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Revert "Change default NetEq sample rate to 48k."

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This reverts commit 38fcd58429b29c9474f1647efed7ebeb543c0637.

Reason for revert: Breaks downstream test

Original change's description:
> Change default NetEq sample rate to 48k.
>
> This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k).
>
> Bug: none
> Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662
> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38536}

Bug: none
Change-Id: I03181168ab14d2d99320767c3a25ba3cfb726b2c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281441
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Auto-Submit: Jakob Ivarsson‎ <jakobi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38538}
This commit is contained in:
Jakob Ivarsson‎
2022-11-02 15:56:03 +00:00
committed by WebRTC LUCI CQ
parent 67416b51e4
commit 8f7ad88d0e
2 changed files with 5 additions and 5 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
api/neteq/neteq.h
View File

@ -128,7 +128,7 @@ class NetEq {
std::string ToString() const; std::string ToString() const;
int sample_rate_hz = 48000; // Initial value. Will change with input data. int sample_rate_hz = 16000; // Initial value. Will change with input data.
bool enable_post_decode_vad = false; bool enable_post_decode_vad = false;
size_t max_packets_in_buffer = 200; size_t max_packets_in_buffer = 200;
int max_delay_ms = 0; int max_delay_ms = 0;

8
modules/audio_coding/acm2/audio_coding_module_unittest.cc
View File

@ -1030,7 +1030,7 @@ class AcmSenderBitExactnessNewApi : public AcmSenderBitExactnessOldApi {};
defined(NDEBUG) && defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) defined(NDEBUG) && defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) { TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480)); ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
Run(/*audio_checksum_ref=*/"37ecdabad1698a857cf811e6d1fa91df", Run(/*audio_checksum_ref=*/"a3077ac01b0137e8bbc237fb1f9816a5",
/*payload_checksum_ref=*/"3c79f16f34218271f3dca4e2b1dfe1bb", /*payload_checksum_ref=*/"3c79f16f34218271f3dca4e2b1dfe1bb",
/*expected_packets=*/33, /*expected_packets=*/33,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
@ -1038,7 +1038,7 @@ TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) { TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960)); ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
Run(/*audio_checksum_ref=*/"0e9078d23454901496a88362ba0740c3", Run(/*audio_checksum_ref=*/"76da9b7514f986fc2bb32b1c3170e8d4",
/*payload_checksum_ref=*/"9e0a0ab743ad987b55b8e14802769c56", /*payload_checksum_ref=*/"9e0a0ab743ad987b55b8e14802769c56",
/*expected_packets=*/16, /*expected_packets=*/16,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
@ -1067,7 +1067,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) { TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160)); ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160));
Run(/*audio_checksum_ref=*/"f95c87bdd33f631bcf80f4b19445bbd2", Run(/*audio_checksum_ref=*/"bc6ab94d12a464921763d7544fdbd07e",
/*payload_checksum_ref=*/"ad786526383178b08d80d6eee06e9bad", /*payload_checksum_ref=*/"ad786526383178b08d80d6eee06e9bad",
/*expected_packets=*/100, /*expected_packets=*/100,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
@ -1151,7 +1151,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Ilbc_30ms) {
#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) #if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
TEST_F(AcmSenderBitExactnessOldApi, G722_20ms) { TEST_F(AcmSenderBitExactnessOldApi, G722_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160)); ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
Run(/*audio_checksum_ref=*/"f5264affff25cf2cbd2e1e8a5217f9a3", Run(/*audio_checksum_ref=*/"a87a91ec0124510a64967f5d768554ff",
/*payload_checksum_ref=*/"fc68a87e1380614e658087cb35d5ca10", /*payload_checksum_ref=*/"fc68a87e1380614e658087cb35d5ca10",
/*expected_packets=*/50, /*expected_packets=*/50,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);