Peter Kasting dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00

49 lines
1.4 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_UNITTEST_H
#define WEBRTC_COMMON_AUDIO_VAD_VAD_UNITTEST_H
#include <stddef.h> // size_t
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/typedefs.h"
namespace {
// Modes we support
const int kModes[] = { 0, 1, 2, 3 };
const size_t kModesSize = sizeof(kModes) / sizeof(*kModes);
// Rates we support.
const int kRates[] = { 8000, 12000, 16000, 24000, 32000, 48000 };
const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
// Frame lengths we support.
const size_t kMaxFrameLength = 1440;
const size_t kFrameLengths[] = { 80, 120, 160, 240, 320, 480, 640, 960,
kMaxFrameLength };
const size_t kFrameLengthsSize = sizeof(kFrameLengths) / sizeof(*kFrameLengths);
} // namespace
class VadTest : public ::testing::Test {
protected:
VadTest();
virtual void SetUp();
virtual void TearDown();
// Returns true if the rate and frame length combination is valid.
bool ValidRatesAndFrameLengths(int rate, size_t frame_length);
};
#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_UNITTEST_H