Fix the maximum native sample rate in AudioProcessing
BUG=webrtc:4983
R=andrew@webrtc.org, henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1338833002 .
Cr-Commit-Position: refs/heads/master@{#10037}
diff --git a/webrtc/voice_engine/utility_unittest.cc b/webrtc/voice_engine/utility_unittest.cc
index 5f02f51..226e383 100644
--- a/webrtc/voice_engine/utility_unittest.cc
+++ b/webrtc/voice_engine/utility_unittest.cc
@@ -21,11 +21,6 @@
namespace voe {
namespace {
-enum FunctionToTest {
- TestRemixAndResample,
- TestDownConvertToCodecFormat
-};
-
class UtilityTest : public ::testing::Test {
protected:
UtilityTest() {
@@ -36,9 +31,10 @@
golden_frame_.CopyFrom(src_frame_);
}
- void RunResampleTest(int src_channels, int src_sample_rate_hz,
- int dst_channels, int dst_sample_rate_hz,
- FunctionToTest function);
+ void RunResampleTest(int src_channels,
+ int src_sample_rate_hz,
+ int dst_channels,
+ int dst_sample_rate_hz);
PushResampler<int16_t> resampler_;
AudioFrame src_frame_;
@@ -130,8 +126,7 @@
void UtilityTest::RunResampleTest(int src_channels,
int src_sample_rate_hz,
int dst_channels,
- int dst_sample_rate_hz,
- FunctionToTest function) {
+ int dst_sample_rate_hz) {
PushResampler<int16_t> resampler; // Create a new one with every test.
const int16_t kSrcLeft = 30; // Shouldn't overflow for any used sample rate.
const int16_t kSrcRight = 15;
@@ -168,20 +163,7 @@
kInputKernelDelaySamples * dst_channels * 2);
printf("(%d, %d Hz) -> (%d, %d Hz) ", // SNR reported on the same line later.
src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
- if (function == TestRemixAndResample) {
- RemixAndResample(src_frame_, &resampler, &dst_frame_);
- } else {
- int16_t mono_buffer[kMaxMonoDataSizeSamples];
- DownConvertToCodecFormat(src_frame_.data_,
- src_frame_.samples_per_channel_,
- src_frame_.num_channels_,
- src_frame_.sample_rate_hz_,
- dst_frame_.num_channels_,
- dst_frame_.sample_rate_hz_,
- mono_buffer,
- &resampler,
- &dst_frame_);
- }
+ RemixAndResample(src_frame_, &resampler, &dst_frame_);
if (src_sample_rate_hz == 96000 && dst_sample_rate_hz == 8000) {
// The sinc resampler gives poor SNR at this extreme conversion, but we
@@ -232,28 +214,7 @@
for (int src_channel = 0; src_channel < kChannelsSize; src_channel++) {
for (int dst_channel = 0; dst_channel < kChannelsSize; dst_channel++) {
RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
- kChannels[dst_channel], kSampleRates[dst_rate],
- TestRemixAndResample);
- }
- }
- }
- }
-}
-
-TEST_F(UtilityTest, ConvertToCodecFormatSucceeds) {
- const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000, 96000};
- const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
- const int kChannels[] = {1, 2};
- const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
- for (int src_rate = 0; src_rate < kSampleRatesSize; src_rate++) {
- for (int dst_rate = 0; dst_rate < kSampleRatesSize; dst_rate++) {
- for (int src_channel = 0; src_channel < kChannelsSize; src_channel++) {
- for (int dst_channel = 0; dst_channel < kChannelsSize; dst_channel++) {
- if (dst_rate <= src_rate && dst_channel <= src_channel) {
- RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
- kChannels[src_channel], kSampleRates[dst_rate],
- TestDownConvertToCodecFormat);
- }
+ kChannels[dst_channel], kSampleRates[dst_rate]);
}
}
}
