Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 423 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc
new file mode 100644
index 0000000000..e6c1809fb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc
@@ -0,0 +1,423 @@
+/*
+ *  Copyright (c) 2011 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/test/neteq_decoding_test.h"
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/rtp_headers.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/test/result_sink.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/testsupport/file_utils.h"
+
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
+#else
+#include "modules/audio_coding/neteq/neteq_unittest.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+namespace {
+
+void LoadDecoders(webrtc::NetEq* neteq) {
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(0, SdpAudioFormat("pcmu", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(8, SdpAudioFormat("pcma", 8000, 1)));
+#ifdef WEBRTC_CODEC_ILBC
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(102, SdpAudioFormat("ilbc", 8000, 1)));
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(103, SdpAudioFormat("isac", 16000, 1)));
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(104, SdpAudioFormat("isac", 32000, 1)));
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(
+                111, SdpAudioFormat("opus", 48000, 2, {{"stereo", "0"}})));
+#endif
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(93, SdpAudioFormat("L16", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(94, SdpAudioFormat("L16", 16000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(95, SdpAudioFormat("L16", 32000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(13, SdpAudioFormat("cn", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(98, SdpAudioFormat("cn", 16000, 1)));
+}
+
+}  // namespace
+
+const int NetEqDecodingTest::kTimeStepMs;
+const size_t NetEqDecodingTest::kBlockSize8kHz;
+const size_t NetEqDecodingTest::kBlockSize16kHz;
+const size_t NetEqDecodingTest::kBlockSize32kHz;
+const int NetEqDecodingTest::kInitSampleRateHz;
+
+NetEqDecodingTest::NetEqDecodingTest()
+    : clock_(0),
+      config_(),
+      output_sample_rate_(kInitSampleRateHz),
+      algorithmic_delay_ms_(0) {
+  config_.sample_rate_hz = kInitSampleRateHz;
+}
+
+void NetEqDecodingTest::SetUp() {
+  auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+  neteq_ = DefaultNetEqFactory().CreateNetEq(config_, decoder_factory, &clock_);
+  NetEqNetworkStatistics stat;
+  ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
+  algorithmic_delay_ms_ = stat.current_buffer_size_ms;
+  ASSERT_TRUE(neteq_);
+  LoadDecoders(neteq_.get());
+}
+
+void NetEqDecodingTest::TearDown() {}
+
+void NetEqDecodingTest::OpenInputFile(absl::string_view rtp_file) {
+  rtp_source_.reset(test::RtpFileSource::Create(rtp_file));
+}
+
+void NetEqDecodingTest::Process() {
+  // Check if time to receive.
+  while (packet_ && clock_.TimeInMilliseconds() >= packet_->time_ms()) {
+    if (packet_->payload_length_bytes() > 0) {
+#ifndef WEBRTC_CODEC_ISAC
+      // Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
+      if (packet_->header().payloadType != 104)
+#endif
+        ASSERT_EQ(
+            0, neteq_->InsertPacket(
+                   packet_->header(),
+                   rtc::ArrayView<const uint8_t>(
+                       packet_->payload(), packet_->payload_length_bytes())));
+    }
+    // Get next packet.
+    packet_ = rtp_source_->NextPacket();
+  }
+
+  // Get audio from NetEq.
+  bool muted;
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_FALSE(muted);
+  ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize16kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize32kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize48kHz));
+  output_sample_rate_ = out_frame_.sample_rate_hz_;
+  EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
+
+  // Increase time.
+  clock_.AdvanceTimeMilliseconds(kTimeStepMs);
+}
+
+void NetEqDecodingTest::DecodeAndCompare(
+    absl::string_view rtp_file,
+    absl::string_view output_checksum,
+    absl::string_view network_stats_checksum,
+    bool gen_ref) {
+  OpenInputFile(rtp_file);
+
+  std::string ref_out_file =
+      gen_ref ? webrtc::test::OutputPath() + "neteq_universal_ref.pcm" : "";
+  ResultSink output(ref_out_file);
+
+  std::string stat_out_file =
+      gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : "";
+  ResultSink network_stats(stat_out_file);
+
+  packet_ = rtp_source_->NextPacket();
+  int i = 0;
+  uint64_t last_concealed_samples = 0;
+  uint64_t last_total_samples_received = 0;
+  while (packet_) {
+    rtc::StringBuilder ss;
+    ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
+    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+    ASSERT_NO_FATAL_FAILURE(Process());
+    ASSERT_NO_FATAL_FAILURE(
+        output.AddResult(out_frame_.data(), out_frame_.samples_per_channel_));
+
+    // Query the network statistics API once per second
+    if (clock_.TimeInMilliseconds() % 1000 == 0) {
+      // Process NetworkStatistics.
