Add pcm16b quality test supporting 48khz.
Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68
Bug: webrtc:8836
Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68
Reviewed-on: https://webrtc-review.googlesource.com/47400
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21878}diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 96d97eb..4ed6377 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1303,6 +1303,7 @@
":neteq_ilbc_quality_test",
":neteq_isac_quality_test",
":neteq_opus_quality_test",
+ ":neteq_pcm16b_quality_test",
":neteq_pcmu_quality_test",
":neteq_speed_test",
":rtp_analyze",
@@ -1895,6 +1896,24 @@
]
}
+ rtc_executable("neteq_pcm16b_quality_test") {
+ testonly = true
+
+ sources = [
+ "neteq/test/neteq_pcm16b_quality_test.cc",
+ ]
+
+ deps = [
+ ":neteq",
+ ":neteq_quality_test_support",
+ ":pcm16b",
+ "../../rtc_base:checks",
+ "../../rtc_base:rtc_base_approved",
+ "../../test:test_main",
+ "//testing/gtest",
+ ]
+ }
+
rtc_executable("isac_fix_test") {
testonly = true
diff --git a/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
new file mode 100644
index 0000000..8872b94
--- /dev/null
+++ b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2018 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 <memory>
+
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/flags.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/fileutils.h"
+
+using testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 48;
+static const int kOutputSampleRateKhz = 48;
+
+DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+} // namespace
+
+class NetEqPcm16bQualityTest : public NetEqQualityTest {
+ protected:
+ NetEqPcm16bQualityTest()
+ : NetEqQualityTest(FLAG_frame_size_ms,
+ kInputSampleRateKhz,
+ kOutputSampleRateKhz,
+ NetEqDecoder::kDecoderPCM16Bswb48kHz) {
+ // Flag validation
+ RTC_CHECK(FLAG_frame_size_ms >= 10 && FLAG_frame_size_ms <= 60 &&
+ (FLAG_frame_size_ms % 10) == 0)
+ << "Invalid frame size, should be 10, 20, ..., 60 ms.";
+ }
+
+ void SetUp() override {
+ AudioEncoderPcm16B::Config config;
+ config.frame_size_ms = FLAG_frame_size_ms;
+ config.sample_rate_hz = 48000;
+ config.num_channels = channels_;
+ encoder_.reset(new AudioEncoderPcm16B(config));
+ NetEqQualityTest::SetUp();
+ }
+
+ int EncodeBlock(int16_t* in_data,
+ size_t block_size_samples,
+ rtc::Buffer* payload,
+ size_t max_bytes) override {
+ const size_t kFrameSizeSamples = 480; // Samples per 10 ms.
+ size_t encoded_samples = 0;
+ uint32_t dummy_timestamp = 0;
+ AudioEncoder::EncodedInfo info;
+ do {
+ info = encoder_->Encode(dummy_timestamp,
+ rtc::ArrayView<const int16_t>(
+ in_data + encoded_samples, kFrameSizeSamples),
+ payload);
+ encoded_samples += kFrameSizeSamples;
+ } while (info.encoded_bytes == 0);
+ return rtc::checked_cast<int>(info.encoded_bytes);
+ }
+
+ private:
+ std::unique_ptr<AudioEncoderPcm16B> encoder_;
+};
+
+TEST_F(NetEqPcm16bQualityTest, Test) {
+ Simulate();
+}
+
+} // namespace test
+} // namespace webrtc
