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Unified Diff: webrtc/modules/audio_coding/codecs/isac/unittest.cc

Issue 1319683002: AudioDecoder: Replace Init() with Reset() (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@buffer
Patch Set: review fixes Created 5 years, 4 months ago
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Index: webrtc/modules/audio_coding/codecs/isac/unittest.cc
diff --git a/webrtc/modules/audio_coding/codecs/isac/unittest.cc b/webrtc/modules/audio_coding/codecs/isac/unittest.cc
index d05ffa6e48f646767d709052ef4a2ea9dda3eee5..673d2906ae645d4a3a7217ee2f0ae872b16c2724 100644
--- a/webrtc/modules/audio_coding/codecs/isac/unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/unittest.cc
@@ -111,7 +111,7 @@ void TestGetSetBandwidthInfo(const int16_t* speech_data,
typename T::instance_type* encdec;
ASSERT_EQ(0, T::Create(&encdec));
ASSERT_EQ(0, T::EncoderInit(encdec, adaptive ? 0 : 1));
- ASSERT_EQ(0, T::DecoderInit(encdec));
+ T::DecoderInit(encdec);
ASSERT_EQ(0, T::SetEncSampRate(encdec, sample_rate_hz));
if (adaptive)
ASSERT_EQ(0, T::ControlBwe(encdec, bit_rate, frame_size_ms, false));
@@ -129,7 +129,7 @@ void TestGetSetBandwidthInfo(const int16_t* speech_data,
ASSERT_EQ(0, T::Control(enc, bit_rate, frame_size_ms));
typename T::instance_type* dec;
ASSERT_EQ(0, T::Create(&dec));
- ASSERT_EQ(0, T::DecoderInit(dec));
+ T::DecoderInit(dec);
T::SetInitialBweBottleneck(dec, bit_rate);
T::SetEncSampRateInDecoder(dec, sample_rate_hz);
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