Index: webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc |
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc |
index a2ef5b0bbb47ac6896c5378853855395ec606bd0..470f690ed9c131825c07af5d2c9fa5f45bb04a0a 100644 |
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc |
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc |
@@ -205,7 +205,9 @@ class AudioCodingModuleTestOldApi : public ::testing::Test { |
virtual void PullAudio() { |
AudioFrame audio_frame; |
- ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame)); |
+ bool muted; |
+ ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame, &muted)); |
+ ASSERT_FALSE(muted); |
} |
virtual void InsertAudio() { |
@@ -296,7 +298,9 @@ TEST_F(AudioCodingModuleTestOldApi, MAYBE_NetEqCalls) { |
TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) { |
AudioFrame audio_frame; |
const int kSampleRateHz = 32000; |
- EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame)); |
+ bool muted; |
+ EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted)); |
+ ASSERT_FALSE(muted); |
EXPECT_EQ(id_, audio_frame.id_); |
EXPECT_EQ(0u, audio_frame.timestamp_); |
EXPECT_GT(audio_frame.num_channels_, 0u); |
@@ -307,7 +311,8 @@ TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) { |
TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) { |
AudioFrame audio_frame; |
- EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame)); |
+ bool muted; |
+ EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame, &muted)); |
} |
// Checks that the transport callback is invoked once for each speech packet. |
@@ -806,8 +811,13 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi { |
// Pull audio. |
for (int i = 0; i < rtc::CheckedDivExact(kPacketSizeMs, 10); ++i) { |
AudioFrame audio_frame; |
+ bool muted; |
EXPECT_EQ(0, acm_->PlayoutData10Ms(-1 /* default output frequency */, |
- &audio_frame)); |
+ &audio_frame, &muted)); |
+ if (muted) { |
+ ADD_FAILURE(); |
+ return false; |
+ } |
fake_clock_->AdvanceTimeMilliseconds(10); |
} |
rtp_utility_->Forward(&rtp_header_); |