Changed OLA window for neteq. Old code didnt work well with 48khz
fixing white spaces
updated authors file
Changed OLA window to use Q14 as Q5 dosnt work with 48khz. 1 ms @ 48 khz is > 2^5
BUG=webrtc:1361
Review-Url: https://codereview.webrtc.org/2763273003
Cr-Commit-Position: refs/heads/master@{#17611}
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 98569a6..58e7b69 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -979,31 +979,31 @@
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
- Run(8000, PlatformChecksum("25cda36a1b967e75c0eb580924247681",
- "bbfe6a07f8bca872b5370885825ee061",
- "d5b9ae44d03dbd7c921dd9c228e03cc5",
- "4d851d1f2e4b8a2f1727fac8fba4b1e1"));
+ Run(8000, PlatformChecksum("2adede965c6f87de7142c51552111d08",
+ "028c0fc414b1c9ab7e582dccdf381e98",
+ "36c95170c1393d4b765d1c17b61ef977",
+ "4598140b5e4f7ee66c5adad609e65a3e"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
- Run(16000, PlatformChecksum("9c7b6f586c4b9d6d0195372660991353",
- "1ab45baa674e681ec394e0d3824d8605",
- "dd4e7f2521b5f47c0016b12f06c08695",
- "5401b64b6dbe7f090f846e89b0d858ce"));
+ Run(16000, PlatformChecksum("c2550a3db7632de409e8db0093df1c12",
+ "edd31f4b6665cd5b9041fb93f2316594",
+ "22128bca51650cb61c80bed63b595603",
+ "f2aad418af974a3b1694d5ae5cc2c3c7"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
- Run(32000, PlatformChecksum("599b9484ca89615641ebd767cccb149f",
- "9f7d51569647eff38026dd815d43ca91",
- "78d00d2a3f8f307fc3835ca588a18f3a",
- "d335eebc72f4d087aa397a9cf8f4967b"));
+ Run(32000, PlatformChecksum("85e28d7950132d56f90b099c90f82153",
+ "7b903f5c89997f271b405e63c245ef45",
+ "8b8fc6c6fd1dcdcfb3dd90e1ce597f10",
+ "100869c8dcde51346c2073e52a272d98"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
- Run(48000, PlatformChecksum("5d3b4357c9044264bb4a601b6548bd55",
- "8607778183d7ad02b8ce37eeeba4f37c",
- "fd71398d336b88cbd4fb5002846e91c6",
- "8ce7e0e1c381d920ee7b57751b257de8"));
+ Run(48000, PlatformChecksum("ab611510e8fd6d5210a23cc04d3f0e8e",
+ "d8609bc9b495d81f29779344c68bcc47",
+ "ec5ebb90cda0ea5bb89e79d698af65de",
+ "bd44bf97e7899186532f91235cef444d"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
@@ -1083,10 +1083,10 @@
rtc::scoped_refptr<rtc::RefCountedObject<ADFactory>> factory(
new rtc::RefCountedObject<ADFactory>);
- Run(48000, PlatformChecksum("5d3b4357c9044264bb4a601b6548bd55",
- "8607778183d7ad02b8ce37eeeba4f37c",
- "fd71398d336b88cbd4fb5002846e91c6",
- "8ce7e0e1c381d920ee7b57751b257de8"),
+ Run(48000, PlatformChecksum("ab611510e8fd6d5210a23cc04d3f0e8e",
+ "d8609bc9b495d81f29779344c68bcc47",
+ "ec5ebb90cda0ea5bb89e79d698af65de",
+ "bd44bf97e7899186532f91235cef444d"),
factory, [](AudioCodingModule* acm) {
acm->RegisterReceiveCodec(0, {"MockPCMu", 8000, 1});
});
@@ -1273,10 +1273,10 @@
TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
- "0b58f9eeee43d5891f5f6c75e77984a3",
- "c7e5bdadfa2871df95639fcc297cf23d",
- "0499ca260390769b3172136faad925b9",
- "866abf524acd2807efbe65e133c23f95"),
+ "2c9cb15d4ed55b5a0cadd04883bc73b0",
+ "9336a9b993cbd8a751f0e8958e66c89c",
+ "bd4682225f7c4ad5f2049f6769713ac2",
+ "343f1f42be0607c61e6516aece424609"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"3c79f16f34218271f3dca4e2b1dfe1bb",
"d42cb5195463da26c8129bbfe73a22e6",
@@ -1290,7 +1290,7 @@
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"1ad29139a04782a33daad8c2b9b35875",
"14d63c5f08127d280e722e3191b73bdd",
- "8da003e16c5371af2dc2be79a50f9076",
+ "edcf26694c289e3d9691faf79b74f09f",
"ef75e900e6f375e3061163c53fd09a63"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"9e0a0ab743ad987b55b8e14802769c56",
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 47073fb..33b4005 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -443,10 +443,10 @@
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
const std::string output_checksum = PlatformChecksum(
- "5a8184bc60c0d7dddb50af8966360675476a8d8b",
- "be982d2c5685dd1ca4ea5d352283df50e8e5b46d",
- "5a8184bc60c0d7dddb50af8966360675476a8d8b",
- "c86aec95439748f4949de95b50c94be291118615");
+ "09fa7646e2ad032a0b156177b95f09012430f81f",
+ "1c64eb8b55ce8878676c6a1e6ddd78f48de0668b",
+ "09fa7646e2ad032a0b156177b95f09012430f81f",
+ "759fef89a5de52bd17e733dc255c671ce86be909");
const std::string network_stats_checksum = PlatformChecksum(
"f59b3dfdb9b1b8bbb61abedd7c8cf3fc47c21f5f",
@@ -480,10 +480,10 @@
webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
const std::string output_checksum = PlatformChecksum(
- "9d7d52bc94e941d106aa518f324f16a58d231586",
- "9d7d52bc94e941d106aa518f324f16a58d231586",
- "9d7d52bc94e941d106aa518f324f16a58d231586",
- "9d7d52bc94e941d106aa518f324f16a58d231586");
+ "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
+ "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
+ "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
+ "6237dd113ad80d7764fe4c90b55b2ec035eae64e");
const std::string network_stats_checksum = PlatformChecksum(
"d8379381d5a619f0616bb3c0a8a9eea1704a8ab8",
diff --git a/webrtc/modules/audio_coding/neteq/normal.cc b/webrtc/modules/audio_coding/neteq/normal.cc
index 3dee39a..1a7bc68 100644
--- a/webrtc/modules/audio_coding/neteq/normal.cc
+++ b/webrtc/modules/audio_coding/neteq/normal.cc
@@ -130,23 +130,25 @@
// Interpolate the expanded data into the new vector.
