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Use a deterministic input in NetEqBgnTest

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This test has been failing every now and then. This is likely due to the
random input that was used. With this change, the input is now read from
an audio file, making it identical on each run.

The encoding is moved to inside the main test loop, so that new data is
added with each packet. (Before this change, the same payload was added
over and over again; only the RTP header was updated.)

BUG=3715
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6948 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2014-08-21 08:27:44 +00:00
parent 6b2659c660
commit 9b8102cf0e
3 changed files with 73 additions and 38 deletions
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4
webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
View File

@ -35,9 +35,9 @@ extern "C" {
* Returned value : Size in bytes of speechOut16b
*/
int16_t WebRtcPcm16b_EncodeW16(int16_t *speechIn16b,
int16_t WebRtcPcm16b_EncodeW16(const int16_t* speechIn16b,
int16_t len,
int16_t *speechOut16b);
int16_t* speechOut16b);
/****************************************************************************
* WebRtcPcm16b_Encode(...)

4
webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
View File

@ -25,9 +25,9 @@
/* Encoder with int16_t Output */
int16_t WebRtcPcm16b_EncodeW16(int16_t *speechIn16b,
int16_t WebRtcPcm16b_EncodeW16(const int16_t* speechIn16b,
int16_t len,
int16_t *speechOut16b)
int16_t* speechOut16b)
{
#ifdef WEBRTC_ARCH_BIG_ENDIAN
WEBRTC_SPL_MEMCPY_W16(speechOut16b, speechIn16b, len);

103
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -26,6 +26,7 @@
#include "gflags/gflags.h"
#include "gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
@ -870,15 +871,12 @@ TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
}
}
class NetEqBgnTest
: public NetEqDecodingTest,
public ::testing::WithParamInterface<NetEq::BackgroundNoiseMode> {
class NetEqBgnTest : public NetEqDecodingTest {
protected:
NetEqBgnTest() : NetEqDecodingTest() {
config_.background_noise_mode = GetParam();
}
virtual void TestCondition(double sum_squared_noise,
bool should_be_faded) = 0;
void CheckBgnOff(int sampling_rate_hz) {
void CheckBgn(int sampling_rate_hz) {
int expected_samples_per_channel = 0;
uint8_t payload_type = 0xFF; // Invalid.
if (sampling_rate_hz == 8000) {
@ -896,21 +894,17 @@ class NetEqBgnTest
NetEqOutputType type;
int16_t output[kBlockSize32kHz]; // Maximum size is chosen.
int16_t input[kBlockSize32kHz]; // Maximum size is chosen.
test::AudioLoop input;
// We are using the same 32 kHz input file for all tests, regardless of
// |sampling_rate_hz|. The output may sound weird, but the test is still
// valid.
ASSERT_TRUE(input.Init(
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
10 * sampling_rate_hz, // Max 10 seconds loop length.
expected_samples_per_channel));
// Payload of 10 ms of PCM16 32 kHz.
uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
// Random payload.
for (int n = 0; n < expected_samples_per_channel; ++n) {
input[n] = (rand() & ((1 << 10) - 1)) - ((1 << 5) - 1);
}
int enc_len_bytes =
WebRtcPcm16b_EncodeW16(input,
expected_samples_per_channel,
reinterpret_cast<int16_t*>(payload));
ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.header.payloadType = payload_type;
@ -920,6 +914,12 @@ class NetEqBgnTest
uint32_t receive_timestamp = 0;
for (int n = 0; n < 10; ++n) { // Insert few packets and get audio.
int enc_len_bytes =
WebRtcPcm16b_EncodeW16(input.GetNextBlock(),
expected_samples_per_channel,
reinterpret_cast<int16_t*>(payload));
ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
number_channels = 0;
samples_per_channel = 0;
ASSERT_EQ(0,
@ -981,12 +981,7 @@ class NetEqBgnTest
double sum_squared = 0;
for (int k = 0; k < number_channels * samples_per_channel; ++k)
sum_squared += output[k] * output[k];
if (config_.background_noise_mode == NetEq::kBgnOn) {
EXPECT_NE(0, sum_squared);
} else if (config_.background_noise_mode == NetEq::kBgnOff ||
n > kFadingThreshold) {
EXPECT_EQ(0, sum_squared);
}
TestCondition(sum_squared, n > kFadingThreshold);
} else {
EXPECT_EQ(kOutputPLC, type);
}
@ -995,17 +990,57 @@ class NetEqBgnTest
}
};
TEST_P(NetEqBgnTest, BackgroundNoise) {
CheckBgnOff(8000);
CheckBgnOff(16000);
CheckBgnOff(32000);
class NetEqBgnTestOn : public NetEqBgnTest {
protected:
NetEqBgnTestOn() : NetEqBgnTest() {
config_.background_noise_mode = NetEq::kBgnOn;
}
void TestCondition(double sum_squared_noise, bool /*should_be_faded*/) {
EXPECT_NE(0, sum_squared_noise);
}
};
class NetEqBgnTestOff : public NetEqBgnTest {
protected:
NetEqBgnTestOff() : NetEqBgnTest() {
config_.background_noise_mode = NetEq::kBgnOff;
}
void TestCondition(double sum_squared_noise, bool /*should_be_faded*/) {
EXPECT_EQ(0, sum_squared_noise);
}
};
class NetEqBgnTestFade : public NetEqBgnTest {
protected:
NetEqBgnTestFade() : NetEqBgnTest() {
config_.background_noise_mode = NetEq::kBgnFade;
}
void TestCondition(double sum_squared_noise, bool should_be_faded) {
if (should_be_faded)
EXPECT_EQ(0, sum_squared_noise);
}
};
TEST_F(NetEqBgnTestOn, RunTest) {
CheckBgn(8000);
CheckBgn(16000);
CheckBgn(32000);
}
INSTANTIATE_TEST_CASE_P(BgnModes,
NetEqBgnTest,
::testing::Values(NetEq::kBgnOn,
NetEq::kBgnOff,
NetEq::kBgnFade));
TEST_F(NetEqBgnTestOff, RunTest) {
CheckBgn(8000);
CheckBgn(16000);
CheckBgn(32000);
}
TEST_F(NetEqBgnTestFade, RunTest) {
CheckBgn(8000);
CheckBgn(16000);
CheckBgn(32000);
}
TEST_F(NetEqDecodingTest, SyncPacketInsert) {
WebRtcRTPHeader rtp_info;