NetEq4: Make the algorithm buffer a member variable
This reduces the alloc count by more than 100,000 for
NetEqDecodingTest.TestBitExactness.
BUG=1363
TEST=out/Release/modules_unittests --gtest_filter=NetEqDecodingTest.TestBitExactness
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4651 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
index 6e495d2..1124f8b 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
@@ -70,6 +70,7 @@
payload_splitter_(payload_splitter),
timestamp_scaler_(timestamp_scaler),
vad_(new PostDecodeVad()),
+ algorithm_buffer_(NULL),
sync_buffer_(NULL),
expand_(NULL),
comfort_noise_(NULL),
@@ -112,6 +113,7 @@
NetEqImpl::~NetEqImpl() {
LOG(LS_INFO) << "Deleting NetEqImpl object.";
+ delete algorithm_buffer_;
delete sync_buffer_;
delete background_noise_;
delete expand_;
@@ -620,58 +622,55 @@
vad_->Update(decoded_buffer_.get(), length, speech_type,
sid_frame_available, fs_hz_);
- AudioMultiVector<int16_t> algorithm_buffer(sync_buffer_->Channels());
+ algorithm_buffer_->Clear();
switch (operation) {
case kNormal: {
- DoNormal(decoded_buffer_.get(), length, speech_type, play_dtmf,
- &algorithm_buffer);
+ DoNormal(decoded_buffer_.get(), length, speech_type, play_dtmf);
break;
}
case kMerge: {
- DoMerge(decoded_buffer_.get(), length, speech_type, play_dtmf,
- &algorithm_buffer);
+ DoMerge(decoded_buffer_.get(), length, speech_type, play_dtmf);
break;
}
case kExpand: {
- return_value = DoExpand(play_dtmf, &algorithm_buffer);
+ return_value = DoExpand(play_dtmf);
break;
}
case kAccelerate: {
return_value = DoAccelerate(decoded_buffer_.get(), length, speech_type,
- play_dtmf, &algorithm_buffer);
+ play_dtmf);
break;
}
case kPreemptiveExpand: {
return_value = DoPreemptiveExpand(decoded_buffer_.get(), length,
- speech_type, play_dtmf,
- &algorithm_buffer);
+ speech_type, play_dtmf);
break;
}
case kRfc3389Cng:
case kRfc3389CngNoPacket: {
- return_value = DoRfc3389Cng(&packet_list, play_dtmf, &algorithm_buffer);
+ return_value = DoRfc3389Cng(&packet_list, play_dtmf);
break;
}
case kCodecInternalCng: {
// This handles the case when there is no transmission and the decoder
// should produce internal comfort noise.
// TODO(hlundin): Write test for codec-internal CNG.
- DoCodecInternalCng(&algorithm_buffer);
+ DoCodecInternalCng();
break;
}
case kDtmf: {
// TODO(hlundin): Write test for this.
- return_value = DoDtmf(dtmf_event, &play_dtmf, &algorithm_buffer);
+ return_value = DoDtmf(dtmf_event, &play_dtmf);
break;
}
case kAlternativePlc: {
// TODO(hlundin): Write test for this.
- DoAlternativePlc(false, &algorithm_buffer);
+ DoAlternativePlc(false);
break;
}
case kAlternativePlcIncreaseTimestamp: {
// TODO(hlundin): Write test for this.
- DoAlternativePlc(true, &algorithm_buffer);
+ DoAlternativePlc(true);
break;
}
case kAudioRepetitionIncreaseTimestamp: {
@@ -684,7 +683,7 @@
// TODO(hlundin): Write test for this.
// Copy last |output_size_samples_| from |sync_buffer_| to
// |algorithm_buffer|.
- algorithm_buffer.PushBackFromIndex(
+ algorithm_buffer_->PushBackFromIndex(
*sync_buffer_, sync_buffer_->Size() - output_size_samples_);
expand_->Reset();
break;
@@ -705,7 +704,7 @@
}
// Copy from |algorithm_buffer| to |sync_buffer_|.
