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Bugfix in AudioDecoderTest

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When the encoded frame size (L ms) was larger than 10 ms, the test would
repeat the first 10 ms L/10 times for each encoded frame. This is now
fixed.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7833 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2014-12-08 21:07:59 +00:00
parent edeea91803
commit 130fef89dd
1 changed files with 7 additions and 3 deletions
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10
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
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@ -140,15 +140,19 @@ class AudioDecoderTest : public ::testing::Test {
size_t input_len_samples,
uint8_t* output) {
size_t enc_len_bytes = 0;
const size_t samples_per_10ms = audio_encoder_->sample_rate_hz() / 100;
CHECK_EQ(samples_per_10ms * audio_encoder_->Num10MsFramesInNextPacket(),
input_len_samples);
scoped_ptr<int16_t[]> interleaved_input(
new int16_t[channels_ * input_len_samples]);
new int16_t[channels_ * samples_per_10ms]);
for (int i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {
EXPECT_EQ(0u, enc_len_bytes);
// Duplicate the mono input signal to however many channels the test
// wants.
test::InputAudioFile::DuplicateInterleaved(
input, input_len_samples, channels_, interleaved_input.get());
test::InputAudioFile::DuplicateInterleaved(input + i * samples_per_10ms,
samples_per_10ms, channels_,
interleaved_input.get());
EXPECT_TRUE(audio_encoder_->Encode(
0, interleaved_input.get(), audio_encoder_->sample_rate_hz() / 100,