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Add tests for the sample doubling converter and make it work correctly

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This commit is contained in:
Thomas Goyne 2014-07-10 10:14:54 -07:00
parent 93ad15efb1
commit f32bdff84a
2 changed files with 53 additions and 15 deletions
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32
libaegisub/audio/provider_convert.cpp
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@ -140,25 +140,27 @@ public:
}
void FillBuffer(void *buf, int64_t start, int64_t count) const override {
bool not_end = start + count < num_samples;
int64_t src_count = count / 2;
source->GetAudio(buf, start / 2, src_count + not_end);
int16_t *src, *dst = static_cast<int16_t *>(buf);
auto buf16 = reinterpret_cast<int16_t*>(buf);
if (!not_end) {
// We weren't able to request a sample past the end so just
// duplicate the last sample
buf16[src_count] = buf16[src_count + 1];
// We need to always get at least two samples to be able to interpolate
int16_t srcbuf[2];
if (count == 1) {
source->GetAudio(srcbuf, start / 2, 2);
src = srcbuf;
}
else {
source->GetAudio(buf, start / 2, (start + count) / 2 - start / 2 + 1);
src = dst;
}
if (count % 2)
buf16[count - 1] = buf16[src_count];
// walking backwards so that the conversion can be done in place
for (int64_t i = src_count - 1; i >= 0; --i) {
buf16[i * 2] = buf16[i];
buf16[i * 2 + 1] = (int16_t)(((int32_t)buf16[i] + buf16[i + 1]) / 2);
for (; count > 0; --count) {
auto src_index = (start + count - 1) / 2 - start / 2;
auto i = count - 1;
if ((start + i) & 1)
dst[i] = (int16_t)(((int32_t)src[src_index] + src[src_index + 1]) / 2);
else
dst[i] = src[src_index];
}
}
};

36
tests/tests/audio.cpp
View file

@ -200,6 +200,42 @@ TEST(lagi_audio, convert_32bit) {
EXPECT_EQ(SHRT_MAX, sample);
}
TEST(lagi_audio, sample_doubling) {
struct AudioProvider : agi::AudioProvider {
AudioProvider() {
channels = 1;
num_samples = 90 * 20000;
decoded_samples = num_samples;
sample_rate = 20000;
bytes_per_sample = 2;
float_samples = false;
}
void FillBuffer(void *buf, int64_t start, int64_t count) const override {
auto out = static_cast<int16_t *>(buf);
for (int64_t end = start + count; start < end; ++start)
*out++ = (int16_t)(start * 2);
}
};
auto provider = agi::CreateConvertAudioProvider(agi::make_unique<AudioProvider>());
EXPECT_EQ(40000, provider->GetSampleRate());
int16_t samples[6];
for (int k = 0; k < 6; ++k) {
SCOPED_TRACE(k);
for (int i = k; i < 6; ++i) {
SCOPED_TRACE(i);
memset(samples, 0, sizeof(samples));
provider->GetAudio(samples, k, i - k);
for (int j = 0; j < i - k; ++j)
EXPECT_EQ(j + k, samples[j]);
for (int j = i - k; j < 6 - k; ++j)
EXPECT_EQ(0, samples[j]);
}
}
}
TEST(lagi_audio, pcm_simple) {
auto path = agi::Path().Decode("?temp/pcm_simple");
{