Remove legacy delay manger field trial and update default config.
Bug: webrtc:10333
Change-Id: I20e55d8d111d93657d1afe556fe3a325337c074c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232820
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35321}
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 87379c8..df8f5a8 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -920,61 +920,61 @@
defined(WEBRTC_CODEC_ILBC)
TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
std::string others_checksum_reference =
- GetCPUInfo(kAVX2) != 0 ? "e0c966d7b8c36ff60167988fa35d33e0"
- : "7d8f6b84abd1e57ec010a53bc2130652";
+ GetCPUInfo(kAVX2) != 0 ? "d8671dd38dab43fc9ca64a45c048c218"
+ : "4710c99559aec2f9f02a983ba2146f2d";
std::string win64_checksum_reference =
GetCPUInfo(kAVX2) != 0 ? "405a50f0bcb8827e20aa944299fc59f6"
: "0ed5830930f5527a01bbec0ba11f8541";
Run(8000, PlatformChecksum(
others_checksum_reference, win64_checksum_reference,
- /*android_arm32=*/"b892ed69c38b21b16c132ec2ce03aa7b",
+ /*android_arm32=*/"4a8ffd7fd235c8bea74d0e18c022fac3",
/*android_arm64=*/"4598140b5e4f7ee66c5adad609e65a3e",
- /*android_arm64_clang=*/"5fec8d770778ef7969ec98c56d9eb10f",
+ /*android_arm64_clang=*/"ad2ae6c6e48b714d728a7af0d3c8dc51",
/*mac_arm64=*/"636efe6d0a148f22c5383f356da3deac"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
std::string others_checksum_reference =
- GetCPUInfo(kAVX2) != 0 ? "a63c578e1195c8420f453962c6d8519c"
- : "6bac83762c1306b932cd25a560155681";
+ GetCPUInfo(kAVX2) != 0 ? "abcb31509af46545edb4f6700728a4de"
+ : "70b3217df49834b7093c631531068bd0";
std::string win64_checksum_reference =
GetCPUInfo(kAVX2) != 0 ? "58fd62a5c49ee513f9fa6fe7dbf62c97"
: "0509cf0672f543efb4b050e8cffefb1d";
Run(16000, PlatformChecksum(
others_checksum_reference, win64_checksum_reference,
- /*android_arm32=*/"3cea9abbeabbdea9a79719941b241af5",
+ /*android_arm32=*/"00d703da221363804d6fccc309a3f684",
/*android_arm64=*/"f2aad418af974a3b1694d5ae5cc2c3c7",
- /*android_arm64_clang=*/"9d4b92c31c00e321a4cff29ad002d6a2",
+ /*android_arm64_clang=*/"2b8525c77a6e10800bb209a83160282a",
/*mac_arm64=*/"1e2d1b482fdc924f79a838503ee7ead5"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
std::string others_checksum_reference =
- GetCPUInfo(kAVX2) != 0 ? "8775ce387f44dc5ff4a26da295d5ee7c"
- : "e319222ca47733709f90fdf33c8574db";
+ GetCPUInfo(kAVX2) != 0 ? "8489b7743d6cd1903807ac81e5ee493d"
+ : "2679e4e596e33259228c62df545eb635";
std::string win64_checksum_reference =
GetCPUInfo(kAVX2) != 0 ? "04ce6a1dac5ffdd8438d804623d0132f"
: "39a4a7a1c455b35baeffb9fd193d7858";
Run(32000, PlatformChecksum(
others_checksum_reference, win64_checksum_reference,
- /*android_arm32=*/"4df55b3b62bcbf4328786d474ae87f61",
+ /*android_arm32=*/"809446f684b8095a93495ad63ec19891",
/*android_arm64=*/"100869c8dcde51346c2073e52a272d98",
- /*android_arm64_clang=*/"ff58d3153d2780a3df6bc2068844cb2d",
+ /*android_arm64_clang=*/"dfe6fba596ed68d5a32d9f9eba5a39cb",
/*mac_arm64=*/"51788e9784a10ae14a030f075a039205"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
std::string others_checksum_reference =
- GetCPUInfo(kAVX2) != 0 ? "7a55700b7ca9aa60237db58b33e55606"
- : "57d1d316c88279f4f3da3511665069a9";
+ GetCPUInfo(kAVX2) != 0 ? "454996a7adb3f62b259a53a09ff624cf"
+ : "f0148c5ef84e74e019ac7057af839102";
std::string win64_checksum_reference =
GetCPUInfo(kAVX2) != 0 ? "f59833d9b0924f4b0704707dd3589f80"
: "74cbe7345e2b6b45c1e455a5d1e921ca";
Run(48000, PlatformChecksum(