+      NetEqNetworkStatistics current_network_stats;
+      ASSERT_EQ(0, neteq_->NetworkStatistics(&current_network_stats));
+      ASSERT_NO_FATAL_FAILURE(network_stats.AddResult(current_network_stats));
+
+      // Verify that liftime stats and network stats report similar loss
+      // concealment rates.
+      auto lifetime_stats = neteq_->GetLifetimeStatistics();
+      const uint64_t delta_concealed_samples =
+          lifetime_stats.concealed_samples - last_concealed_samples;
+      last_concealed_samples = lifetime_stats.concealed_samples;
+      const uint64_t delta_total_samples_received =
+          lifetime_stats.total_samples_received - last_total_samples_received;
+      last_total_samples_received = lifetime_stats.total_samples_received;
+      // The tolerance is 1% but expressed in Q14.
+      EXPECT_NEAR(
+          (delta_concealed_samples << 14) / delta_total_samples_received,
+          current_network_stats.expand_rate, (2 << 14) / 100.0);
+    }
+  }
+
+  SCOPED_TRACE("Check output audio.");
+  output.VerifyChecksum(output_checksum);
+  SCOPED_TRACE("Check network stats.");
+  network_stats.VerifyChecksum(network_stats_checksum);
+}
+
+void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
+                                        int timestamp,
+                                        RTPHeader* rtp_info) {
+  rtp_info->sequenceNumber = frame_index;
+  rtp_info->timestamp = timestamp;
+  rtp_info->ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info->payloadType = 94;  // PCM16b WB codec.
+  rtp_info->markerBit = false;
+}
+
+void NetEqDecodingTest::PopulateCng(int frame_index,
+                                    int timestamp,
+                                    RTPHeader* rtp_info,
+                                    uint8_t* payload,
+                                    size_t* payload_len) {
+  rtp_info->sequenceNumber = frame_index;
+  rtp_info->timestamp = timestamp;
+  rtp_info->ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info->payloadType = 98;  // WB CNG.
+  rtp_info->markerBit = false;
+  payload[0] = 64;   // Noise level -64 dBov, quite arbitrarily chosen.
+  *payload_len = 1;  // Only noise level, no spectral parameters.
+}
+
+void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
+                                 uint32_t start_timestamp,
+                                 const std::set<uint16_t>& drop_seq_numbers,
+                                 bool expect_seq_no_wrap,
+                                 bool expect_timestamp_wrap) {
+  uint16_t seq_no = start_seq_no;
+  uint32_t timestamp = start_timestamp;
+  const int kBlocksPerFrame = 3;  // Number of 10 ms blocks per frame.
+  const int kFrameSizeMs = kBlocksPerFrame * kTimeStepMs;
+  const int kSamples = kBlockSize16kHz * kBlocksPerFrame;
+  const size_t kPayloadBytes = kSamples * sizeof(int16_t);
+  double next_input_time_ms = 0.0;
+
+  // Insert speech for 2 seconds.
+  const int kSpeechDurationMs = 2000;
+  uint16_t last_seq_no;
+  uint32_t last_timestamp;
+  bool timestamp_wrapped = false;
+  bool seq_no_wrapped = false;
+  for (double t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) {
+        // This sequence number was not in the set to drop. Insert it.
+        ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      }
+      NetEqNetworkStatistics network_stats;
+      ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
+
+      EXPECT_LE(network_stats.preferred_buffer_size_ms, 80);
+      EXPECT_LE(network_stats.current_buffer_size_ms,
+                80 + algorithmic_delay_ms_);
+      last_seq_no = seq_no;
+      last_timestamp = timestamp;
+
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += static_cast<double>(kFrameSizeMs);
+
+      seq_no_wrapped |= seq_no < last_seq_no;
+      timestamp_wrapped |= timestamp < last_timestamp;
+    }
+    // Pull out data once.
+    AudioFrame output;
+    bool muted;
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+    ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
+    ASSERT_EQ(1u, output.num_channels_);
+
+    // Expect delay (in samples) to be less than 2 packets.
