generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

NetEq changes.

Browse Source 
BUG=
R=henrik.lundin@webrtc.org, minyue@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5889 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org
2014-04-11 18:47:55 +00:00
parent ffd242432d
commit 8d1cdaa84e
19 changed files with 345 additions and 275 deletions
Download Patch File Download Diff File Expand all files Collapse all files

78
webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
View File

@@ -18,6 +18,7 @@
#include <stdlib.h>
#include <string.h> // memset
#include <algorithm>
#include <set>
#include <string>
#include <vector>
@@ -232,6 +233,7 @@ class NetEqDecodingTest : public ::testing::Test {
unsigned int sim_clock_;
int16_t out_data_[kMaxBlockSize];
int output_sample_rate_;
int algorithmic_delay_ms_;
};
// Allocating the static const so that it can be passed by reference.
@@ -246,12 +248,16 @@ NetEqDecodingTest::NetEqDecodingTest()
: neteq_(NULL),
rtp_fp_(NULL),
sim_clock_(0),
output_sample_rate_(kInitSampleRateHz) {
output_sample_rate_(kInitSampleRateHz),
algorithmic_delay_ms_(0) {
memset(out_data_, 0, sizeof(out_data_));
}
void NetEqDecodingTest::SetUp() {
neteq_ = NetEq::Create(kInitSampleRateHz);
NetEqNetworkStatistics stat;
ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
algorithmic_delay_ms_ = stat.current_buffer_size_ms;
ASSERT_TRUE(neteq_);
LoadDecoders();
}
@@ -483,8 +489,8 @@ void NetEqDecodingTest::CheckBgnOff(int sampling_rate_hz,
ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
// To be able to test the fading of background noise we need at lease to pull
// 610 frames.
const int kFadingThreshold = 610;
// 611 frames.
const int kFadingThreshold = 611;
// Test several CNG-to-PLC packet for the expected behavior. The number 20 is
// arbitrary, but sufficiently large to test enough number of frames.
@@ -1110,12 +1116,16 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketInsert)) {
// First insert several noise like packets, then sync-packets. Decoding all
// packets should not produce error, statistics should not show any packet loss
// and sync-packets should decode to zero.
// TODO(turajs) we will have a better test if we have a referece NetEq, and
// when Sync packets are inserted in "test" NetEq we insert all-zero payload
// in reference NetEq and compare the output of those two.
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
uint8_t payload[kPayloadBytes];
int16_t decoded[kBlockSize16kHz];
int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
for (int n = 0; n < kPayloadBytes; ++n) {
payload[n] = (rand() & 0xF0) + 1; // Non-zero random sequence.
}
@@ -1125,7 +1135,6 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
int num_channels;
int samples_per_channel;
uint32_t receive_timestamp = 0;
int delay_samples = 0;
for (int n = 0; n < 100; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
receive_timestamp));
@@ -1135,16 +1144,15 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
ASSERT_EQ(1, num_channels);
// Even if there is RTP packet in NetEq's buffer, the first frame pulled
// from NetEq starts with few zero samples. Here we measure this delay.
if (n == 0) {
while (decoded[delay_samples] == 0) delay_samples++;
}
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
}
const int kNumSyncPackets = 10;
// Make sure sufficient number of sync packets are inserted that we can
// conduct a test.
ASSERT_GT(kNumSyncPackets, algorithmic_frame_delay);
// Insert sync-packets, the decoded sequence should be all-zero.
for (int n = 0; n < kNumSyncPackets; ++n) {
ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
@@ -1153,30 +1161,37 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
&output_type));
ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
ASSERT_EQ(1, num_channels);
EXPECT_TRUE(IsAllZero(&decoded[delay_samples],
samples_per_channel * num_channels - delay_samples));