- sync_buffer_->PushBack(algorithm_buffer);
+ sync_buffer_->PushBack(*algorithm_buffer_);
// Extract data from |sync_buffer_| to |output|.
int num_output_samples_per_channel = output_size_samples_;
@@ -720,7 +719,7 @@
num_output_samples_per_channel, output);
*num_channels = sync_buffer_->Channels();
LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" <<
- " insert " << algorithm_buffer.Size() << " samples, extract " <<
+ " insert " << algorithm_buffer_->Size() << " samples, extract " <<
samples_from_sync << " samples";
if (samples_from_sync != output_size_samples_) {
LOG_F(LS_ERROR) << "samples_from_sync != output_size_samples_";
@@ -1205,15 +1204,14 @@
}
void NetEqImpl::DoNormal(const int16_t* decoded_buffer, size_t decoded_length,
- AudioDecoder::SpeechType speech_type, bool play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer) {
+ AudioDecoder::SpeechType speech_type, bool play_dtmf) {
assert(decoder_database_.get());
assert(background_noise_);
assert(expand_);
Normal normal(fs_hz_, decoder_database_.get(), *background_noise_, expand_);
assert(mute_factor_array_.get());
normal.Process(decoded_buffer, decoded_length, last_mode_,
- mute_factor_array_.get(), algorithm_buffer);
+ mute_factor_array_.get(), algorithm_buffer_);
if (decoded_length != 0) {
last_mode_ = kModeNormal;
}
@@ -1232,12 +1230,11 @@
}
void NetEqImpl::DoMerge(int16_t* decoded_buffer, size_t decoded_length,
- AudioDecoder::SpeechType speech_type, bool play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer) {
- Merge merge(fs_hz_, algorithm_buffer->Channels(), expand_, sync_buffer_);
+ AudioDecoder::SpeechType speech_type, bool play_dtmf) {
+ Merge merge(fs_hz_, algorithm_buffer_->Channels(), expand_, sync_buffer_);
assert(mute_factor_array_.get());
int new_length = merge.Process(decoded_buffer, decoded_length,
- mute_factor_array_.get(), algorithm_buffer);
+ mute_factor_array_.get(), algorithm_buffer_);
// Update in-call and post-call statistics.
if (expand_->MuteFactor(0) == 0) {
@@ -1259,13 +1256,12 @@
}
}
-int NetEqImpl::DoExpand(bool play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer) {
+int NetEqImpl::DoExpand(bool play_dtmf) {
while ((sync_buffer_->FutureLength() - expand_->overlap_length()) <
static_cast<size_t>(output_size_samples_)) {
- algorithm_buffer->Clear();
- int return_value = expand_->Process(algorithm_buffer);
- int length = algorithm_buffer->Size();
+ algorithm_buffer_->Clear();
+ int return_value = expand_->Process(algorithm_buffer_);
+ int length = algorithm_buffer_->Size();
// Update in-call and post-call statistics.
if (expand_->MuteFactor(0) == 0) {
@@ -1282,8 +1278,8 @@
return return_value;
}
- sync_buffer_->PushBack(*algorithm_buffer);
- algorithm_buffer->Clear();
+ sync_buffer_->PushBack(*algorithm_buffer_);
+ algorithm_buffer_->Clear();
}
if (!play_dtmf) {
dtmf_tone_generator_->Reset();
@@ -1293,11 +1289,10 @@
int NetEqImpl::DoAccelerate(int16_t* decoded_buffer, size_t decoded_length,
AudioDecoder::SpeechType speech_type,
- bool play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer) {
+ bool play_dtmf) {
const size_t required_samples = 240 * fs_mult_; // Must have 30 ms.
int borrowed_samples_per_channel = 0;
- size_t num_channels = algorithm_buffer->Channels();
+ size_t num_channels = algorithm_buffer_->Channels();
size_t decoded_length_per_channel = decoded_length / num_channels;
if (decoded_length_per_channel < required_samples) {
// Must move data from the |sync_buffer_| in order to get 30 ms.