others_checksum_reference, win64_checksum_reference,
- /*android_arm32=*/"f52bc7bf0f499c9da25932fdf176c4ec",
+ /*android_arm32=*/"f5c1290ce96d675aaf52be0b54362bee",
/*android_arm64=*/"bd44bf97e7899186532f91235cef444d",
- /*android_arm64_clang=*/"364d403dae55d73cd69e6dbd6b723a4d",
+ /*android_arm64_clang=*/"7c2e28b943baf8c8af556be203bea256",
/*mac_arm64=*/"71bc5c15a151400517c2119d1602ee9f"));
}
@@ -1054,17 +1054,17 @@
rtc::scoped_refptr<rtc::RefCountedObject<ADFactory>> factory(
new rtc::RefCountedObject<ADFactory>);
std::string others_checksum_reference =
- GetCPUInfo(kAVX2) != 0 ? "7a55700b7ca9aa60237db58b33e55606"
- : "57d1d316c88279f4f3da3511665069a9";
+ GetCPUInfo(kAVX2) != 0 ? "454996a7adb3f62b259a53a09ff624cf"
+ : "f0148c5ef84e74e019ac7057af839102";
std::string win64_checksum_reference =
GetCPUInfo(kAVX2) != 0 ? "f59833d9b0924f4b0704707dd3589f80"
: "74cbe7345e2b6b45c1e455a5d1e921ca";
Run(48000,
PlatformChecksum(
others_checksum_reference, win64_checksum_reference,
- /*android_arm32=*/"f52bc7bf0f499c9da25932fdf176c4ec",
+ /*android_arm32=*/"f5c1290ce96d675aaf52be0b54362bee",
/*android_arm64=*/"bd44bf97e7899186532f91235cef444d",
- /*android_arm64_clang=*/"364d403dae55d73cd69e6dbd6b723a4d",
+ /*android_arm64_clang=*/"7c2e28b943baf8c8af556be203bea256",
/*mac_arm64=*/"71bc5c15a151400517c2119d1602ee9f"),
factory, [](AudioCodingModule* acm) {
acm->SetReceiveCodecs({{0, {"MockPCMu", 8000, 1}},
@@ -1284,12 +1284,12 @@
TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
- /*others=*/"2c9cb15d4ed55b5a0cadd04883bc73b0",
+ /*others=*/"a3077ac01b0137e8bbc237fb1f9816a5",
/*win64=*/"9336a9b993cbd8a751f0e8958e66c89c",
- /*android_arm32=*/"5c2eb46199994506236f68b2c8e51b0d",
+ /*android_arm32=*/"ab39f101ca76efdf6a5b2250550f10c4",
/*android_arm64=*/"343f1f42be0607c61e6516aece424609",
- /*android_arm64_clang=*/"2c9cb15d4ed55b5a0cadd04883bc73b0",
- /*mac_arm64=*/"2c9cb15d4ed55b5a0cadd04883bc73b0"),
+ /*android_arm64_clang=*/"a3077ac01b0137e8bbc237fb1f9816a5",
+ /*mac_arm64=*/"a3077ac01b0137e8bbc237fb1f9816a5"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
/*others=*/"3c79f16f34218271f3dca4e2b1dfe1bb",
/*win64=*/"d42cb5195463da26c8129bbfe73a22e6",
@@ -1303,12 +1303,12 @@
TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
- /*others=*/"1ad29139a04782a33daad8c2b9b35875",
+ /*others=*/"76da9b7514f986fc2bb32b1c3170e8d4",
/*win64=*/"14d63c5f08127d280e722e3191b73bdd",
- /*android_arm32=*/"9a81e467eb1485f84aca796f8ea65011",
+ /*android_arm32=*/"0bd883118ff9c26b9471df7a0c664197",
/*android_arm64=*/"ef75e900e6f375e3061163c53fd09a63",
- /*android_arm64_clang=*/"1ad29139a04782a33daad8c2b9b35875",
- /*mac_arm64=*/"1ad29139a04782a33daad8c2b9b35875"),
+ /*android_arm64_clang=*/"76da9b7514f986fc2bb32b1c3170e8d4",
+ /*mac_arm64=*/"76da9b7514f986fc2bb32b1c3170e8d4"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
/*others=*/"9e0a0ab743ad987b55b8e14802769c56",
/*win64=*/"ebe04a819d3a9d83a83a17f271e1139a",
@@ -1330,10 +1330,10 @@
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacSwb30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
- /*others=*/"5683b58da0fbf2063c7adc2e6bfb3fb8",
+ /*others=*/"f4cf577f28a0dcbac33358b757518e0c",
/*win64=*/"2b3c387d06f00b7b7aad4c9be56fb83d", "android_arm32_audio",
"android_arm64_audio", "android_arm64_clang_audio",
- /*mac_arm64=*/"5683b58da0fbf2063c7adc2e6bfb3fb8"),
+ /*mac_arm64=*/"f4cf577f28a0dcbac33358b757518e0c"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
/*others=*/"ce86106a93419aefb063097108ec94ab",
