Use a deterministic input in NetEqBgnTest
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/webrtc@6948 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/audio_coding/codecs/pcm16b/include/pcm16b.h b/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
index 76ba2ff..505142e 100644
--- a/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
+++ b/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
@@ -35,9 +35,9 @@
* 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(...)
diff --git a/modules/audio_coding/codecs/pcm16b/pcm16b.c b/modules/audio_coding/codecs/pcm16b/pcm16b.c
index 34aadc3..7661dc1 100644
--- a/modules/audio_coding/codecs/pcm16b/pcm16b.c
+++ b/modules/audio_coding/codecs/pcm16b/pcm16b.c
@@ -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);
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 1fd5345..d60077d 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -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 @@
}
}
-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 @@
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 @@
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 @@
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 @@
}
};
-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;