@@ -1315,7 +1310,7 @@
Accelerate accelerate(fs_hz_, num_channels, *background_noise_);
Accelerate::ReturnCodes return_code = accelerate.Process(decoded_buffer,
decoded_length,
- algorithm_buffer,
+ algorithm_buffer_,
&samples_removed);
stats_.AcceleratedSamples(samples_removed);
switch (return_code) {
@@ -1336,22 +1331,22 @@
if (borrowed_samples_per_channel > 0) {
// Copy borrowed samples back to the |sync_buffer_|.
- int length = algorithm_buffer->Size();
+ int length = algorithm_buffer_->Size();
if (length < borrowed_samples_per_channel) {
// This destroys the beginning of the buffer, but will not cause any
// problems.
- sync_buffer_->ReplaceAtIndex(*algorithm_buffer,
+ sync_buffer_->ReplaceAtIndex(*algorithm_buffer_,
sync_buffer_->Size() -
borrowed_samples_per_channel);
sync_buffer_->PushFrontZeros(borrowed_samples_per_channel - length);
- algorithm_buffer->PopFront(length);
- assert(algorithm_buffer->Empty());
+ algorithm_buffer_->PopFront(length);
+ assert(algorithm_buffer_->Empty());
} else {
- sync_buffer_->ReplaceAtIndex(*algorithm_buffer,
+ sync_buffer_->ReplaceAtIndex(*algorithm_buffer_,
borrowed_samples_per_channel,
sync_buffer_->Size() -
borrowed_samples_per_channel);
- algorithm_buffer->PopFront(borrowed_samples_per_channel);
+ algorithm_buffer_->PopFront(borrowed_samples_per_channel);
}
}
@@ -1369,10 +1364,9 @@
int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
size_t decoded_length,
AudioDecoder::SpeechType speech_type,
- bool play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer) {
+ bool play_dtmf) {
const size_t required_samples = 240 * fs_mult_; // Must have 30 ms.
- size_t num_channels = algorithm_buffer->Channels();
+ size_t num_channels = algorithm_buffer_->Channels();
int borrowed_samples_per_channel = 0;
int old_borrowed_samples_per_channel = 0;
size_t decoded_length_per_channel = decoded_length / num_channels;
@@ -1397,7 +1391,7 @@
PreemptiveExpand preemptive_expand(fs_hz_, num_channels, *background_noise_);
PreemptiveExpand::ReturnCodes return_code = preemptive_expand.Process(
decoded_buffer, decoded_length, old_borrowed_samples_per_channel,
- algorithm_buffer, &samples_added);
+ algorithm_buffer_, &samples_added);
stats_.PreemptiveExpandedSamples(samples_added);
switch (return_code) {
case PreemptiveExpand::kSuccess:
@@ -1418,9 +1412,9 @@
if (borrowed_samples_per_channel > 0) {
// Copy borrowed samples back to the |sync_buffer_|.
sync_buffer_->ReplaceAtIndex(
- *algorithm_buffer, borrowed_samples_per_channel,
+ *algorithm_buffer_, borrowed_samples_per_channel,
sync_buffer_->Size() - borrowed_samples_per_channel);
- algorithm_buffer->PopFront(borrowed_samples_per_channel);
+ algorithm_buffer_->PopFront(borrowed_samples_per_channel);
}
// If last packet was decoded as an inband CNG, set mode to CNG instead.