/*win64=*/"bcc2041e7744c7ebd9f701866856849c", "android_arm32_payload",
@@ -1345,61 +1345,61 @@
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
- Run("15396f66b5b0ab6842e151c807395e4c", "c1edd36339ce0326cc4550041ad719a0",
+ Run("69118ed438ac76252d023e0463819471", "c1edd36339ce0326cc4550041ad719a0",
100, test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160));
- Run("54ae004529874c2b362c7f0ccd19cb99", "ad786526383178b08d80d6eee06e9bad",
+ Run("bc6ab94d12a464921763d7544fdbd07e", "ad786526383178b08d80d6eee06e9bad",
100, test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_32000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 1, 109, 320, 320));
- Run("d6a4a68b8c838dcc1e7ae7136467cdf0", "5ef82ea885e922263606c6fdbc49f651",
+ Run("c50244419c5c3a2f04cc69a022c266a2", "5ef82ea885e922263606c6fdbc49f651",
100, test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_8000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 2, 111, 80, 80));
- Run("6b011dab43e3a8a46ccff7e4412ed8a2", "62ce5adb0d4965d0a52ec98ae7f98974",
+ Run("4fccf4cc96f1e8e8de4b9fadf62ded9e", "62ce5adb0d4965d0a52ec98ae7f98974",
100, test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_16000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 2, 112, 160, 160));
- Run("17fc9854358bfe0419408290664bd78e", "41ca8edac4b8c71cd54fd9f25ec14870",
+ Run("e15e388d9d4af8c02a59fe1552fedee3", "41ca8edac4b8c71cd54fd9f25ec14870",
100, test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_32000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 2, 113, 320, 320));
- Run("9ac9a1f64d55da2fc9f3167181cc511d", "50e58502fb04421bf5b857dda4c96879",
+ Run("b240520c0d05003fde7a174ae5957286", "50e58502fb04421bf5b857dda4c96879",
100, test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcmu_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 1, 0, 160, 160));
- Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",
+ Run("c8d1fc677f33c2022ec5f83c7f302280", "8f9b8750bd80fe26b6cbf6659b89f0f9",
50, test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcma_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 1, 8, 160, 160));
- Run("39611f798969053925a49dc06d08de29", "6ad745e55aa48981bfc790d0eeef2dd1",
+ Run("47eb60e855eb12d1b0e6da9c975754a4", "6ad745e55aa48981bfc790d0eeef2dd1",
50, test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcmu_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 2, 110, 160, 160));
- Run("437bec032fdc5cbaa0d5175430af7b18", "60b6f25e8d1e74cb679cfe756dd9bca5",
+ Run("6ef2f57d4934714787fd0a834e3ea18e", "60b6f25e8d1e74cb679cfe756dd9bca5",
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 2, 118, 160, 160));
- Run("a5c6d83c5b7cedbeff734238220a4b0c", "92b282c83efd20e7eeef52ba40842cf7",
+ Run("a84d75e098d87ab6b260687eb4b612a2", "92b282c83efd20e7eeef52ba40842cf7",
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
@@ -1412,10 +1412,10 @@
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Ilbc_30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
- /*others=*/"7b6ec10910debd9af08011d3ed5249f7",
- /*win64=*/"7b6ec10910debd9af08011d3ed5249f7", "android_arm32_audio",
+ /*others=*/"b14dba0de36efa5ec88a32c0b320b70f",
+ /*win64=*/"b14dba0de36efa5ec88a32c0b320b70f", "android_arm32_audio",
"android_arm64_audio", "android_arm64_clang_audio",
- /*mac_arm64=*/"7b6ec10910debd9af08011d3ed5249f7"),
+ /*mac_arm64=*/"b14dba0de36efa5ec88a32c0b320b70f"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
