Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc

Index: webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
diff --git a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
index ec3f4fcddc22dc7a5466f68d71e1d8f9629b0dfb..e4a34a729464ec0f6b145d47f8e7e35dc2bf13ef 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
@@ -159,7 +159,7 @@ void AcmReceiveTestOldApi::Run() {
     while (clock_.TimeInMilliseconds() < packet->time_ms()) {
       AudioFrame output_frame;
       EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
-      EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
+      ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
       const size_t samples_per_block =
           static_cast<size_t>(output_freq_hz_ * 10 / 1000);
       EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);