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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}
51 lines
1.4 KiB
C++
51 lines
1.4 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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#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
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#include <assert.h>
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#include <stdio.h>
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#include <string>
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#include "modules/audio_coding/neteq/tools/audio_sink.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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namespace test {
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class OutputAudioFile : public AudioSink {
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public:
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// Creates an OutputAudioFile, opening a file named |file_name| for writing.
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// The file format is 16-bit signed host-endian PCM.
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explicit OutputAudioFile(const std::string& file_name) {
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out_file_ = fopen(file_name.c_str(), "wb");
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}
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virtual ~OutputAudioFile() {
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if (out_file_)
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fclose(out_file_);
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}
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bool WriteArray(const int16_t* audio, size_t num_samples) override {
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assert(out_file_);
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return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
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}
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private:
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FILE* out_file_;
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RTC_DISALLOW_COPY_AND_ASSIGN(OutputAudioFile);
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
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