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Add test NetEqDecodingTest.CngFirst

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This CL adds a test to verify that it is ok to start the stream with
a comfort noise packet.

BUG=4021
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7769 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2014-12-01 11:42:42 +00:00
parent 83317146ba
commit c93437ef96
1 changed files with 47 additions and 1 deletions
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48
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
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@ -1385,7 +1385,7 @@ void NetEqDecodingTest::DuplicateCng() {
const int algorithmic_delay_samples = std::max(
algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
// Insert three speech packet. Three are needed to get the frame length
// Insert three speech packets. Three are needed to get the frame length
// correct.
int out_len;
int num_channels;
@ -1462,4 +1462,50 @@ uint32_t NetEqDecodingTest::PlayoutTimestamp() {
}
TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { DuplicateCng(); }
TEST_F(NetEqDecodingTest, CngFirst) {
uint16_t seq_no = 0;
uint32_t timestamp = 0;
const int kFrameSizeMs = 10;
const int kSampleRateKhz = 16;
const int kSamples = kFrameSizeMs * kSampleRateKhz;
const int kPayloadBytes = kSamples * 2;
const int kCngPeriodMs = 100;
const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
size_t payload_len;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
ASSERT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
++seq_no;
timestamp += kCngPeriodSamples;
// Pull audio once and make sure CNG is played.
int out_len;
int num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
EXPECT_EQ(kOutputCNG, type);
// Insert some speech packets.
for (int i = 0; i < 3; ++i) {
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
++seq_no;
timestamp += kSamples;
// Pull audio once.
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
}
// Verify speech output.
EXPECT_EQ(kOutputNormal, type);
}
} // namespace webrtc