CL partially auto-generated with:
git grep -l "\bassert(" | grep "\.[c|h]" | \
xargs sed -i 's/\bassert(/RTC_DCHECK(/g'
And with:
git grep -l "RTC_DCHECK(false)" | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'
With some manual changes to include "rtc_base/checks.h" where
needed.
A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.
The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.
This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).
Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
52 lines
1.5 KiB
C++
52 lines
1.5 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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RTC_DCHECK(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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