NetEq: Change NetEq's ramp-up behavior after expansions
NetEq tapers down the audio produced through loss concealment when the
expansion has been going on for some time. When the audio packets starts
coming in again, there is a ramp-up that happens. This ramp-up could
before this change extend over more than one 10 ms block, which made
keeping track of the scaling factor necessary. With this change, we make
this ramp-up quicker in the rare cases when it lasted more than 10 ms,
so that it always ramps up to 100% within one block. This way, we can
remove the mute_factor_array.
This change breaks bit-exactness, but careful listening could not reveal
an audible difference.
This change is a part of a larger refactoring of NetEq's PLC code.
Bug: webrtc:9180
Change-Id: I4c513ce3ed8d66f9beec2abfb1f0c7ffaac7a21e
Reviewed-on: https://webrtc-review.googlesource.com/77180
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23342}diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 239e7eb..e4926b6 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -984,35 +984,35 @@
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC)
TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
- Run(8000, PlatformChecksum("2adede965c6f87de7142c51552111d08",
- "028c0fc414b1c9ab7e582dccdf381e98",
- "36c95170c1393d4b765d1c17b61ef977",
+ Run(8000, PlatformChecksum("7294941b62293e143d6d6c84955923fd",
+ "f26b8c9aa8257c7185925fa5b102f46a",
+ "65e5ef5c3de9c2abf3c8d0989772e9fc",
"4598140b5e4f7ee66c5adad609e65a3e",
- "bac5db6dff44323be401060f1279a532"));
+ "04a1d3e735730b6d7dbd5df10f717d27"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
- Run(16000, PlatformChecksum("c2550a3db7632de409e8db0093df1c12",
- "edd31f4b6665cd5b9041fb93f2316594",
- "22128bca51650cb61c80bed63b595603",
+ Run(16000, PlatformChecksum("de8143dd3cc23241f1e1d5cf14e04b8a",
+ "eada3f321120d8d5afffbe4170a55d2f",
+ "135d8c3c7b92aa13d45cad7c91068e3e",
"f2aad418af974a3b1694d5ae5cc2c3c7",
- "61c3cb9386b9503feebcb829c9be54bd"));
+ "728b69068332efade35b1a9c32029e1b"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
- Run(32000, PlatformChecksum("85e28d7950132d56f90b099c90f82153",
- "7b903f5c89997f271b405e63c245ef45",
- "8b8fc6c6fd1dcdcfb3dd90e1ce597f10",
+ Run(32000, PlatformChecksum("521d336237bdcc9ab44050e9da8917fc",
+ "73d44a7bedb6dfa7c70cf997223d8c10",
+ "f3ee2f14b03fb8e98f526f82583f84d9",
"100869c8dcde51346c2073e52a272d98",
- "fdec5301dc649a47d407382b587e14da"));
+ "5f338b4bc38707d0a14d75a357e1546e"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
- Run(48000, PlatformChecksum("ab611510e8fd6d5210a23cc04d3f0e8e",
- "d8609bc9b495d81f29779344c68bcc47",
- "ec5ebb90cda0ea5bb89e79d698af65de",
+ Run(48000, PlatformChecksum("5955e31373828969de7fb308fb58a84e",
+ "83c0eca235b1a806426ff6ca8655cdf7",
+ "1126a8c03d1ebc6aa7348b9c541e2082",
"bd44bf97e7899186532f91235cef444d",
- "0baae2972cca142027d4af44f95f0bd5"));
+ "9d092dbc96e7ef6870b78c1056e87315"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
@@ -1094,11 +1094,11 @@
rtc::scoped_refptr<rtc::RefCountedObject<ADFactory>> factory(
new rtc::RefCountedObject<ADFactory>);
- Run(48000, PlatformChecksum("ab611510e8fd6d5210a23cc04d3f0e8e",
- "d8609bc9b495d81f29779344c68bcc47",
- "ec5ebb90cda0ea5bb89e79d698af65de",
+ Run(48000, PlatformChecksum("5955e31373828969de7fb308fb58a84e",
+ "83c0eca235b1a806426ff6ca8655cdf7",
+ "1126a8c03d1ebc6aa7348b9c541e2082",
"bd44bf97e7899186532f91235cef444d",
- "0baae2972cca142027d4af44f95f0bd5"),
+ "9d092dbc96e7ef6870b78c1056e87315"),
factory, [](AudioCodingModule* acm) {
acm->RegisterReceiveCodec(0, {"MockPCMu", 8000, 1});
});
diff --git a/modules/audio_coding/neteq/merge.cc b/modules/audio_coding/neteq/merge.cc
index b568ff0..fb0bb0d 100644
--- a/modules/audio_coding/neteq/merge.cc
+++ b/modules/audio_coding/neteq/merge.cc
@@ -44,7 +44,6 @@
Merge::~Merge() = default;
size_t Merge::Process(int16_t* input, size_t input_length,
- int16_t* external_mute_factor_array,
AudioMultiVector* output) {
// TODO(hlundin): Change to an enumerator and skip assert.
assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
@@ -73,20 +72,9 @@
input_length_per_channel, 0, input_channel.get());
expanded_[channel].CopyTo(expanded_length, 0, expanded_channel.get());
- int16_t new_mute_factor = SignalScaling(
- input_channel.get(), input_length_per_channel, expanded_channel.get());
-
- // Adjust muting factor (product of "main" muting factor and expand muting
- // factor).
- int16_t* external_mute_factor = &external_mute_factor_array[channel];
- *external_mute_factor =
- (*external_mute_factor * expand_->MuteFactor(channel)) >> 14;
-
- // Update |external_mute_factor| if it is lower than |new_mute_factor|.
- if (new_mute_factor > *external_mute_factor) {
- *external_mute_factor = std::min(new_mute_factor,
- static_cast<int16_t>(16384));
- }
+ const int16_t new_mute_factor = std::min<int16_t>(
+ 16384, SignalScaling(input_channel.get(), input_length_per_channel,
+ expanded_channel.get()));
if (channel == 0) {
// Downsample, correlate, and find strongest correlation period for the
@@ -110,18 +98,24 @@
expanded_length - best_correlation_index);
interpolation_length = std::min(interpolation_length,
input_length_per_channel);
- if (*external_mute_factor < 16384) {
+
+ RTC_DCHECK_LE(new_mute_factor, 16384);
+ int16_t mute_factor =
+ std::max(expand_->MuteFactor(channel), new_mute_factor);
+ RTC_DCHECK_GE(mute_factor, 0);
+
+ if (mute_factor < 16384) {
// Set a suitable muting slope (Q20). 0.004 for NB, 0.002 for WB,
- // and so on.
- int increment = 4194 / fs_mult_;
- *external_mute_factor =
- static_cast<int16_t>(DspHelper::RampSignal(input_channel.get(),
- interpolation_length,
- *external_mute_factor,
- increment));
+ // and so on, or as fast as it takes to come back to full gain within the
+ // frame length.
+ const int back_to_fullscale_inc = static_cast<int>(
+ ((16384 - mute_factor) << 6) / input_length_per_channel);
+ const int increment = std::max(4194 / fs_mult_, back_to_fullscale_inc);
+ mute_factor = static_cast<int16_t>(DspHelper::RampSignal(
+ input_channel.get(), interpolation_length, mute_factor, increment));
DspHelper::UnmuteSignal(&input_channel[interpolation_length],
input_length_per_channel - interpolation_length,
- external_mute_factor, increment,
+ &mute_factor, increment,
&decoded_output[interpolation_length]);
} else {
// No muting needed.
@@ -134,12 +128,12 @@
// Do overlap and mix linearly.
int16_t increment =
static_cast<int16_t>(16384 / (interpolation_length + 1)); // In Q14.
- int16_t mute_factor = 16384 - increment;
+ int16_t local_mute_factor = 16384 - increment;
memmove(temp_data_.data(), expanded_channel.get(),
sizeof(int16_t) * best_correlation_index);
DspHelper::CrossFade(&expanded_channel[best_correlation_index],
input_channel.get(), interpolation_length,
- &mute_factor, increment, decoded_output);
+ &local_mute_factor, increment, decoded_output);
output_length = best_correlation_index + input_length_per_channel;
if (channel == 0) {
diff --git a/modules/audio_coding/neteq/merge.h b/modules/audio_coding/neteq/merge.h
index e027cd7..6da0b4f 100644
--- a/modules/audio_coding/neteq/merge.h
+++ b/modules/audio_coding/neteq/merge.h
@@ -43,11 +43,8 @@
// |input|, having |input_length| samples in total for all channels
// (interleaved). The result is written to |output|. The number of channels
// allocated in |output| defines the number of channels that will be used when
- // de-interleaving |input|. The values in |external_mute_factor_array| (Q14)
- // will be used to scale the audio, and is updated in the process. The array
- // must have |num_channels_| elements.