+    absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+    ASSERT_TRUE(playout_timestamp);
+    EXPECT_LE(timestamp - *playout_timestamp,
+              static_cast<uint32_t>(kSamples * 2));
+  }
+  // Make sure we have actually tested wrap-around.
+  ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
+  ASSERT_EQ(expect_timestamp_wrap, timestamp_wrapped);
+}
+
+void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
+                                              double network_freeze_ms,
+                                              bool pull_audio_during_freeze,
+                                              int delay_tolerance_ms,
+                                              int max_time_to_speech_ms) {
+  uint16_t seq_no = 0;
+  uint32_t timestamp = 0;
+  const int kFrameSizeMs = 30;
+  const size_t kSamples = kFrameSizeMs * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  double next_input_time_ms = 0.0;
+  double t_ms;
+  bool muted;
+
+  // Insert speech for 5 seconds.
+  const int kSpeechDurationMs = 5000;
+  for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+  absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+  ASSERT_TRUE(playout_timestamp);
+  int32_t delay_before = timestamp - *playout_timestamp;
+
+  // Insert CNG for 1 minute (= 60000 ms).
+  const int kCngPeriodMs = 100;
+  const int kCngPeriodSamples = kCngPeriodMs * 16;  // Period in 16 kHz samples.
+  const int kCngDurationMs = 60000;
+  for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one CNG frame each 100 ms.
+      uint8_t payload[kPayloadBytes];
+      size_t payload_len;
+      RTPHeader rtp_info;
+      PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, payload_len)));
+      ++seq_no;
+      timestamp += kCngPeriodSamples;
+      next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+
+  if (network_freeze_ms > 0) {
+    // First keep pulling audio for `network_freeze_ms` without inserting
+    // any data, then insert CNG data corresponding to `network_freeze_ms`
+    // without pulling any output audio.
+    const double loop_end_time = t_ms + network_freeze_ms;
+    for (; t_ms < loop_end_time; t_ms += 10) {
+      // Pull out data once.
+      ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+      ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+      EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+    }
+    bool pull_once = pull_audio_during_freeze;
+    // If `pull_once` is true, GetAudio will be called once half-way through
+    // the network recovery period.
+    double pull_time_ms = (t_ms + next_input_time_ms) / 2;
+    while (next_input_time_ms <= t_ms) {
+      if (pull_once && next_input_time_ms >= pull_time_ms) {
+        pull_once = false;
+        // Pull out data once.
+        ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+        ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+        EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+        t_ms += 10;
+      }
+      // Insert one CNG frame each 100 ms.
+      uint8_t payload[kPayloadBytes];
+      size_t payload_len;
+      RTPHeader rtp_info;
+      PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, payload_len)));
+      ++seq_no;
+      timestamp += kCngPeriodSamples;
+      next_input_time_ms += kCngPeriodMs * drift_factor;
+    }
+  }
+
+  // Insert speech again until output type is speech.
+  double speech_restart_time_ms = t_ms;
+  while (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += kFrameSizeMs * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+    // Increase clock.
+    t_ms += 10;
+  }
+
+  // Check that the speech starts again within reasonable time.
+  double time_until_speech_returns_ms = t_ms - speech_restart_time_ms;
+  EXPECT_LT(time_until_speech_returns_ms, max_time_to_speech_ms);
+  playout_timestamp = neteq_->GetPlayoutTimestamp();
+  ASSERT_TRUE(playout_timestamp);
+  int32_t delay_after = timestamp - *playout_timestamp;
+  // Compare delay before and after, and make sure it differs less than 20 ms.
+  EXPECT_LE(delay_after, delay_before + delay_tolerance_ms * 16);
+  EXPECT_GE(delay_after, delay_before - delay_tolerance_ms * 16);
+}
+
+void NetEqDecodingTestTwoInstances::SetUp() {
+  NetEqDecodingTest::SetUp();
+  config2_ = config_;
+}
+
+void NetEqDecodingTestTwoInstances::CreateSecondInstance() {
+  auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+  neteq2_ =
+      DefaultNetEqFactory().CreateNetEq(config2_, decoder_factory, &clock_);
+  ASSERT_TRUE(neteq2_);
+  LoadDecoders(neteq2_.get());
+}
+
+}  // namespace webrtc