neteq_performance_test.cc

/*

 *  Copyright (c) 2014 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 "modules/audio_coding/neteq/tools/neteq_performance_test.h"

#include "api/audio/audio_frame.h"

#include "api/audio_codecs/builtin_audio_decoder_factory.h"

#include "api/environment/environment.h"

#include "api/environment/environment_factory.h"

#include "api/neteq/neteq.h"

#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"

#include "modules/audio_coding/neteq/default_neteq_factory.h"

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

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

#include "rtc_base/checks.h"

#include "system_wrappers/include/clock.h"

#include "test/testsupport/file_utils.h"

using webrtc::NetEq;

using webrtc::test::AudioLoop;

using webrtc::test::RtpGenerator;

namespace webrtc {

namespace test {

int64_t NetEqPerformanceTest::Run(int runtime_ms,

                                  int lossrate,

                                  double drift_factor) {

  const std::string kInputFileName =

      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");

  const int kSampRateHz = 32000;

  const std::string kDecoderName = "pcm16-swb32";

  const int kPayloadType = 95;

  // Initialize NetEq instance.

  NetEq::Config config;

  config.sample_rate_hz = kSampRateHz;

  Environment env = CreateEnvironment();

  auto neteq = DefaultNetEqFactory().Create(env, config,

                                            CreateBuiltinAudioDecoderFactory());

  // Register decoder in `neteq`.

  if (!neteq->RegisterPayloadType(kPayloadType,

                                  SdpAudioFormat("l16", kSampRateHz, 1)))

    return -1;

  // Set up AudioLoop object.

  AudioLoop audio_loop;

  const size_t kMaxLoopLengthSamples = kSampRateHz * 10;  // 10 second loop.

  const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000;  // 60 ms.

  if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,

                       kInputBlockSizeSamples))

    return -1;

  int32_t time_now_ms = 0;

  // Get first input packet.

  RTPHeader rtp_header;

  RtpGenerator rtp_gen(kSampRateHz / 1000);

  // Start with positive drift first half of simulation.

  rtp_gen.set_drift_factor(drift_factor);

  bool drift_flipped = false;

  int32_t packet_input_time_ms =

      rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);

  auto input_samples = audio_loop.GetNextBlock();

  if (input_samples.empty())

    exit(1);

  uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];

  size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),

                                           input_samples.size(), input_payload);

  RTC_CHECK_EQ(sizeof(input_payload), payload_len);

  // Main loop.

  int64_t start_time_ms = env.clock().TimeInMilliseconds();

  AudioFrame out_frame;

  while (time_now_ms < runtime_ms) {

    while (packet_input_time_ms <= time_now_ms) {

      // Drop every N packets, where N = FLAG_lossrate.

      bool lost = false;

      if (lossrate > 0) {

        lost = ((rtp_header.sequenceNumber - 1) % lossrate) == 0;

      if (!lost) {

        // Insert packet.

        int error = neteq->InsertPacket(rtp_header, input_payload,

                                        Timestamp::Millis(time_now_ms));

        if (error != NetEq::kOK)

          return -1;

      // Get next packet.

      packet_input_time_ms = rtp_gen.GetRtpHeader(

          kPayloadType, kInputBlockSizeSamples, &rtp_header);

      input_samples = audio_loop.GetNextBlock();

      if (input_samples.empty())

        return -1;

      payload_len = WebRtcPcm16b_Encode(input_samples.data(),

                                        input_samples.size(), input_payload);

      RTC_DCHECK_EQ(payload_len, kInputBlockSizeSamples * sizeof(int16_t));

    // Get output audio, but don't do anything with it.

    bool muted;

    int error = neteq->GetAudio(&out_frame, &muted);

    RTC_CHECK(!muted);

    if (error != NetEq::kOK)

      return -1;

    RTC_DCHECK_EQ(out_frame.samples_per_channel_, (kSampRateHz * 10) / 1000);

    static const int kOutputBlockSizeMs = 10;

    time_now_ms += kOutputBlockSizeMs;

    if (time_now_ms >= runtime_ms / 2 && !drift_flipped) {

      // Apply negative drift second half of simulation.

      rtp_gen.set_drift_factor(-drift_factor);

      drift_flipped = true;

  int64_t end_time_ms = env.clock().TimeInMilliseconds();

  return end_time_ms - start_time_ms;

}  // namespace test

}  // namespace webrtc