+ // de-interleaving |input|.
virtual size_t Process(int16_t* input, size_t input_length,
- int16_t* external_mute_factor_array,
AudioMultiVector* output);
virtual size_t RequiredFutureSamples();
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 80bfdaf..688162b 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -1536,9 +1536,8 @@
void NetEqImpl::DoNormal(const int16_t* decoded_buffer, size_t decoded_length,
AudioDecoder::SpeechType speech_type, bool play_dtmf) {
assert(normal_.get());
- assert(mute_factor_array_.get());
normal_->Process(decoded_buffer, decoded_length, last_mode_,
- mute_factor_array_.get(), algorithm_buffer_.get());
+ algorithm_buffer_.get());
if (decoded_length != 0) {
last_mode_ = kModeNormal;
}
@@ -1558,10 +1557,8 @@
void NetEqImpl::DoMerge(int16_t* decoded_buffer, size_t decoded_length,
AudioDecoder::SpeechType speech_type, bool play_dtmf) {
- assert(mute_factor_array_.get());
assert(merge_.get());
size_t new_length = merge_->Process(decoded_buffer, decoded_length,
- mute_factor_array_.get(),
algorithm_buffer_.get());
// Correction can be negative.
int expand_length_correction =
@@ -1803,9 +1800,8 @@
void NetEqImpl::DoCodecInternalCng(const int16_t* decoded_buffer,
size_t decoded_length) {
RTC_DCHECK(normal_.get());
- RTC_DCHECK(mute_factor_array_.get());
normal_->Process(decoded_buffer, decoded_length, last_mode_,
- mute_factor_array_.get(), algorithm_buffer_.get());
+ algorithm_buffer_.get());
last_mode_ = kModeCodecInternalCng;
expand_->Reset();
}
@@ -2065,12 +2061,6 @@
last_mode_ = kModeNormal;
- // Create a new array of mute factors and set all to 1.
- mute_factor_array_.reset(new int16_t[channels]);
- for (size_t i = 0; i < channels; ++i) {
- mute_factor_array_[i] = 16384; // 1.0 in Q14.
- }
-
ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
if (cng_decoder)
cng_decoder->Reset();
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index ce75ce0..17589aa 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -418,7 +418,6 @@
size_t decoder_frame_length_ RTC_GUARDED_BY(crit_sect_);
Modes last_mode_ RTC_GUARDED_BY(crit_sect_);
Operations last_operation_ RTC_GUARDED_BY(crit_sect_);
- std::unique_ptr<int16_t[]> mute_factor_array_ RTC_GUARDED_BY(crit_sect_);
size_t decoded_buffer_length_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(crit_sect_);
uint32_t playout_timestamp_ RTC_GUARDED_BY(crit_sect_);
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 42dbcf3..dff4568 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -463,18 +463,18 @@
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
const std::string output_checksum = PlatformChecksum(
- "09fa7646e2ad032a0b156177b95f09012430f81f",
- "1c64eb8b55ce8878676c6a1e6ddd78f48de0668b",
+ "0c6dc227f781c81a229970f8fceda1a012498cba",
+ "15c4a2202877a414515e218bdb7992f0ad53e5af",
"not used",
- "09fa7646e2ad032a0b156177b95f09012430f81f",
- "759fef89a5de52bd17e733dc255c671ce86be909");
+ "0c6dc227f781c81a229970f8fceda1a012498cba",
+ "25fc4c863caa499aa447a5b8d059f5452cbcc500");
const std::string network_stats_checksum =
- PlatformChecksum("5b4262ca328e5f066af5d34f3380521583dd20de",
- "80235b6d727281203acb63b98f9a9e85d95f7ec0",
+ PlatformChecksum("4b2370f5c794741d2a46be5c7935c66ef3fb53e9",
+ "e339cb2adf5ab3dfc21cb7205d670a34751e8336",
"not used",
- "5b4262ca328e5f066af5d34f3380521583dd20de",
- "5b4262ca328e5f066af5d34f3380521583dd20de");
+ "4b2370f5c794741d2a46be5c7935c66ef3fb53e9",
+ "4b2370f5c794741d2a46be5c7935c66ef3fb53e9");
const std::string rtcp_stats_checksum = PlatformChecksum(
"b8880bf9fed2487efbddcb8d94b9937a29ae521d",
@@ -502,18 +502,18 @@
webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
const std::string output_checksum = PlatformChecksum(
- "e0e4063d55941792a725fde1f781ebbbe83b8165",
- "b77d2f25db1e59e054160a89e7defc7bfdb12f31",
- "a8a525c476a922456a35b5b13861b55cd8c9c7b8",
- "e0e4063d55941792a725fde1f781ebbbe83b8165",