@@ -1434,8 +1428,7 @@
return 0;
}
-int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer) {
+int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) {
if (!packet_list->empty()) {
// Must have exactly one SID frame at this point.
assert(packet_list->size() == 1);
@@ -1470,12 +1463,12 @@
if (comfort_noise_->UpdateParameters(packet) ==
ComfortNoise::kInternalError) {
LOG_FERR0(LS_WARNING, UpdateParameters);
- algorithm_buffer->Zeros(output_size_samples_);
+ algorithm_buffer_->Zeros(output_size_samples_);
return -comfort_noise_->internal_error_code();
}
}
int cn_return = comfort_noise_->Generate(output_size_samples_,
- algorithm_buffer);
+ algorithm_buffer_);
expand_->Reset();
last_mode_ = kModeRfc3389Cng;
if (!play_dtmf) {
@@ -1492,8 +1485,7 @@
return 0;
}
-void NetEqImpl::DoCodecInternalCng(
- AudioMultiVector<int16_t>* algorithm_buffer) {
+void NetEqImpl::DoCodecInternalCng() {
int length = 0;
// TODO(hlundin): Will probably need a longer buffer for multi-channel.
int16_t decoded_buffer[kMaxFrameSize];
@@ -1506,13 +1498,12 @@
Normal normal(fs_hz_, decoder_database_.get(), *background_noise_, expand_);
assert(mute_factor_array_.get());
normal.Process(decoded_buffer, length, last_mode_, mute_factor_array_.get(),
- algorithm_buffer);
+ algorithm_buffer_);
last_mode_ = kModeCodecInternalCng;
expand_->Reset();
}
-int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer) {
+int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) {
// This block of the code and the block further down, handling |dtmf_switch|
// are commented out. Otherwise playing out-of-band DTMF would fail in VoE
// test, DtmfTest.ManualSuccessfullySendsOutOfBandTelephoneEvents. This is
@@ -1540,11 +1531,11 @@
if (dtmf_return_value == 0) {
// Generate DTMF signal.
dtmf_return_value = dtmf_tone_generator_->Generate(output_size_samples_,
- algorithm_buffer);
+ algorithm_buffer_);
}
if (dtmf_return_value < 0) {
- algorithm_buffer->Zeros(output_size_samples_);
+ algorithm_buffer_->Zeros(output_size_samples_);
return dtmf_return_value;
}
@@ -1562,10 +1553,10 @@
// int required_length = sync_buffer_->FutureLength();
// assert(dtmf_tone_generator_->initialized());
// dtmf_return_value = dtmf_tone_generator_->Generate(required_length,
- // algorithm_buffer);
- // assert((size_t) required_length == algorithm_buffer->Size());
+ // algorithm_buffer_);
+ // assert((size_t) required_length == algorithm_buffer_->Size());
// if (dtmf_return_value < 0) {
- // algorithm_buffer->Zeros(output_size_samples_);
+ // algorithm_buffer_->Zeros(output_size_samples_);
// return dtmf_return_value;
// }
//
@@ -1573,13 +1564,13 @@
// // data.
// // TODO(hlundin): It seems that this overwriting has gone lost.
// // Not adapted for multi-channel yet.
- // assert(algorithm_buffer->Channels() == 1);
- // if (algorithm_buffer->Channels() != 1) {
+ // assert(algorithm_buffer_->Channels() == 1);
+ // if (algorithm_buffer_->Channels() != 1) {
// LOG(LS_WARNING) << "DTMF not supported for more than one channel";
// return kStereoNotSupported;
// }
// // Shuffle the remaining data to the beginning of algorithm buffer.
- // algorithm_buffer->PopFront(sync_buffer_->FutureLength());
+ // algorithm_buffer_->PopFront(sync_buffer_->FutureLength());
// }
sync_buffer_->IncreaseEndTimestamp(output_size_samples_);
@@ -1591,8 +1582,7 @@
return 0;
}
-void NetEqImpl::DoAlternativePlc(bool increase_timestamp,
- AudioMultiVector<int16_t>* algorithm_buffer) {
+void NetEqImpl::DoAlternativePlc(bool increase_timestamp) {
AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
int length;
if (decoder && decoder->HasDecodePlc()) {
@@ -1601,14 +1591,14 @@
int16_t decoded_buffer[kMaxFrameSize];
length = decoder->DecodePlc(1, decoded_buffer);
if (length > 0) {
- algorithm_buffer->PushBackInterleaved(decoded_buffer, length);
+ algorithm_buffer_->PushBackInterleaved(decoded_buffer, length);
} else {
length = 0;
}
} else {
// Do simple zero-stuffing.
