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Update expected bitrate in Opus tests

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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 <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23306}
This commit is contained in:
Gustaf Ullberg
2018-05-18 14:48:48 +02:00
committed by Commit Bot
parent 2d9a3b1aba
commit 79ded653fe
1 changed files with 12 additions and 24 deletions
Download Patch File Download Diff File Expand all files Collapse all files

36
modules/audio_coding/acm2/audio_coding_module_unittest.cc
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@ -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
}