From 79ded653fee7183d5c0d94c5addf570bcfb29c9e Mon Sep 17 00:00:00 2001 From: Gustaf Ullberg Date: Fri, 18 May 2018 14:48:48 +0200 Subject: [PATCH] Update expected bitrate in Opus tests Upstream changes to Opus DTX behavior changes the bitrates of Opus. This CL re-enables recently disabled unittests and updates the expected bitrates. Bug: webrtc:9280 Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2 Reviewed-on: https://webrtc-review.googlesource.com/77766 Reviewed-by: Henrik Lundin Commit-Queue: Gustaf Ullberg Cr-Commit-Position: refs/heads/master@{#23306} --- .../acm2/audio_coding_module_unittest.cc | 36 +++++++------------ 1 file changed, 12 insertions(+), 24 deletions(-) diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc index 57471ec106..1664b26e90 100644 --- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc +++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc @@ -1634,20 +1634,16 @@ class AcmSetBitRateNewApi : public AcmSetBitRateTest { void Run(int expected_total_bits) { RunInner(expected_total_bits); } }; -// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled -// into WebRTC. -TEST_F(AcmSetBitRateOldApi, DISABLED_Opus_48khz_20ms_10kbps) { +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, 8680); + Run(10000, 8696); #endif // WEBRTC_ANDROID } -// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled -// into WebRTC. -TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_10kbps) { +TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) { const auto config = AudioEncoderOpus::SdpToConfig( SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}})); const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107); @@ -1656,24 +1652,20 @@ TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_10kbps) { #if defined(WEBRTC_ANDROID) RunInner(8640); #else - RunInner(8680); + RunInner(8696); #endif // WEBRTC_ANDROID } -// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled -// into WebRTC. -TEST_F(AcmSetBitRateOldApi, DISABLED_Opus_48khz_20ms_50kbps) { +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, 45520); + Run(50000, 45600); #endif // WEBRTC_ANDROID } -// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled -// into WebRTC. -TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_50kbps) { +TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) { const auto config = AudioEncoderOpus::SdpToConfig( SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}})); const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107); @@ -1682,7 +1674,7 @@ TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_50kbps) { #if defined(WEBRTC_ANDROID) RunInner(45792); #else - RunInner(45520); + RunInner(45600); #endif // WEBRTC_ANDROID } @@ -1774,25 +1766,21 @@ class AcmChangeBitRateOldApi : public AcmSetBitRateOldApi { uint32_t frame_size_samples_; }; -// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled -// into WebRTC. -TEST_F(AcmChangeBitRateOldApi, DISABLED_Opus_48khz_20ms_10kbps_2) { +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, 5368); + Run(10000, 32200, 5208); #endif // WEBRTC_ANDROID } -// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled -// into WebRTC. -TEST_F(AcmChangeBitRateOldApi, DISABLED_Opus_48khz_20ms_50kbps_2) { +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, 23920); + Run(50000, 32200, 23928); #endif // WEBRTC_ANDROID }