- "e0e4063d55941792a725fde1f781ebbbe83b8165");
+ "14a63b3c7b925c82296be4bafc71bec85f2915c2",
+ "b7b7ed802b0e18ee416973bf3b9ae98599b0181d",
+ "5876e52dda90d5ca433c3726555b907b97c86374",
+ "14a63b3c7b925c82296be4bafc71bec85f2915c2",
+ "14a63b3c7b925c82296be4bafc71bec85f2915c2");
const std::string network_stats_checksum =
- PlatformChecksum("9e72233c78baf685e500dd6c94212b30a4c5f27d",
- "c237d7ca04cbb1ea2e3b27a7c8963015deb985e7",
- "4f1e9734bc80a290faaf9d611efcb8d7802dbc4f",
- "9e72233c78baf685e500dd6c94212b30a4c5f27d",
- "9e72233c78baf685e500dd6c94212b30a4c5f27d");
+ PlatformChecksum("adb3272498e436d1c019cbfd71610e9510c54497",
+ "fa935a91abc7291db47428a2d7c5361b98713a92",
+ "42106aa5267300f709f63737707ef07afd9dac61",
+ "adb3272498e436d1c019cbfd71610e9510c54497",
+ "adb3272498e436d1c019cbfd71610e9510c54497");
const std::string rtcp_stats_checksum = PlatformChecksum(
"e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc
index 48d723a..f10158c 100644
--- a/modules/audio_coding/neteq/normal.cc
+++ b/modules/audio_coding/neteq/normal.cc
@@ -27,7 +27,6 @@
int Normal::Process(const int16_t* input,
size_t length,
Modes last_mode,
- int16_t* external_mute_factor_array,
AudioMultiVector* output) {
if (length == 0) {
// Nothing to process.
@@ -66,10 +65,8 @@
size_t length_per_channel = length / output->Channels();
std::unique_ptr<int16_t[]> signal(new int16_t[length_per_channel]);
for (size_t channel_ix = 0; channel_ix < output->Channels(); ++channel_ix) {
- // Adjust muting factor (main muting factor times expand muting factor).
- external_mute_factor_array[channel_ix] = static_cast<int16_t>(
- (external_mute_factor_array[channel_ix] *
- expand_->MuteFactor(channel_ix)) >> 14);
+ // Set muting factor to the same as expand muting factor.
+ int16_t mute_factor = expand_->MuteFactor(channel_ix);
(*output)[channel_ix].CopyTo(length_per_channel, 0, signal.get());
@@ -92,7 +89,7 @@
energy = 0;
}
- int mute_factor;
+ int local_mute_factor = 16384; // 1.0 in Q14.
if ((energy != 0) &&
(energy > background_noise_.Energy(channel_ix))) {
// Normalize new frame energy to 15 bits.
@@ -103,29 +100,30 @@
int16_t energy_scaled =
static_cast<int16_t>(WEBRTC_SPL_SHIFT_W32(energy, scaling));
int32_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
- mute_factor = WebRtcSpl_SqrtFloor(ratio << 14);
- } else {
- mute_factor = 16384; // 1.0 in Q14.
+ local_mute_factor =
+ std::min(local_mute_factor, WebRtcSpl_SqrtFloor(ratio << 14));
}
- if (mute_factor > external_mute_factor_array[channel_ix]) {
- external_mute_factor_array[channel_ix] =
- static_cast<int16_t>(std::min(mute_factor, 16384));
- }
+ mute_factor = std::max<int16_t>(mute_factor, local_mute_factor);
+ RTC_DCHECK_LE(mute_factor, 16384);
+ RTC_DCHECK_GE(mute_factor, 0);
- // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
- int increment = 64 / fs_mult;
+ // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14),
+ // or as fast as it takes to come back to full gain within the frame
+ // length.
+ const int back_to_fullscale_inc =
+ static_cast<int>((16384 - mute_factor) / length_per_channel);
+ const int increment = std::max(64 / fs_mult, back_to_fullscale_inc);
for (size_t i = 0; i < length_per_channel; i++) {
// Scale with mute factor.
RTC_DCHECK_LT(channel_ix, output->Channels());
RTC_DCHECK_LT(i, output->Size());
- int32_t scaled_signal = (*output)[channel_ix][i] *
- external_mute_factor_array[channel_ix];
+ int32_t scaled_signal = (*output)[channel_ix][i] * mute_factor;
// Shift 14 with proper rounding.
(*output)[channel_ix][i] =
static_cast<int16_t>((scaled_signal + 8192) >> 14);
// Increase mute_factor towards 16384.
- external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
- external_mute_factor_array[channel_ix] + increment, 16384));
+ mute_factor =
+ static_cast<int16_t>(std::min(mute_factor + increment, 16384));
}
// Interpolate the expanded data into the new vector.