delay_samples = 0; // Delay only matters in the first frame.
if (n > algorithmic_frame_delay) {
EXPECT_TRUE(IsAllZero(decoded, samples_per_channel * num_channels));
}
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
}
// We insert a regular packet, if sync packet are not correctly buffered then
// We insert regular packets, if sync packet are not correctly buffered then
// network statistics would show some packet loss.
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
&samples_per_channel, &num_channels,
&output_type));
// Make sure the last inserted packet is decoded and there are non-zero
// samples.
EXPECT_FALSE(IsAllZero(decoded, samples_per_channel * num_channels));
for (int n = 0; n <= algorithmic_frame_delay + 10; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
&samples_per_channel, &num_channels,
&output_type));
if (n >= algorithmic_frame_delay + 1) {
// Expect that this frame contain samples from regular RTP.
EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels));
}
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
}
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
// Expecting a "clean" network.
EXPECT_EQ(0, network_stats.packet_loss_rate);
EXPECT_EQ(0, network_stats.expand_rate);
EXPECT_EQ(0, network_stats.accelerate_rate);
EXPECT_EQ(0, network_stats.preemptive_rate);
EXPECT_LE(network_stats.preemptive_rate, 150);
}
// Test if the size of the packet buffer reported correctly when containing
@@ -1199,7 +1214,8 @@ TEST_F(NetEqDecodingTest,
int num_channels;
int samples_per_channel;
uint32_t receive_timestamp = 0;
for (int n = 0; n < 1; ++n) {
int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
for (int n = 0; n < algorithmic_frame_delay; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
@@ -1225,7 +1241,8 @@ TEST_F(NetEqDecodingTest,
}
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
EXPECT_EQ(kNumSyncPackets * 10, network_stats.current_buffer_size_ms);
EXPECT_EQ(kNumSyncPackets * 10 + algorithmic_delay_ms_,
network_stats.current_buffer_size_ms);
// Rewind |rtp_info| to that of the first sync packet.
memcpy(&rtp_info, &first_sync_packet_rtp_info, sizeof(rtp_info));
@@ -1298,7 +1315,8 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
if (packets_inserted > 4) {
// Expect preferred and actual buffer size to be no more than 2 frames.
EXPECT_LE(network_stats.preferred_buffer_size_ms, kFrameSizeMs * 2);
EXPECT_LE(network_stats.current_buffer_size_ms, kFrameSizeMs * 2);
EXPECT_LE(network_stats.current_buffer_size_ms, kFrameSizeMs * 2 +
algorithmic_delay_ms_);
}
last_seq_no = seq_no;
last_timestamp = timestamp;
@@ -1362,6 +1380,8 @@ void NetEqDecodingTest::DuplicateCng() {
const int kSamples = kFrameSizeMs * kSampleRateKhz;
const int kPayloadBytes = kSamples * 2;
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
// correct.
int out_len;
@@ -1398,7 +1418,7 @@ void NetEqDecodingTest::DuplicateCng() {
kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
EXPECT_EQ(kOutputCNG, type);
EXPECT_EQ(timestamp - 10, neteq_->PlayoutTimestamp());
EXPECT_EQ(timestamp - algorithmic_delay_samples, neteq_->PlayoutTimestamp());
// Insert the same CNG packet again. Note that at this point it is old, since
// we have already decoded the first copy of it.
@@ -1412,7 +1432,8 @@ void NetEqDecodingTest::DuplicateCng() {
kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
EXPECT_EQ(kOutputCNG, type);
EXPECT_EQ(timestamp - 10, neteq_->PlayoutTimestamp());
EXPECT_EQ(timestamp - algorithmic_delay_samples,
neteq_->PlayoutTimestamp());
}
// Insert speech again.
@@ -1427,7 +1448,8 @@ void NetEqDecodingTest::DuplicateCng() {
kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
EXPECT_EQ(kOutputNormal, type);
EXPECT_EQ(timestamp + kSamples - 10, neteq_->PlayoutTimestamp());
EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
neteq_->PlayoutTimestamp());
}
TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { DuplicateCng(); }