/*others=*/"cfae2e9f6aba96e145f2bcdd5050ce78",
/*win64=*/"cfae2e9f6aba96e145f2bcdd5050ce78", "android_arm32_payload",
@@ -1433,10 +1433,10 @@
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
- /*others=*/"e99c89be49a46325d03c0d990c292d68",
- /*win64=*/"e99c89be49a46325d03c0d990c292d68", "android_arm32_audio",
+ /*others=*/"a87a91ec0124510a64967f5d768554ff",
+ /*win64=*/"a87a91ec0124510a64967f5d768554ff", "android_arm32_audio",
"android_arm64_audio", "android_arm64_clang_audio",
- /*mac_arm64=*/"e99c89be49a46325d03c0d990c292d68"),
+ /*mac_arm64=*/"a87a91ec0124510a64967f5d768554ff"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
/*others=*/"fc68a87e1380614e658087cb35d5ca10",
/*win64=*/"fc68a87e1380614e658087cb35d5ca10", "android_arm32_payload",
@@ -1453,10 +1453,10 @@
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
- /*others=*/"e280aed283e499d37091b481ca094807",
- /*win64=*/"e280aed283e499d37091b481ca094807", "android_arm32_audio",
+ /*others=*/"be0b8528ff9db3a2219f55ddd36faf7f",
+ /*win64=*/"be0b8528ff9db3a2219f55ddd36faf7f", "android_arm32_audio",
"android_arm64_audio", "android_arm64_clang_audio",
- /*mac_arm64=*/"e280aed283e499d37091b481ca094807"),
+ /*mac_arm64=*/"be0b8528ff9db3a2219f55ddd36faf7f"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
/*others=*/"66516152eeaa1e650ad94ff85f668dac",
/*win64=*/"66516152eeaa1e650ad94ff85f668dac", "android_arm32_payload",
@@ -1831,7 +1831,7 @@
ASSERT_TRUE(SetUpSender(kTestFileMono32kHz, 32000));
ASSERT_NO_FATAL_FAILURE(
SetUpTestExternalEncoder(std::move(mock_encoder), config.payload_type));
- Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",
+ Run("c8d1fc677f33c2022ec5f83c7f302280", "8f9b8750bd80fe26b6cbf6659b89f0f9",
50, test::AcmReceiveTestOldApi::kMonoOutput);
}
diff --git a/modules/audio_coding/neteq/delay_manager.cc b/modules/audio_coding/neteq/delay_manager.cc
index 97d1d2e..9f6b269 100644
--- a/modules/audio_coding/neteq/delay_manager.cc
+++ b/modules/audio_coding/neteq/delay_manager.cc
@@ -20,6 +20,7 @@
#include "modules/include/module_common_types_public.h"
#include "rtc_base/checks.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
@@ -45,9 +46,17 @@
} // namespace
DelayManager::Config::Config() {
- Parser()->Parse(webrtc::field_trial::FindFullName(
- "WebRTC-Audio-NetEqDelayManagerConfig"));
- MaybeUpdateFromLegacyFieldTrial();
+ StructParametersParser::Create( //
+ "quantile", &quantile, //
+ "forget_factor", &forget_factor, //
+ "start_forget_weight", &start_forget_weight, //
+ "resample_interval_ms", &resample_interval_ms, //
+ "max_history_ms", &max_history_ms, //
+ "use_reorder_optimizer", &use_reorder_optimizer, //
+ "reorder_forget_factor", &reorder_forget_factor, //
+ "ms_per_loss_percent", &ms_per_loss_percent)
+ ->Parse(webrtc::field_trial::FindFullName(
+ "WebRTC-Audio-NetEqDelayManagerConfig"));
}
void DelayManager::Config::Log() {
@@ -63,42 +72,6 @@
<< " ms_per_loss_percent=" << ms_per_loss_percent;
}
-std::unique_ptr<StructParametersParser> DelayManager::Config::Parser() {
- return StructParametersParser::Create( //
- "quantile", &quantile, //
- "forget_factor", &forget_factor, //
- "start_forget_weight", &start_forget_weight, //
- "resample_interval_ms", &resample_interval_ms, //
- "max_history_ms", &max_history_ms, //
- "use_reorder_optimizer", &use_reorder_optimizer, //
- "reorder_forget_factor", &reorder_forget_factor, //
- "ms_per_loss_percent", &ms_per_loss_percent);
-}
-
-// TODO(jakobi): remove legacy field trial.
