25ab3228f3
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}
63 lines
2.0 KiB
C++
63 lines
2.0 KiB
C++
/*
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* Copyright (c) 2012 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 "modules/audio_coding/acm2/acm_resampler.h"
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#include <assert.h>
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#include <string.h>
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#include "rtc_base/logging.h"
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namespace webrtc {
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namespace acm2 {
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ACMResampler::ACMResampler() {}
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ACMResampler::~ACMResampler() {}
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int ACMResampler::Resample10Msec(const int16_t* in_audio,
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int in_freq_hz,
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int out_freq_hz,
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size_t num_audio_channels,
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size_t out_capacity_samples,
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int16_t* out_audio) {
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size_t in_length = in_freq_hz * num_audio_channels / 100;
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if (in_freq_hz == out_freq_hz) {
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if (out_capacity_samples < in_length) {
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RTC_NOTREACHED();
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return -1;
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}
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memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
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return static_cast<int>(in_length / num_audio_channels);
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}
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if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
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num_audio_channels) != 0) {
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RTC_LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", "
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<< out_freq_hz << ", " << num_audio_channels
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<< ") failed.";
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return -1;
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}
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int out_length =
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resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
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if (out_length == -1) {
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RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
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<< out_audio << ", " << out_capacity_samples
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<< ") failed.";
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return -1;
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}
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return static_cast<int>(out_length / num_audio_channels);
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}
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} // namespace acm2
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} // namespace webrtc
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