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NetEq: Add functionality to assist with delay analysis and tooling

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This CL adds a few methods to the NetEq API that will be used for
delay analysis and plotting.

BUG=webrtc:7467

Review-Url: https://codereview.webrtc.org/2839163002
Cr-Commit-Position: refs/heads/master@{#17889}
This commit is contained in:
henrik.lundin
2017-04-26 07:47:32 -07:00
committed by Commit bot
parent 2e79958c42
commit 114c1b3afa
6 changed files with 126 additions and 18 deletions
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81
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
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@ -270,8 +270,6 @@ class NetEqDecodingTest : public ::testing::Test {
void DuplicateCng();
rtc::Optional<uint32_t> PlayoutTimestamp();
NetEq* neteq_;
NetEq::Config config_;
std::unique_ptr<test::RtpFileSource> rtp_source_;
@ -644,7 +642,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
}
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
rtc::Optional<uint32_t> playout_timestamp = PlayoutTimestamp();
rtc::Optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
int32_t delay_before = timestamp - *playout_timestamp;
@ -736,7 +734,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
// Check that the speech starts again within reasonable time.
double time_until_speech_returns_ms = t_ms - speech_restart_time_ms;
EXPECT_LT(time_until_speech_returns_ms, max_time_to_speech_ms);
playout_timestamp = PlayoutTimestamp();
playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
int32_t delay_after = timestamp - *playout_timestamp;
// Compare delay before and after, and make sure it differs less than 20 ms.
@ -1128,7 +1126,7 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
ASSERT_EQ(1u, output.num_channels_);
// Expect delay (in samples) to be less than 2 packets.
rtc::Optional<uint32_t> playout_timestamp = PlayoutTimestamp();
rtc::Optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
EXPECT_LE(timestamp - *playout_timestamp,
static_cast<uint32_t>(kSamples * 2));
@ -1207,7 +1205,8 @@ void NetEqDecodingTest::DuplicateCng() {
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
EXPECT_FALSE(PlayoutTimestamp()); // Returns empty value during CNG.
EXPECT_FALSE(
neteq_->GetPlayoutTimestamp()); // Returns empty value during CNG.
EXPECT_EQ(timestamp - algorithmic_delay_samples,
out_frame_.timestamp_ + out_frame_.samples_per_channel_);
@ -1223,7 +1222,8 @@ void NetEqDecodingTest::DuplicateCng() {
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
EXPECT_FALSE(PlayoutTimestamp()); // Returns empty value during CNG.
EXPECT_FALSE(
neteq_->GetPlayoutTimestamp()); // Returns empty value during CNG.
EXPECT_EQ(timestamp - algorithmic_delay_samples,
out_frame_.timestamp_ + out_frame_.samples_per_channel_);
}
@ -1238,16 +1238,12 @@ void NetEqDecodingTest::DuplicateCng() {
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
rtc::Optional<uint32_t> playout_timestamp = PlayoutTimestamp();
rtc::Optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
*playout_timestamp);
}
rtc::Optional<uint32_t> NetEqDecodingTest::PlayoutTimestamp() {
return neteq_->GetPlayoutTimestamp();
}
TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { DuplicateCng(); }
TEST_F(NetEqDecodingTest, CngFirst) {
@ -1590,4 +1586,65 @@ TEST_F(NetEqDecodingTestTwoInstances, CompareMutedStateOnOff) {
EXPECT_FALSE(muted);
}
TEST_F(NetEqDecodingTest, LastDecodedTimestampsEmpty) {
EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
// Pull out data once.
AudioFrame output;
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
}
TEST_F(NetEqDecodingTest, LastDecodedTimestampsOneDecoded) {
// Insert one packet with PCM16b WB data (this is what PopulateRtpInfo does by
// default). Make the length 10 ms.
constexpr size_t kPayloadSamples = 16 * 10;
constexpr size_t kPayloadBytes = 2 * kPayloadSamples;
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
constexpr uint32_t kRtpTimestamp = 0x1234;
PopulateRtpInfo(0, kRtpTimestamp, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
// Pull out data once.
AudioFrame output;
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp}),
neteq_->LastDecodedTimestamps());
// Nothing decoded on the second call.
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
}
TEST_F(NetEqDecodingTest, LastDecodedTimestampsTwoDecoded) {
// Insert two packets with PCM16b WB data (this is what PopulateRtpInfo does
// by default). Make the length 5 ms so that NetEq must decode them both in
// the same GetAudio call.
constexpr size_t kPayloadSamples = 16 * 5;
constexpr size_t kPayloadBytes = 2 * kPayloadSamples;
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
constexpr uint32_t kRtpTimestamp1 = 0x1234;
PopulateRtpInfo(0, kRtpTimestamp1, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp1 + kPayloadSamples;
PopulateRtpInfo(1, kRtpTimestamp2, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
// Pull out data once.
AudioFrame output;
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp1, kRtpTimestamp2}),
neteq_->LastDecodedTimestamps());
}
} // namespace webrtc