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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 01:13:27 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 01:13:27 +0000
commit40a355a42d4a9444dc753c04c6608dade2f06a23 (patch)
tree871fc667d2de662f171103ce5ec067014ef85e61 /dom/media/gtest/TestAudioSegment.cpp
parentAdding upstream version 124.0.1. (diff)
downloadfirefox-40a355a42d4a9444dc753c04c6608dade2f06a23.tar.xz
firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.zip
Adding upstream version 125.0.1.upstream/125.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'dom/media/gtest/TestAudioSegment.cpp')
-rw-r--r--dom/media/gtest/TestAudioSegment.cpp16
1 files changed, 9 insertions, 7 deletions
diff --git a/dom/media/gtest/TestAudioSegment.cpp b/dom/media/gtest/TestAudioSegment.cpp
index ee44839283..ea0e43f0cd 100644
--- a/dom/media/gtest/TestAudioSegment.cpp
+++ b/dom/media/gtest/TestAudioSegment.cpp
@@ -31,7 +31,7 @@ float GetLowValue<float>() {
template <>
int16_t GetLowValue<short>() {
- return -INT16_MAX;
+ return INT16_MIN;
}
template <>
@@ -62,7 +62,7 @@ const T* const* GetPlanarChannelArray(size_t aChannels, size_t aSize) {
for (size_t c = 0; c < aChannels; c++) {
channels[c] = new T[aSize];
for (size_t i = 0; i < aSize; i++) {
- channels[c][i] = FloatToAudioSample<T>(1. / (c + 1));
+ channels[c][i] = ConvertAudioSample<T>(1.f / static_cast<float>(c + 1));
}
}
return channels;
@@ -104,7 +104,7 @@ const T* GetInterleavedChannelArray(size_t aChannels, size_t aSize) {
T* samples = new T[sampleCount];
for (size_t i = 0; i < sampleCount; i++) {
uint32_t channel = (i % aChannels) + 1;
- samples[i] = FloatToAudioSample<T>(1. / channel);
+ samples[i] = ConvertAudioSample<T>(1.f / static_cast<float>(channel));
}
return samples;
}
@@ -128,8 +128,9 @@ void TestInterleaveAndConvert() {
uint32_t channelIndex = 0;
for (size_t i = 0; i < arraySize * channels; i++) {
- ASSERT_TRUE(FuzzyEqual(
- dst[i], FloatToAudioSample<DstT>(1. / (channelIndex + 1))));
+ ASSERT_TRUE(
+ FuzzyEqual(dst[i], ConvertAudioSample<DstT>(
+ 1.f / static_cast<float>(channelIndex + 1))));
channelIndex++;
channelIndex %= channels;
}
@@ -151,8 +152,9 @@ void TestDeinterleaveAndConvert() {
for (size_t channel = 0; channel < channels; channel++) {
for (size_t i = 0; i < arraySize; i++) {
- ASSERT_TRUE(FuzzyEqual(dst[channel][i],
- FloatToAudioSample<DstT>(1. / (channel + 1))));
+ ASSERT_TRUE(FuzzyEqual(
+ dst[channel][i],
+ ConvertAudioSample<DstT>(1.f / static_cast<float>(channel + 1))));
}
}