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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
78 lines
2.2 KiB
C++
78 lines
2.2 KiB
C++
/*
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* Copyright (c) 2014 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 "test/rtp_file_writer.h"
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#include <stdint.h>
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#include <string.h>
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#include <memory>
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#include "test/gtest.h"
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#include "test/rtp_file_reader.h"
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#include "test/testsupport/file_utils.h"
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namespace webrtc {
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class RtpFileWriterTest : public ::testing::Test {
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public:
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void Init(const std::string& filename) {
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filename_ = test::OutputPath() + filename;
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rtp_writer_.reset(
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test::RtpFileWriter::Create(test::RtpFileWriter::kRtpDump, filename_));
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}
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void WriteRtpPackets(int num_packets) {
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ASSERT_TRUE(rtp_writer_.get() != NULL);
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test::RtpPacket packet;
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for (int i = 1; i <= num_packets; ++i) {
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packet.length = i;
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packet.original_length = i;
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packet.time_ms = i;
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memset(packet.data, i, packet.length);
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EXPECT_TRUE(rtp_writer_->WritePacket(&packet));
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}
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}
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void CloseOutputFile() { rtp_writer_.reset(); }
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void VerifyFileContents(int expected_packets) {
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ASSERT_TRUE(rtp_writer_.get() == NULL)
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<< "Must call CloseOutputFile before VerifyFileContents";
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std::unique_ptr<test::RtpFileReader> rtp_reader(
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test::RtpFileReader::Create(test::RtpFileReader::kRtpDump, filename_));
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ASSERT_TRUE(rtp_reader.get() != NULL);
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test::RtpPacket packet;
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int i = 0;
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while (rtp_reader->NextPacket(&packet)) {
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++i;
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EXPECT_EQ(static_cast<size_t>(i), packet.length);
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EXPECT_EQ(static_cast<size_t>(i), packet.original_length);
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EXPECT_EQ(static_cast<uint32_t>(i), packet.time_ms);
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for (int j = 0; j < i; ++j) {
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EXPECT_EQ(i, packet.data[j]);
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}
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}
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EXPECT_EQ(expected_packets, i);
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}
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private:
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std::unique_ptr<test::RtpFileWriter> rtp_writer_;
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std::string filename_;
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};
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TEST_F(RtpFileWriterTest, WriteToRtpDump) {
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Init("test_rtp_file_writer.rtp");
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WriteRtpPackets(10);
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CloseOutputFile();
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VerifyFileContents(10);
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}
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} // namespace webrtc
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