-void DelayManager::Config::MaybeUpdateFromLegacyFieldTrial() {
- constexpr char kDelayHistogramFieldTrial[] =
- "WebRTC-Audio-NetEqDelayHistogram";
- if (!webrtc::field_trial::IsEnabled(kDelayHistogramFieldTrial)) {
- return;
- }
- const auto field_trial_string =
- webrtc::field_trial::FindFullName(kDelayHistogramFieldTrial);
- double percentile = -1.0;
- double forget_factor = -1.0;
- double start_forget_weight = -1.0;
- if (sscanf(field_trial_string.c_str(), "Enabled-%lf-%lf-%lf", &percentile,
- &forget_factor, &start_forget_weight) >= 2 &&
- percentile >= 0.0 && percentile <= 100.0 && forget_factor >= 0.0 &&
- forget_factor <= 1.0) {
- this->quantile = percentile / 100;
- this->forget_factor = forget_factor;
- this->start_forget_weight = start_forget_weight >= 1
- ? absl::make_optional(start_forget_weight)
- : absl::nullopt;
- }
-}
-
DelayManager::DelayManager(const Config& config, const TickTimer* tick_timer)
: max_packets_in_buffer_(config.max_packets_in_buffer),
underrun_optimizer_(tick_timer,
diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h
index 277b80d..410aa94 100644
--- a/modules/audio_coding/neteq/delay_manager.h
+++ b/modules/audio_coding/neteq/delay_manager.h
@@ -23,7 +23,6 @@
#include "modules/audio_coding/neteq/reorder_optimizer.h"
#include "modules/audio_coding/neteq/underrun_optimizer.h"
#include "rtc_base/constructor_magic.h"
-#include "rtc_base/experiments/struct_parameters_parser.h"
namespace webrtc {
@@ -34,10 +33,10 @@
void Log();
// Options that can be configured via field trial.
- double quantile = 0.97;
- double forget_factor = 0.9993;
+ double quantile = 0.95;
+ double forget_factor = 0.983;
absl::optional<double> start_forget_weight = 2;
- absl::optional<int> resample_interval_ms;
+ absl::optional<int> resample_interval_ms = 500;
int max_history_ms = 2000;
bool use_reorder_optimizer = true;
@@ -47,12 +46,6 @@
// Options that are externally populated.
int max_packets_in_buffer = 200;
int base_minimum_delay_ms = 0;
-
- private:
- std::unique_ptr<StructParametersParser> Parser();
-
- // TODO(jakobi): remove legacy field trial.
- void MaybeUpdateFromLegacyFieldTrial();
};
DelayManager(const Config& config, const TickTimer* tick_timer);
diff --git a/modules/audio_coding/neteq/delay_manager_unittest.cc b/modules/audio_coding/neteq/delay_manager_unittest.cc
index fc4e0cb..ee35306 100644
--- a/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -44,8 +44,8 @@
absl::optional<int> InsertNextPacket();
void IncreaseTime(int inc_ms);
- DelayManager dm_;
TickTimer tick_timer_;
+ DelayManager dm_;
uint32_t ts_;
};
@@ -74,39 +74,18 @@
}
TEST_F(DelayManagerTest, UpdateNormal) {
- // First packet arrival.
- InsertNextPacket();
- // Advance time by one frame size.
- IncreaseTime(kFrameSizeMs);
- // Second packet arrival.
- InsertNextPacket();
+ for (int i = 0; i < 50; ++i) {
+ InsertNextPacket();
+ IncreaseTime(kFrameSizeMs);
+ }
EXPECT_EQ(20, dm_.TargetDelayMs());
}
-TEST_F(DelayManagerTest, UpdateLongInterArrivalTime) {
- // First packet arrival.
- InsertNextPacket();
- // Advance time by two frame size.
- IncreaseTime(2 * kFrameSizeMs);
- // Second packet arrival.
- InsertNextPacket();
- EXPECT_EQ(40, dm_.TargetDelayMs());
-}
-
TEST_F(DelayManagerTest, MaxDelay) {
- const int kExpectedTarget = 5 * kFrameSizeMs;
- // First packet arrival.
InsertNextPacket();
- // Second packet arrival.
- IncreaseTime(kExpectedTarget);
- InsertNextPacket();
-
- // No limit is set.
- EXPECT_EQ(kExpectedTarget, dm_.TargetDelayMs());
-
- const int kMaxDelayMs = 3 * kFrameSizeMs;
+ const int kMaxDelayMs = 60;
+ EXPECT_GT(dm_.TargetDelayMs(), kMaxDelayMs);
EXPECT_TRUE(dm_.SetMaximumDelay(kMaxDelayMs));
- IncreaseTime(kFrameSizeMs);
InsertNextPacket();
EXPECT_EQ(kMaxDelayMs, dm_.TargetDelayMs());
@@ -115,17 +94,9 @@
}
TEST_F(DelayManagerTest, MinDelay) {
- const int kExpectedTarget = 5 * kFrameSizeMs;
- // First packet arrival.
InsertNextPacket();
- // Second packet arrival.
- IncreaseTime(kExpectedTarget);
- InsertNextPacket();
-
- // No limit is applied.
- EXPECT_EQ(kExpectedTarget, dm_.TargetDelayMs());
-
int kMinDelayMs = 7 * kFrameSizeMs;
+ EXPECT_LT(dm_.TargetDelayMs(), kMinDelayMs);
dm_.SetMinimumDelay(kMinDelayMs);
IncreaseTime(kFrameSizeMs);
InsertNextPacket();
@@ -251,48 +222,11 @@
}
TEST_F(DelayManagerTest, BaseMinimumDelay) {
- const int kExpectedTarget = 5 * kFrameSizeMs;
// First packet arrival.
