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10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
80 lines
2.5 KiB
C++
80 lines
2.5 KiB
C++
/*
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* Copyright (c) 2015 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/g711/audio_encoder_pcm.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/file_utils.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 = 8;
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static const int kOutputSampleRateKhz = 8;
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WEBRTC_DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
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} // namespace
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class NetEqPcmuQualityTest : public NetEqQualityTest {
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protected:
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NetEqPcmuQualityTest()
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: NetEqQualityTest(FLAG_frame_size_ms,
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kInputSampleRateKhz,
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kOutputSampleRateKhz,
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SdpAudioFormat("pcmu", 8000, 1)) {
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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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ASSERT_EQ(1u, channels_) << "PCMu supports only mono audio.";
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AudioEncoderPcmU::Config config;
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config.frame_size_ms = FLAG_frame_size_ms;
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encoder_.reset(new AudioEncoderPcmU(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 = 80; // 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<AudioEncoderPcmU> encoder_;
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};
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TEST_F(NetEqPcmuQualityTest, 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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