length = output_size_samples_;
- algorithm_buffer->Zeros(length);
+ algorithm_buffer_->Zeros(length);
// By not advancing the timestamp, NetEq inserts samples.
stats_.AddZeros(length);
}
@@ -1756,6 +1746,12 @@
assert(vad_.get()); // Cannot be NULL here.
vad_->Init();
+ // Delete algorithm buffer and create a new one.
+ if (algorithm_buffer_) {
+ delete algorithm_buffer_;
+ }
+ algorithm_buffer_ = new AudioMultiVector<int16_t>(channels);
+
// Delete sync buffer and create a new one.
if (sync_buffer_) {
delete sync_buffer_;
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.h b/webrtc/modules/audio_coding/neteq4/neteq_impl.h
index 56dde77..b000961 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.h
@@ -222,49 +222,42 @@
// Sub-method which calls the Normal class to perform the normal operation.
void DoNormal(const int16_t* decoded_buffer, size_t decoded_length,
- AudioDecoder::SpeechType speech_type, bool play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer);
+ AudioDecoder::SpeechType speech_type, bool play_dtmf);
// Sub-method which calls the Merge class to perform the merge operation.
void DoMerge(int16_t* decoded_buffer, size_t decoded_length,
- AudioDecoder::SpeechType speech_type, bool play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer);
+ AudioDecoder::SpeechType speech_type, bool play_dtmf);
// Sub-method which calls the Expand class to perform the expand operation.
- int DoExpand(bool play_dtmf, AudioMultiVector<int16_t>* algorithm_buffer);
+ int DoExpand(bool play_dtmf);
// Sub-method which calls the Accelerate class to perform the accelerate
// operation.
int DoAccelerate(int16_t* decoded_buffer, size_t decoded_length,
- AudioDecoder::SpeechType speech_type, bool play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer);
+ AudioDecoder::SpeechType speech_type, bool play_dtmf);
// Sub-method which calls the PreemptiveExpand class to perform the
// preemtive expand operation.
int DoPreemptiveExpand(int16_t* decoded_buffer, size_t decoded_length,
- AudioDecoder::SpeechType speech_type, bool play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer);
+ AudioDecoder::SpeechType speech_type, bool play_dtmf);
// Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
// noise. |packet_list| can either contain one SID frame to update the
// noise parameters, or no payload at all, in which case the previously
// received parameters are used.
- int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer);
+ int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf);
// Calls the audio decoder to generate codec-internal comfort noise when
// no packet was received.
- void DoCodecInternalCng(AudioMultiVector<int16_t>* algorithm_buffer);
+ void DoCodecInternalCng();
// Calls the DtmfToneGenerator class to generate DTMF tones.
- int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf,
- AudioMultiVector<int16_t>* algorithm_buffer);
+ int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf);
// Produces packet-loss concealment using alternative methods. If the codec
// has an internal PLC, it is called to generate samples. Otherwise, the
// method performs zero-stuffing.
- void DoAlternativePlc(bool increase_timestamp,
- AudioMultiVector<int16_t>* algorithm_buffer);
+ void DoAlternativePlc(bool increase_timestamp);
// Overdub DTMF on top of |output|.
int DtmfOverdub(const DtmfEvent& dtmf_event, size_t num_channels,
@@ -296,6 +289,7 @@
scoped_ptr<TimestampScaler> timestamp_scaler_;
scoped_ptr<DecisionLogic> decision_logic_;
scoped_ptr<PostDecodeVad> vad_;
+ AudioMultiVector<int16_t>* algorithm_buffer_;
SyncBuffer* sync_buffer_;
Expand* expand_;
RandomVector random_vector_;