blob: f56ddb7cf79c4578bf530709a689d39f7bfe1e0e [file] [log] [blame]
* 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.
#include <map>
#include <set>
#include <string>
#include <vector>
#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 {
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<std::pair<int64_t, float>>;
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;
struct TimingData {
explicit TimingData(int64_t at) : arrival_time_ms(at) {}
int64_t arrival_time_ms;
absl::optional<int64_t> decode_get_audio_count;
absl::optional<int64_t> sync_delay_ms;
absl::optional<int> target_delay_ms;
absl::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