// (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
- RTC_DCHECK_LT(fs_shift, 3); // Will always be 0, 1, or, 2.
- increment = 4 >> fs_shift;
- int fraction = increment;
- // Don't interpolate over more samples than what is in output. When this
- // cap strikes, the interpolation will likely sound worse, but this is an
- // emergency operation in response to unexpected input.
- const size_t interp_len_samples =
- std::min(static_cast<size_t>(8 * fs_mult), output->Size());
- for (size_t i = 0; i < interp_len_samples; ++i) {
- // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8
- // now for legacy bit-exactness.
- RTC_DCHECK_LT(channel_ix, output->Channels());
- RTC_DCHECK_LT(i, output->Size());
+ size_t win_length = samples_per_ms_;
+ int16_t win_slope_Q14 = default_win_slope_Q14_;
+ RTC_DCHECK_LT(channel_ix, output->Channels());
+ if (win_length > output->Size()) {
+ win_length = output->Size();
+ win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
+ }
+ int16_t win_up_Q14 = 0;
+ for (size_t i = 0; i < win_length; i++) {
+ win_up_Q14 += win_slope_Q14;
(*output)[channel_ix][i] =
- static_cast<int16_t>((fraction * (*output)[channel_ix][i] +
- (32 - fraction) * expanded[channel_ix][i] + 8) >> 5);
- fraction += increment;
+ (win_up_Q14 * (*output)[channel_ix][i] +
+ ((1 << 14) - win_up_Q14) * expanded[channel_ix][i] + (1 << 13)) >>
+ 14;
+ }
+ if (fs_hz_ == 48000) {
+ RTC_DCHECK_EQ(win_up_Q14, (1 << 14) - 16);
+ } else {
+ RTC_DCHECK_EQ(win_up_Q14, 1 << 14);
}
}
} else if (last_mode == kModeRfc3389Cng) {
@@ -171,15 +173,24 @@
}
// Interpolate the CNG into the new vector.
// (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
- RTC_DCHECK_LT(fs_shift, 3); // Will always be 0, 1, or, 2.
- int16_t increment = 4 >> fs_shift;
- int16_t fraction = increment;
- for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
- // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
- // for legacy bit-exactness.
- (*output)[0][i] = (fraction * (*output)[0][i] +
- (32 - fraction) * cng_output[i] + 8) >> 5;
- fraction += increment;
+ size_t win_length = samples_per_ms_;
+ int16_t win_slope_Q14 = default_win_slope_Q14_;
+ if (win_length > kCngLength) {
+ win_length = kCngLength;
+ win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
+ }
+ int16_t win_up_Q14 = 0;
+ for (size_t i = 0; i < win_length; i++) {
+ win_up_Q14 += win_slope_Q14;
+ (*output)[0][i] =
+ (win_up_Q14 * (*output)[0][i] +
+ ((1 << 14) - win_up_Q14) * cng_output[i] + (1 << 13)) >>
+ 14;
+ }
+ if (fs_hz_ == 48000) {
+ RTC_DCHECK_EQ(win_up_Q14, (1 << 14) - 16);
+ } else {
+ RTC_DCHECK_EQ(win_up_Q14, 1 << 14);
}
} else if (external_mute_factor_array[0] < 16384) {
// Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
diff --git a/webrtc/modules/audio_coding/neteq/normal.h b/webrtc/modules/audio_coding/neteq/normal.h
index 23887f5..019bcf8 100644
--- a/webrtc/modules/audio_coding/neteq/normal.h
+++ b/webrtc/modules/audio_coding/neteq/normal.h
@@ -15,7 +15,9 @@
#include <vector>
+#include "webrtc/base/checks.h"
#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/safe_conversions.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/defines.h"
#include "webrtc/typedefs.h"
@@ -32,14 +34,17 @@
// no other "special circumstances" are at hand.
class Normal {
public:
- Normal(int fs_hz, DecoderDatabase* decoder_database,
+ Normal(int fs_hz,
+ DecoderDatabase* decoder_database,
const BackgroundNoise& background_noise,
Expand* expand)
: fs_hz_(fs_hz),
decoder_database_(decoder_database),
background_noise_(background_noise),
- expand_(expand) {
- }
+ expand_(expand),
+ samples_per_ms_(rtc::CheckedDivExact(fs_hz_, 1000)),
+ default_win_slope_Q14_(
+ rtc::dchecked_cast<uint16_t>((1 << 14) / samples_per_ms_)) {}
virtual ~Normal() {}
@@ -60,6 +65,8 @@
DecoderDatabase* decoder_database_;
const BackgroundNoise& background_noise_;
Expand* expand_;
+ const size_t samples_per_ms_;
+ const int16_t default_win_slope_Q14_;
RTC_DISALLOW_COPY_AND_ASSIGN(Normal);
};