InsertNextPacket();
- // Second packet arrival.
- IncreaseTime(kExpectedTarget);
- InsertNextPacket();
-
- // No limit is applied.
- EXPECT_EQ(kExpectedTarget, dm_.TargetDelayMs());
constexpr int kBaseMinimumDelayMs = 7 * kFrameSizeMs;
- EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
- EXPECT_EQ(dm_.GetBaseMinimumDelay(), kBaseMinimumDelayMs);
-
- IncreaseTime(kFrameSizeMs);
- InsertNextPacket();
- EXPECT_EQ(dm_.GetBaseMinimumDelay(), kBaseMinimumDelayMs);
- EXPECT_EQ(kBaseMinimumDelayMs, dm_.TargetDelayMs());
-}
-
-TEST_F(DelayManagerTest, BaseMinimumDelayAffectsTargetDelay) {
- const int kExpectedTarget = 5;
- const int kTimeIncrement = kExpectedTarget * kFrameSizeMs;
- // First packet arrival.
- InsertNextPacket();
- // Second packet arrival.
- IncreaseTime(kTimeIncrement);
- InsertNextPacket();
-
- // No limit is applied.
- EXPECT_EQ(kTimeIncrement, dm_.TargetDelayMs());
-
- // Minimum delay is lower than base minimum delay, that is why base minimum
- // delay is used to calculate target level.
- constexpr int kMinimumDelayPackets = kExpectedTarget + 1;
- constexpr int kBaseMinimumDelayPackets = kExpectedTarget + 2;
-
- constexpr int kMinimumDelayMs = kMinimumDelayPackets * kFrameSizeMs;
- constexpr int kBaseMinimumDelayMs = kBaseMinimumDelayPackets * kFrameSizeMs;
-
- EXPECT_TRUE(kMinimumDelayMs < kBaseMinimumDelayMs);
- EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+ EXPECT_LT(dm_.TargetDelayMs(), kBaseMinimumDelayMs);
EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
EXPECT_EQ(dm_.GetBaseMinimumDelay(), kBaseMinimumDelayMs);
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 45c5229..3696427 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -1370,7 +1370,7 @@
}
}
- timestamp_ = end_timestamp;
+ timestamp_ = sync_buffer_->end_timestamp();
return 0;
}
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 7010adf..e6e3eb4 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -1026,22 +1026,37 @@
EXPECT_CALL(mock_decoder, PacketDuration(nullptr, 0))
.WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
- // Pointee(x) verifies that first byte of the payload equals x, this makes it
- // possible to verify that the correct payload is fed to Decode().
- EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(0), kPayloadLengthBytes,
- kSampleRateKhz * 1000, _, _))
- .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
- dummy_output + kPayloadLengthSamples),
- SetArgPointee<4>(AudioDecoder::kSpeech),
- Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+ EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+ SdpAudioFormat("opus", 48000, 2)));
- EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(1), kPayloadLengthBytes,
- kSampleRateKhz * 1000, _, _))
- .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
- dummy_output + kPayloadLengthSamples),
- SetArgPointee<4>(AudioDecoder::kComfortNoise),
- Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+ struct Packet {
+ int sequence_number_delta;
+ int timestamp_delta;
+ AudioDecoder::SpeechType decoder_output_type;
+ };
+ std::vector<Packet> packets = {
+ {0, 0, AudioDecoder::kSpeech},
+ {1, kPayloadLengthSamples, AudioDecoder::kComfortNoise},
+ {2, 2 * kPayloadLengthSamples, AudioDecoder::kSpeech},
+ {1, kPayloadLengthSamples, AudioDecoder::kSpeech}};
+ for (size_t i = 0; i < packets.size(); ++i) {
+ rtp_header.sequenceNumber += packets[i].sequence_number_delta;
+ rtp_header.timestamp += packets[i].timestamp_delta;
+ payload[0] = i;
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+ // Pointee(x) verifies that first byte of the payload equals x, this makes
+ // it possible to verify that the correct payload is fed to Decode().
+ EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(i), kPayloadLengthBytes,
+ kSampleRateKhz * 1000, _, _))
+ .WillOnce(DoAll(SetArrayArgument<3>(
+ dummy_output, dummy_output + kPayloadLengthSamples),
+ SetArgPointee<4>(packets[i].decoder_output_type),
+ Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+ }
+
+ // Expect comfort noise to be returned by the decoder.
