1/*
2 *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
12
13#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
14#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
15#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
16#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
17#include "webrtc/system_wrappers/include/clock.h"
18#include "webrtc/test/testsupport/fileutils.h"
19#include "webrtc/typedefs.h"
20
21using webrtc::NetEq;
22using webrtc::test::AudioLoop;
23using webrtc::test::RtpGenerator;
24using webrtc::WebRtcRTPHeader;
25
26namespace webrtc {
27namespace test {
28
29int64_t NetEqPerformanceTest::Run(int runtime_ms,
30                                  int lossrate,
31                                  double drift_factor) {
32  const std::string kInputFileName =
33      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
34  const int kSampRateHz = 32000;
35  const webrtc::NetEqDecoder kDecoderType =
36      webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz;
37  const std::string kDecoderName = "pcm16-swb32";
38  const int kPayloadType = 95;
39
40  // Initialize NetEq instance.
41  NetEq::Config config;
42  config.sample_rate_hz = kSampRateHz;
43  NetEq* neteq = NetEq::Create(config);
44  // Register decoder in |neteq|.
45  if (neteq->RegisterPayloadType(kDecoderType, kDecoderName, kPayloadType) != 0)
46    return -1;
47
48  // Set up AudioLoop object.
49  AudioLoop audio_loop;
50  const size_t kMaxLoopLengthSamples = kSampRateHz * 10;  // 10 second loop.
51  const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000;  // 60 ms.
52  if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
53                       kInputBlockSizeSamples))
54    return -1;
55
56  int32_t time_now_ms = 0;
57
58  // Get first input packet.
59  WebRtcRTPHeader rtp_header;
60  RtpGenerator rtp_gen(kSampRateHz / 1000);
61  // Start with positive drift first half of simulation.
62  rtp_gen.set_drift_factor(drift_factor);
63  bool drift_flipped = false;
64  int32_t packet_input_time_ms =
65      rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
66  auto input_samples = audio_loop.GetNextBlock();
67  if (input_samples.empty())
68    exit(1);
69  uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
70  size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
71                                           input_samples.size(), input_payload);
72  RTC_CHECK_EQ(sizeof(input_payload), payload_len);
73
74  // Main loop.
75  webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
76  int64_t start_time_ms = clock->TimeInMilliseconds();
77  while (time_now_ms < runtime_ms) {
78    while (packet_input_time_ms <= time_now_ms) {
79      // Drop every N packets, where N = FLAGS_lossrate.
80      bool lost = false;
81      if (lossrate > 0) {
82        lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0;
83      }
84      if (!lost) {
85        // Insert packet.
86        int error =
87            neteq->InsertPacket(rtp_header, input_payload,
88                                packet_input_time_ms * kSampRateHz / 1000);
89        if (error != NetEq::kOK)
90          return -1;
91      }
92
93      // Get next packet.
94      packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType,
95                                                  kInputBlockSizeSamples,
96                                                  &rtp_header);
97      input_samples = audio_loop.GetNextBlock();
98      if (input_samples.empty())
99        return -1;
100      payload_len = WebRtcPcm16b_Encode(input_samples.data(),
101                                        input_samples.size(), input_payload);
102      assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
103    }
104
105    // Get output audio, but don't do anything with it.
106    static const int kMaxChannels = 1;
107    static const size_t kMaxSamplesPerMs = 48000 / 1000;
108    static const int kOutputBlockSizeMs = 10;
109    static const size_t kOutDataLen =
110        kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
111    int16_t out_data[kOutDataLen];
112    size_t num_channels;
113    size_t samples_per_channel;
114    int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
115                                &num_channels, NULL);
116    if (error != NetEq::kOK)
117      return -1;
118
119    assert(samples_per_channel == static_cast<size_t>(kSampRateHz * 10 / 1000));
120
121    time_now_ms += kOutputBlockSizeMs;
122    if (time_now_ms >= runtime_ms / 2 && !drift_flipped) {
123      // Apply negative drift second half of simulation.
124      rtp_gen.set_drift_factor(-drift_factor);
125      drift_flipped = true;
126    }
127  }
128  int64_t end_time_ms = clock->TimeInMilliseconds();
129  delete neteq;
130  return end_time_ms - start_time_ms;
131}
132
133}  // namespace test
134}  // namespace webrtc
135