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NetEq: Change NetEq's ramp-up behavior after expansions

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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}
This commit is contained in:
Henrik Lundin 2018-05-22 10:40:23 +02:00 committed by Commit Bot
parent 7a84fcf47a
commit 6dc82e8f8b
9 changed files with 81 additions and 150 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

40
modules/audio_coding/acm2/audio_coding_module_unittest.cc
View File

@ -984,35 +984,35 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
#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 @@ TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
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});
});

46
modules/audio_coding/neteq/merge.cc
View File

@ -44,7 +44,6 @@ Merge::Merge(int fs_hz,
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 @@ size_t Merge::Process(int16_t* input, size_t input_length,
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 @@ size_t Merge::Process(int16_t* input, size_t input_length,
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 @@ size_t Merge::Process(int16_t* input, size_t input_length,
// 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) {

5
modules/audio_coding/neteq/merge.h
View File

@ -43,11 +43,8 @@ class Merge {
// |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();

14
modules/audio_coding/neteq/neteq_impl.cc
View File

@ -1536,9 +1536,8 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation,
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::DoNormal(const int16_t* decoded_buffer, size_t decoded_length,
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 @@ int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) {
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 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
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();

1
modules/audio_coding/neteq/neteq_impl.h
View File

@ -418,7 +418,6 @@ class NetEqImpl : public webrtc::NetEq {
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_);

36
modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -463,18 +463,18 @@ TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
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 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
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",

60
modules/audio_coding/neteq/normal.cc
View File

@ -27,7 +27,6 @@ namespace webrtc {
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 @@ int Normal::Process(const int16_t* input,
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 @@ int Normal::Process(const int16_t* input,
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 @@ int Normal::Process(const int16_t* input,
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.
}
if (mute_factor > external_mute_factor_array[channel_ix]) {
external_mute_factor_array[channel_ix] =
static_cast<int16_t>(std::min(mute_factor, 16384));
local_mute_factor =
std::min(local_mute_factor, WebRtcSpl_SqrtFloor(ratio << 14));
}
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 @@ int Normal::Process(const int16_t* input,
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 @@ int Normal::Process(const int16_t* input,
}
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);

4
modules/audio_coding/neteq/normal.h
View File

@ -53,11 +53,9 @@ class Normal {
// 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:

25
modules/audio_coding/neteq/normal_unittest.cc
View File

@ -68,16 +68,10 @@ TEST(Normal, AvoidDivideByZero) {
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 @@ TEST(Normal, AvoidDivideByZero) {
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 @@ TEST(Normal, InputLengthAndChannelsDoNotMatch) {
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 @@ TEST(Normal, LastModeExpand120msPacket) {
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 @@ TEST(Normal, LastModeExpand120msPacket) {
normal.Process(input,
kPacketsizeBytes,
kModeExpand,
mute_factor_array.get(),
&output));
EXPECT_EQ(kPacketsizeBytes, output.Size());