EXPECT_CALL(mock_decoder,
DecodeInternal(IsNull(), 0, kSampleRateKhz * 1000, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
@@ -1049,87 +1064,24 @@
SetArgPointee<4>(AudioDecoder::kComfortNoise),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
- EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(2), kPayloadLengthBytes,
- kSampleRateKhz * 1000, _, _))
- .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
- dummy_output + kPayloadLengthSamples),
- SetArgPointee<4>(AudioDecoder::kSpeech),
- Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+ std::vector<AudioFrame::SpeechType> expected_output = {
+ AudioFrame::kNormalSpeech, AudioFrame::kCNG, AudioFrame::kNormalSpeech};
+ size_t output_index = 0;
- EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
- SdpAudioFormat("opus", 48000, 2)));
-
- const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateKhz);
- AudioFrame output;
- AudioFrame::SpeechType expected_type[8] = {
- AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech, AudioFrame::kCNG,
- AudioFrame::kCNG, AudioFrame::kCNG, AudioFrame::kCNG,
- AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech};
- int expected_timestamp_increment[8] = {
- -1, // will not be used.
- 10 * kSampleRateKhz,
- -1,
- -1, // timestamp will be empty during CNG mode; indicated by -1 here.
- -1,
- -1,
- 50 * kSampleRateKhz,
- 10 * kSampleRateKhz};
-
- // Insert one packet (decoder will return speech).
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
-
- bool muted;
- EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
- absl::optional<uint32_t> last_timestamp = neteq_->GetPlayoutTimestamp();
- ASSERT_TRUE(last_timestamp);
-
- // Insert second packet (decoder will return CNG).
- payload[0] = 1;
- rtp_header.sequenceNumber++;
- rtp_header.timestamp += kPayloadLengthSamples;
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
-
- // Lambda for verifying the timestamps.
- auto verify_timestamp = [&last_timestamp, &expected_timestamp_increment](
- absl::optional<uint32_t> ts, size_t i) {
- if (expected_timestamp_increment[i] == -1) {
- // Expect to get an empty timestamp value during CNG and PLC.
- EXPECT_FALSE(ts) << "i = " << i;
+ int timeout_counter = 0;
+ while (!packet_buffer_->Empty()) {
+ ASSERT_LT(timeout_counter++, 20) << "Test timed out";
+ AudioFrame output;
+ bool muted;
+ EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+ if (output_index + 1 < expected_output.size() &&
+ output.speech_type_ == expected_output[output_index + 1]) {
+ ++output_index;
} else {
- ASSERT_TRUE(ts) << "i = " << i;
- EXPECT_EQ(*ts, *last_timestamp + expected_timestamp_increment[i])
- << "i = " << i;
- last_timestamp = ts;
+ EXPECT_EQ(output.speech_type_, expected_output[output_index]);
}
- };
-
- for (size_t i = 1; i < 6; ++i) {
- ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
- EXPECT_EQ(1u, output.num_channels_);
- EXPECT_EQ(expected_type[i - 1], output.speech_type_);
- EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
- SCOPED_TRACE("");
- verify_timestamp(neteq_->GetPlayoutTimestamp(), i);
}
- // Insert third packet, which leaves a gap from last packet.
- payload[0] = 2;
- rtp_header.sequenceNumber += 2;
- rtp_header.timestamp += 2 * kPayloadLengthSamples;
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
-
- for (size_t i = 6; i < 8; ++i) {
- ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
- EXPECT_EQ(1u, output.num_channels_);
- EXPECT_EQ(expected_type[i - 1], output.speech_type_);
- EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
- SCOPED_TRACE("");
- verify_timestamp(neteq_->GetPlayoutTimestamp(), i);
- }
-
- // Now check the packet buffer, and make sure it is empty.
- EXPECT_TRUE(packet_buffer_->Empty());
-
EXPECT_CALL(mock_decoder, Die());
}
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index 862edaf..2c68501 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -274,7 +274,7 @@
// Next we introduce packet losses.
SetPacketLossRate(0.1);
- expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 1065;
+ expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 898;
RunTest(50, expects);
// Next we enable FEC.
