Add min and max allowed bitrate in Opus bitrate tests
Instead of checking for an exact bitrate check that the bitrate is between
the min and max values.
Also relax a threshold in a bandwith adaptation test.
Bug: webrtc:9280
Change-Id: I465d785a53759f73242198ee1ccd7da1a26c48b7
Reviewed-on: https://webrtc-review.googlesource.com/78041
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23330}diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 1664b26..239e7eb 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -1591,13 +1591,14 @@
return send_test_->RegisterExternalCodec(external_speech_encoder);
}
- void RunInner(int expected_total_bits) {
+ void RunInner(int min_expected_total_bits, int max_expected_total_bits) {
int nr_bytes = 0;
while (std::unique_ptr<test::Packet> next_packet =
send_test_->NextPacket()) {
nr_bytes += rtc::checked_cast<int>(next_packet->payload_length_bytes());
}
- EXPECT_EQ(expected_total_bits, nr_bytes * 8);
+ EXPECT_LE(min_expected_total_bits, nr_bytes * 8);
+ EXPECT_GE(max_expected_total_bits, nr_bytes * 8);
}
void SetUpTest(const char* codec_name,
@@ -1620,10 +1621,12 @@
protected:
// Runs the test. SetUpSender() must have been called and a codec must be set
// up before calling this method.
- void Run(int target_bitrate_bps, int expected_total_bits) {
+ void Run(int target_bitrate_bps,
+ int min_expected_total_bits,
+ int max_expected_total_bits) {
ASSERT_TRUE(send_test_->acm());
send_test_->acm()->SetBitRate(target_bitrate_bps);
- RunInner(expected_total_bits);
+ RunInner(min_expected_total_bits, max_expected_total_bits);
}
};
@@ -1631,16 +1634,14 @@
protected:
// Runs the test. SetUpSender() must have been called and a codec must be set
// up before calling this method.
- void Run(int expected_total_bits) { RunInner(expected_total_bits); }
+ void Run(int min_expected_total_bits, int max_expected_total_bits) {
+ RunInner(min_expected_total_bits, max_expected_total_bits);
+ }
};
TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
-#if defined(WEBRTC_ANDROID)
- Run(10000, 8640);
-#else
- Run(10000, 8696);
-#endif // WEBRTC_ANDROID
+ Run(10000, 8000, 12000);
}
TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) {
@@ -1649,20 +1650,12 @@
const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107);
ASSERT_TRUE(SetUpSender());
ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
-#if defined(WEBRTC_ANDROID)
- RunInner(8640);
-#else
- RunInner(8696);
-#endif // WEBRTC_ANDROID
+ RunInner(8000, 12000);
}
TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
-#if defined(WEBRTC_ANDROID)
- Run(50000, 45792);
-#else
- Run(50000, 45600);
-#endif // WEBRTC_ANDROID
+ Run(50000, 40000, 60000);
}
TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
@@ -1671,11 +1664,7 @@
const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107);
ASSERT_TRUE(SetUpSender());
ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
-#if defined(WEBRTC_ANDROID)
- RunInner(45792);
-#else
- RunInner(45600);
-#endif // WEBRTC_ANDROID
+ RunInner(40000, 60000);
}
// The result on the Android platforms is inconsistent for this test case.
@@ -1691,7 +1680,7 @@
#endif
TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
- Run(100000, 100832);
+ Run(100000, 80000, 120000);
}
TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_100kbps) {
@@ -1700,18 +1689,18 @@
const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107);
ASSERT_TRUE(SetUpSender());
ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
- RunInner(100832);
+ RunInner(80000, 120000);
}
// These next 2 tests ensure that the SetBitRate function has no effect on PCM
TEST_F(AcmSetBitRateOldApi, Pcm16_8khz_10ms_8kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
- Run(8000, 128000);
+ Run(8000, 128000, 128000);
}
TEST_F(AcmSetBitRateOldApi, Pcm16_8khz_10ms_32kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
- Run(32000, 128000);
+ Run(32000, 128000, 128000);
}
// This test is for verifying the SetBitRate function. The bitrate is changed
@@ -1758,8 +1747,11 @@
next_packet->payload_length_bytes());
packet_counter++;
}
- EXPECT_EQ(expected_before_switch_bits, nr_bytes_before * 8);
- EXPECT_EQ(expected_after_switch_bits, nr_bytes_after * 8);
+ // Check that bitrate is 80-120 percent of expected value.
+ EXPECT_GE(expected_before_switch_bits, nr_bytes_before * 8 * 8 / 10);
+ EXPECT_LE(expected_before_switch_bits, nr_bytes_before * 8 * 12 / 10);
+ EXPECT_GE(expected_after_switch_bits, nr_bytes_after * 8 * 8 / 10);
+ EXPECT_LE(expected_after_switch_bits, nr_bytes_after * 8 * 12 / 10);
}
uint32_t sampling_freq_hz_;
@@ -1768,33 +1760,17 @@
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
-#if defined(WEBRTC_ANDROID)
- Run(10000, 29512, 4800);
-#else
Run(10000, 32200, 5208);
-#endif // WEBRTC_ANDROID
}
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
-#if defined(WEBRTC_ANDROID)
- Run(50000, 29512, 23304);
-#else
Run(50000, 32200, 23928);
-#endif // WEBRTC_ANDROID
}
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
-#if defined(WEBRTC_ANDROID)
- #if defined(WEBRTC_ARCH_ARM64)
- Run(100000, 29512, 50440);
- #else
- Run(100000, 29512, 50496);
- #endif // WEBRTC_ARCH_ARM64
-#else
Run(100000, 32200, 50448);
-#endif // WEBRTC_ANDROID
}
// These next 2 tests ensure that the SetBitRate function has no effect on PCM
diff --git a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
index 4394949..3c4cb02 100644
--- a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
@@ -105,7 +105,7 @@
test::ScopedFieldTrials override_field_trials(
"WebRTC-AdjustOpusBandwidth/Enabled/");
- constexpr float kMaxNarrowbandRatio = 0.003f;
+ constexpr float kMaxNarrowbandRatio = 0.0035f;
constexpr float kMinWidebandRatio = 0.03f;
// Create encoder.