2014-04-22 21:00:04 +00:00
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/*
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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 WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
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2016-09-14 05:23:22 -07:00
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#include "webrtc/base/checks.h"
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2014-04-22 21:00:04 +00:00
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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namespace webrtc {
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Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12 16:26:35 -08:00
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static inline size_t ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
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2014-04-22 21:00:04 +00:00
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switch (layout) {
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case AudioProcessing::kMono:
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case AudioProcessing::kMonoAndKeyboard:
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return 1;
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case AudioProcessing::kStereo:
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case AudioProcessing::kStereoAndKeyboard:
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return 2;
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}
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2016-09-14 05:23:22 -07:00
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RTC_NOTREACHED();
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Misc. small cleanups.
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks
All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1534193008
Cr-Commit-Position: refs/heads/master@{#11191}
2016-01-08 13:50:27 -08:00
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return 0;
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2014-04-22 21:00:04 +00:00
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}
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
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