates a python script for plotting the delay profile.

    bool pythonplot = false;

    // Prints concealment events.

    bool concealment_events = false;

    // Maximum allowed number of packets in the buffer.

    static constexpr int default_max_nr_packets_in_buffer() { return 200; }

    int max_nr_packets_in_buffer = default_max_nr_packets_in_buffer();

    // Number of dummy packets to put in the packet buffer at the start of the

    // simulation.

    static constexpr int default_initial_dummy_packets() { return 0; }

    int initial_dummy_packets = default_initial_dummy_packets();

    // Number of getAudio events to skip at the start of the simulation.

    static constexpr int default_skip_get_audio_events() { return 0; }

    int skip_get_audio_events = default_skip_get_audio_events();

    // Enables jitter buffer fast accelerate.

    bool enable_fast_accelerate = false;

    // Dumps events that describes the simulation on a step-by-step basis.

    bool textlog = false;

    // If specified and `textlog` is true, the output of `textlog` is written to

    // the specified file name.

    std::optional<std::string> textlog_filename;

    // Base name for the output script files for plotting the delay profile.

    std::optional<std::string> plot_scripts_basename;

    // Path to the output audio file.

    std::optional<std::string> output_audio_filename;

    // Field trials to use during the simulation.

    std::string field_trial_string;

};

  std::unique_ptr<NetEqTest> InitializeTestFromFile(

      absl::string_view input_filename,

      NetEqFactory* neteq_factory,

      const Config& config);

  std::unique_ptr<NetEqTest> InitializeTestFromString(

      absl::string_view input_string,

      NetEqFactory* neteq_factory,

      const Config& config);

 private:

  std::unique_ptr<NetEqTest> InitializeTest(std::unique_ptr<NetEqInput> input,

                                            NetEqFactory* neteq_factory,

                                            const Config& config);

  std::unique_ptr<SsrcSwitchDetector> ssrc_switch_detector_;

  std::unique_ptr<NetEqStatsPlotter> stats_plotter_;

};

}  // namespace test

}  // namespace webrtc

#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_