@@ -153,8 +151,6 @@
static const size_t kCngLength = 48;
RTC_DCHECK_LE(8 * fs_mult, kCngLength);
int16_t cng_output[kCngLength];
- // Reset mute factor and start up fresh.
- external_mute_factor_array[0] = 16384;
ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
if (cng_decoder) {
@@ -186,28 +182,6 @@
}
RTC_DCHECK_GT(win_up_Q14,
(1 << 14) - 32); // Worst case rouding is a length of 34
- } else if (external_mute_factor_array[0] < 16384) {
- // Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
- // still ramping up from previous muting.
- // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
- int increment = 64 / fs_mult;
- size_t length_per_channel = length / output->Channels();
- for (size_t i = 0; i < length_per_channel; i++) {
- for (size_t channel_ix = 0; channel_ix < output->Channels();
- ++channel_ix) {
- // Scale with mute factor.
- RTC_DCHECK_LT(channel_ix, output->Channels());
- RTC_DCHECK_LT(i, output->Size());
- int32_t scaled_signal = (*output)[channel_ix][i] *
- external_mute_factor_array[channel_ix];
- // Shift 14 with proper rounding.
- (*output)[channel_ix][i] =
- static_cast<int16_t>((scaled_signal + 8192) >> 14);
- // Increase mute_factor towards 16384.
- external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
- 16384, external_mute_factor_array[channel_ix] + increment));
- }
- }
}
return static_cast<int>(length);
diff --git a/modules/audio_coding/neteq/normal.h b/modules/audio_coding/neteq/normal.h
index ab02217..14323ea 100644
--- a/modules/audio_coding/neteq/normal.h
+++ b/modules/audio_coding/neteq/normal.h
@@ -53,11 +53,9 @@
// result is written to |output|. The number of channels allocated in
// |output| defines the number of channels that will be used when
// de-interleaving |input|. |last_mode| contains the mode used in the previous
- // GetAudio call (i.e., not the current one), and |external_mute_factor| is
- // a pointer to the mute factor in the NetEqImpl class.
+ // GetAudio call (i.e., not the current one).
int Process(const int16_t* input, size_t length,
Modes last_mode,
- int16_t* external_mute_factor_array,
AudioMultiVector* output);
private:
diff --git a/modules/audio_coding/neteq/normal_unittest.cc b/modules/audio_coding/neteq/normal_unittest.cc
index b0655d9..ab99d9a 100644
--- a/modules/audio_coding/neteq/normal_unittest.cc
+++ b/modules/audio_coding/neteq/normal_unittest.cc
@@ -68,16 +68,10 @@
Normal normal(fs, &db, bgn, &expand);
int16_t input[1000] = {0};
- std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
- for (size_t i = 0; i < channels; ++i) {
- mute_factor_array[i] = 16384;
- }
AudioMultiVector output(channels);
// Zero input length.
- EXPECT_EQ(
- 0,
- normal.Process(input, 0, kModeExpand, mute_factor_array.get(), &output));
+ EXPECT_EQ(0, normal.Process(input, 0, kModeExpand, &output));
EXPECT_EQ(0u, output.Size());
// Try to make energy_length >> scaling = 0;
@@ -93,7 +87,6 @@
normal.Process(input,
input_size_samples,
kModeExpand,
- mute_factor_array.get(),
&output));
EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
@@ -114,18 +107,11 @@
Normal normal(fs, &db, bgn, &expand);
int16_t input[1000] = {0};
- std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
- for (size_t i = 0; i < channels; ++i) {
- mute_factor_array[i] = 16384;
- }
AudioMultiVector output(channels);
// Let the number of samples be one sample less than 80 samples per channel.
size_t input_len = 80 * channels - 1;
- EXPECT_EQ(
- 0,
- normal.Process(
- input, input_len, kModeExpand, mute_factor_array.get(), &output));
+ EXPECT_EQ(0, normal.Process(input, input_len, kModeExpand, &output));
EXPECT_EQ(0u, output.Size());
EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
@@ -147,12 +133,6 @@
Normal normal(kFs, &db, bgn, &expand);
int16_t input[kPacketsizeBytes] = {0};
-
- std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[kChannels]);
- for (size_t i = 0; i < kChannels; ++i) {
- mute_factor_array[i] = 16384;
- }
-
AudioMultiVector output(kChannels);
EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
@@ -162,7 +142,6 @@
normal.Process(input,
kPacketsizeBytes,
kModeExpand,
- mute_factor_array.get(),
&output));
EXPECT_EQ(kPacketsizeBytes, output.Size());
