1 files changed, 183 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
new file mode 100644
index 0000000000..5a2df24ef6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
@@ -0,0 +1,183 @@
+/*
+ *  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 "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+
+ABSL_FLAG(int, bit_rate_kbps, 32, "Target bit rate (kbps).");
+
+ABSL_FLAG(int,
+          complexity,
+          10,
+          "Complexity: 0 ~ 10 -- defined as in Opus"
+          "specification.");
+
+ABSL_FLAG(int, maxplaybackrate, 48000, "Maximum playback rate (Hz).");
+
+ABSL_FLAG(int, application, 0, "Application mode: 0 -- VOIP, 1 -- Audio.");
+
+ABSL_FLAG(int, reported_loss_rate, 10, "Reported percentile of packet loss.");
+
+ABSL_FLAG(bool, fec, false, "Enable FEC for encoding (-nofec to disable).");
+
+ABSL_FLAG(bool, dtx, false, "Enable DTX for encoding (-nodtx to disable).");
+
+ABSL_FLAG(int, sub_packets, 1, "Number of sub packets to repacketize.");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+
+static const int kOpusBlockDurationMs = 20;
+static const int kOpusSamplingKhz = 48;
+}  // namespace
+
+class NetEqOpusQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqOpusQualityTest();
+  void SetUp() override;
+  void TearDown() override;
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override;
+
+ private:
+  WebRtcOpusEncInst* opus_encoder_;
+  OpusRepacketizer* repacketizer_;
+  size_t sub_block_size_samples_;
+  int bit_rate_kbps_;
+  bool fec_;
+  bool dtx_;
+  int complexity_;
+  int maxplaybackrate_;
+  int target_loss_rate_;
+  int sub_packets_;
+  int application_;
+};
+
+NetEqOpusQualityTest::NetEqOpusQualityTest()
+    : NetEqQualityTest(kOpusBlockDurationMs * absl::GetFlag(FLAGS_sub_packets),
+                       kOpusSamplingKhz,
+                       kOpusSamplingKhz,
+                       SdpAudioFormat("opus", 48000, 2)),
+      opus_encoder_(NULL),
+      repacketizer_(NULL),
+      sub_block_size_samples_(
+          static_cast<size_t>(kOpusBlockDurationMs * kOpusSamplingKhz)),
+      bit_rate_kbps_(absl::GetFlag(FLAGS_bit_rate_kbps)),
+      fec_(absl::GetFlag(FLAGS_fec)),
+      dtx_(absl::GetFlag(FLAGS_dtx)),
+      complexity_(absl::GetFlag(FLAGS_complexity)),
+      maxplaybackrate_(absl::GetFlag(FLAGS_maxplaybackrate)),
+      target_loss_rate_(absl::GetFlag(FLAGS_reported_loss_rate)),
+      sub_packets_(absl::GetFlag(FLAGS_sub_packets)) {
+  // Flag validation
+  RTC_CHECK(absl::GetFlag(FLAGS_bit_rate_kbps) >= 6 &&
+            absl::GetFlag(FLAGS_bit_rate_kbps) <= 510)
+      << "Invalid bit rate, should be between 6 and 510 kbps.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_complexity) >= -1 &&
+            absl::GetFlag(FLAGS_complexity) <= 10)
+      << "Invalid complexity setting, should be between 0 and 10.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_application) == 0 ||
+            absl::GetFlag(FLAGS_application) == 1)
+      << "Invalid application mode, should be 0 or 1.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_reported_loss_rate) >= 0 &&
+            absl::GetFlag(FLAGS_reported_loss_rate) <= 100)
+      << "Invalid packet loss percentile, should be between 0 and 100.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_sub_packets) >= 1 &&
+            absl::GetFlag(FLAGS_sub_packets) <= 3)
+      << "Invalid number of sub packets, should be between 1 and 3.";
+
+  // Redefine decoder type if input is stereo.
+  if (channels_ > 1) {
+    audio_format_ = SdpAudioFormat("opus", 48000, 2,
+                                   SdpAudioFormat::Parameters{{"stereo", "1"}});
+  }
+  application_ = absl::GetFlag(FLAGS_application);
+}
+
+void NetEqOpusQualityTest::SetUp() {
+  // Create encoder memory.
+  WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_, 48000);
+  ASSERT_TRUE(opus_encoder_);
+
+  // Create repacketizer.
+  repacketizer_ = opus_repacketizer_create();
+  ASSERT_TRUE(repacketizer_);
+
+  // Set bitrate.
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_kbps_ * 1000));
+  if (fec_) {
+    EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
+  }
+  if (dtx_) {
+    EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
+  }
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity_));
+  EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, maxplaybackrate_));
+  EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, target_loss_rate_));
+  NetEqQualityTest::SetUp();
+}
+
+void NetEqOpusQualityTest::TearDown() {
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  opus_repacketizer_destroy(repacketizer_);
+  NetEqQualityTest::TearDown();
+}
+
+int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
+                                      size_t block_size_samples,
+                                      rtc::Buffer* payload,
+                                      size_t max_bytes) {
+  EXPECT_EQ(block_size_samples, sub_block_size_samples_ * sub_packets_);
+  int16_t* pointer = in_data;
+  int value;
+  opus_repacketizer_init(repacketizer_);
+  for (int idx = 0; idx < sub_packets_; idx++) {
+    payload->AppendData(max_bytes, [&](rtc::ArrayView<uint8_t> payload) {
+      value = WebRtcOpus_Encode(opus_encoder_, pointer, sub_block_size_samples_,
+                                max_bytes, payload.data());
+
+      Log() << "Encoded a frame with Opus mode "
+            << (value == 0 ? 0 : payload[0] >> 3) << std::endl;
+
+      return (value >= 0) ? static_cast<size_t>(value) : 0;
+    });
+
+    if (OPUS_OK !=
+        opus_repacketizer_cat(repacketizer_, payload->data(), value)) {
+      opus_repacketizer_init(repacketizer_);
+      // If the repacketization fails, we discard this frame.
+      return 0;
+    }
+    pointer += sub_block_size_samples_ * channels_;
+  }
+  value = opus_repacketizer_out(repacketizer_, payload->data(),
+                                static_cast<opus_int32>(max_bytes));
+  EXPECT_GE(value, 0);
+  return value;
+}
+
+TEST_F(NetEqOpusQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc