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}
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@ -1303,6 +1303,7 @@ if (rtc_include_tests) {
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":neteq_ilbc_quality_test",
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":neteq_isac_quality_test",
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":neteq_opus_quality_test",
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":neteq_pcm16b_quality_test",
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":neteq_pcmu_quality_test",
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":neteq_speed_test",
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":rtp_analyze",
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@ -1895,6 +1896,24 @@ if (rtc_include_tests) {
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]
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}
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rtc_executable("neteq_pcm16b_quality_test") {
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testonly = true
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sources = [
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"neteq/test/neteq_pcm16b_quality_test.cc",
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]
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deps = [
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":neteq",
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":neteq_quality_test_support",
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":pcm16b",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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"../../test:test_main",
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"//testing/gtest",
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]
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}
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rtc_executable("isac_fix_test") {
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testonly = true
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82
modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
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82
modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
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@ -0,0 +1,82 @@
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/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <memory>
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#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
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#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/flags.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "test/testsupport/fileutils.h"
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using testing::InitGoogleTest;
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namespace webrtc {
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namespace test {
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namespace {
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static const int kInputSampleRateKhz = 48;
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static const int kOutputSampleRateKhz = 48;
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DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
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} // namespace
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class NetEqPcm16bQualityTest : public NetEqQualityTest {
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protected:
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NetEqPcm16bQualityTest()
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: NetEqQualityTest(FLAG_frame_size_ms,
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kInputSampleRateKhz,
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kOutputSampleRateKhz,
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NetEqDecoder::kDecoderPCM16Bswb48kHz) {
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// Flag validation
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RTC_CHECK(FLAG_frame_size_ms >= 10 && FLAG_frame_size_ms <= 60 &&
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(FLAG_frame_size_ms % 10) == 0)
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<< "Invalid frame size, should be 10, 20, ..., 60 ms.";
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}
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void SetUp() override {
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AudioEncoderPcm16B::Config config;
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config.frame_size_ms = FLAG_frame_size_ms;
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config.sample_rate_hz = 48000;
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config.num_channels = channels_;
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encoder_.reset(new AudioEncoderPcm16B(config));
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NetEqQualityTest::SetUp();
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}
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int EncodeBlock(int16_t* in_data,
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size_t block_size_samples,
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rtc::Buffer* payload,
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size_t max_bytes) override {
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const size_t kFrameSizeSamples = 480; // Samples per 10 ms.
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size_t encoded_samples = 0;
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uint32_t dummy_timestamp = 0;
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AudioEncoder::EncodedInfo info;
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do {
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info = encoder_->Encode(dummy_timestamp,
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rtc::ArrayView<const int16_t>(
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in_data + encoded_samples, kFrameSizeSamples),
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payload);
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encoded_samples += kFrameSizeSamples;
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} while (info.encoded_bytes == 0);
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return rtc::checked_cast<int>(info.encoded_bytes);
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}
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private:
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std::unique_ptr<AudioEncoderPcm16B> encoder_;
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};
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TEST_F(NetEqPcm16bQualityTest, Test) {
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Simulate();
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}
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} // namespace test
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} // namespace webrtc
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