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 341ef6b..c55f6f1 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -82,17 +82,17 @@
const std::string input_rtp_file =
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
- const std::string output_checksum = PlatformChecksum(
- "6c35140ce4d75874bdd60aa1872400b05fd05ca2",
- "ab451bb8301d9a92fbf4de91556b56f1ea38b4ce", "not used",
- "6c35140ce4d75874bdd60aa1872400b05fd05ca2",
- "64b46bb3c1165537a880ae8404afce2efba456c0");
+ const std::string output_checksum =
+ PlatformChecksum("ba4fae83a52f5e9d95b0910f05d540114285697b",
+ "aa557f30f7fdcebbbbf99d7f235ccba3a1c98983", "not used",
+ "ba4fae83a52f5e9d95b0910f05d540114285697b",
+ "64b46bb3c1165537a880ae8404afce2efba456c0");
- const std::string network_stats_checksum = PlatformChecksum(
- "90594d85fa31d3d9584d79293bf7aa4ee55ed751",
- "77b9c3640b81aff6a38d69d07dd782d39c15321d", "not used",
- "90594d85fa31d3d9584d79293bf7aa4ee55ed751",
- "90594d85fa31d3d9584d79293bf7aa4ee55ed751");
+ const std::string network_stats_checksum =
+ PlatformChecksum("fa878a8464ef1cb3d01503b7f927c3e2ce6f02c4",
+ "300ccc2aaee7ed1971afb2f9a20247ed8760441d", "not used",
+ "fa878a8464ef1cb3d01503b7f927c3e2ce6f02c4",
+ "fa878a8464ef1cb3d01503b7f927c3e2ce6f02c4");
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
absl::GetFlag(FLAGS_gen_ref));
@@ -531,11 +531,16 @@
out_frame_.timestamp_ + out_frame_.samples_per_channel_);
}
- // Insert speech again.
++seq_no;
timestamp += kCngPeriodSamples;
- PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+ uint32_t first_speech_timestamp = timestamp;
+ // Insert speech again.
+ for (int i = 0; i < 3; ++i) {
+ PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+ ++seq_no;
+ timestamp += kSamples;
+ }
// Pull audio once and verify that the output is speech again.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
@@ -543,7 +548,7 @@
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
- EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
+ EXPECT_EQ(first_speech_timestamp + kSamples - algorithmic_delay_samples,
*playout_timestamp);
}
@@ -1263,7 +1268,7 @@
// The base delay values are taken from the resuts of the non-delayed case in
// NetEqOutputDelayTest.RunTest above.
EXPECT_EQ(20 + kExpectedDelayMs, result.target_delay_ms);
- EXPECT_EQ(24 + kExpectedDelayMs, result.filtered_current_delay_ms);
+ EXPECT_EQ(60 + kExpectedDelayMs, result.filtered_current_delay_ms);
}
// Set a non-multiple-of-10 value in the field trial, and verify that we don't
@@ -1278,7 +1283,7 @@
// The base delay values are taken from the resuts of the non-delayed case in
// NetEqOutputDelayTest.RunTest above.
EXPECT_EQ(20 + kRoundedDelayMs, result.target_delay_ms);
- EXPECT_EQ(24 + kRoundedDelayMs, result.filtered_current_delay_ms);
+ EXPECT_EQ(60 + kRoundedDelayMs, result.filtered_current_delay_ms);
}
} // namespace test
diff --git a/modules/audio_coding/neteq/relative_arrival_delay_tracker.cc b/modules/audio_coding/neteq/relative_arrival_delay_tracker.cc
index 02c5a43..b50ac80 100644
--- a/modules/audio_coding/neteq/relative_arrival_delay_tracker.cc
+++ b/modules/audio_coding/neteq/relative_arrival_delay_tracker.cc
@@ -49,7 +49,7 @@
void RelativeArrivalDelayTracker::Reset() {
delay_history_.clear();
- packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
+ packet_iat_stopwatch_.reset();
newest_timestamp_ = absl::nullopt;
last_timestamp_ = absl::nullopt;
}
diff --git a/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
index 1c70f14..a82f93b 100644
--- a/modules/audio_coding/neteq/test/neteq_decoding_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
@@ -245,15 +245,9 @@
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
- // Due to internal NetEq logic, preferred buffer-size is about 4 times the
- // packet size for first few packets. Therefore we refrain from checking
- // the criteria.
- 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 + algorithmic_delay_ms_);
- }
+ EXPECT_LE(network_stats.preferred_buffer_size_ms, 80);
+ EXPECT_LE(network_stats.current_buffer_size_ms,
+ 80 + algorithmic_delay_ms_);
last_seq_no = seq_no;
last_timestamp = timestamp;
diff --git a/modules/audio_coding/neteq/underrun_optimizer.cc b/modules/audio_coding/neteq/underrun_optimizer.cc
index dad0424..baed812 100644
--- a/modules/audio_coding/neteq/underrun_optimizer.cc
+++ b/modules/audio_coding/neteq/underrun_optimizer.cc
@@ -63,7 +63,7 @@
void UnderrunOptimizer::Reset() {
histogram_.Reset();
- resample_stopwatch_ = tick_timer_->GetNewStopwatch();
+ resample_stopwatch_.reset();
max_delay_in_interval_ms_ = 0;
optimal_delay_ms_.reset();
}
