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
This commit is contained in:
henrik.lundin@webrtc.org
parent
cadf9040cb
commit
c487c6abb0
@ -70,6 +70,7 @@ NetEqImpl::NetEqImpl(int fs,
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payload_splitter_(payload_splitter),
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timestamp_scaler_(timestamp_scaler),
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vad_(new PostDecodeVad()),
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algorithm_buffer_(NULL),
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sync_buffer_(NULL),
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expand_(NULL),
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comfort_noise_(NULL),
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@ -112,6 +113,7 @@ NetEqImpl::NetEqImpl(int fs,
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NetEqImpl::~NetEqImpl() {
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LOG(LS_INFO) << "Deleting NetEqImpl object.";
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delete algorithm_buffer_;
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delete sync_buffer_;
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delete background_noise_;
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delete expand_;
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@ -620,58 +622,55 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
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vad_->Update(decoded_buffer_.get(), length, speech_type,
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sid_frame_available, fs_hz_);
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AudioMultiVector<int16_t> algorithm_buffer(sync_buffer_->Channels());
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algorithm_buffer_->Clear();
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switch (operation) {
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case kNormal: {
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DoNormal(decoded_buffer_.get(), length, speech_type, play_dtmf,
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&algorithm_buffer);
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DoNormal(decoded_buffer_.get(), length, speech_type, play_dtmf);
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break;
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}
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case kMerge: {
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DoMerge(decoded_buffer_.get(), length, speech_type, play_dtmf,
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&algorithm_buffer);
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DoMerge(decoded_buffer_.get(), length, speech_type, play_dtmf);
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break;
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}
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case kExpand: {
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return_value = DoExpand(play_dtmf, &algorithm_buffer);
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return_value = DoExpand(play_dtmf);
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break;
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}
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case kAccelerate: {
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return_value = DoAccelerate(decoded_buffer_.get(), length, speech_type,
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play_dtmf, &algorithm_buffer);
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play_dtmf);
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break;
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}
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case kPreemptiveExpand: {
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return_value = DoPreemptiveExpand(decoded_buffer_.get(), length,
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speech_type, play_dtmf,
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&algorithm_buffer);
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speech_type, play_dtmf);
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break;
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}
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case kRfc3389Cng:
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case kRfc3389CngNoPacket: {
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return_value = DoRfc3389Cng(&packet_list, play_dtmf, &algorithm_buffer);
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return_value = DoRfc3389Cng(&packet_list, play_dtmf);
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break;
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}
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case kCodecInternalCng: {
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// This handles the case when there is no transmission and the decoder
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// should produce internal comfort noise.
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// TODO(hlundin): Write test for codec-internal CNG.
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DoCodecInternalCng(&algorithm_buffer);
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DoCodecInternalCng();
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break;
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}
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case kDtmf: {
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// TODO(hlundin): Write test for this.
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return_value = DoDtmf(dtmf_event, &play_dtmf, &algorithm_buffer);
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return_value = DoDtmf(dtmf_event, &play_dtmf);
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break;
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}
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case kAlternativePlc: {
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// TODO(hlundin): Write test for this.
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DoAlternativePlc(false, &algorithm_buffer);
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DoAlternativePlc(false);
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break;
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}
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case kAlternativePlcIncreaseTimestamp: {
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// TODO(hlundin): Write test for this.
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DoAlternativePlc(true, &algorithm_buffer);
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DoAlternativePlc(true);
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break;
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}
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case kAudioRepetitionIncreaseTimestamp: {
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@ -684,7 +683,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
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// TODO(hlundin): Write test for this.
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// Copy last |output_size_samples_| from |sync_buffer_| to
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// |algorithm_buffer|.
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algorithm_buffer.PushBackFromIndex(
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algorithm_buffer_->PushBackFromIndex(
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*sync_buffer_, sync_buffer_->Size() - output_size_samples_);
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expand_->Reset();
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break;
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@ -705,7 +704,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
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}
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// Copy from |algorithm_buffer| to |sync_buffer_|.
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sync_buffer_->PushBack(algorithm_buffer);
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sync_buffer_->PushBack(*algorithm_buffer_);
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// Extract data from |sync_buffer_| to |output|.
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int num_output_samples_per_channel = output_size_samples_;
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@ -720,7 +719,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
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num_output_samples_per_channel, output);
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*num_channels = sync_buffer_->Channels();
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LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" <<
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" insert " << algorithm_buffer.Size() << " samples, extract " <<
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" insert " << algorithm_buffer_->Size() << " samples, extract " <<
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samples_from_sync << " samples";
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if (samples_from_sync != output_size_samples_) {
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LOG_F(LS_ERROR) << "samples_from_sync != output_size_samples_";
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@ -1205,15 +1204,14 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
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}
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void NetEqImpl::DoNormal(const int16_t* decoded_buffer, size_t decoded_length,
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AudioDecoder::SpeechType speech_type, bool play_dtmf,
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AudioMultiVector<int16_t>* algorithm_buffer) {
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AudioDecoder::SpeechType speech_type, bool play_dtmf) {
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assert(decoder_database_.get());
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assert(background_noise_);
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assert(expand_);
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Normal normal(fs_hz_, decoder_database_.get(), *background_noise_, expand_);
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assert(mute_factor_array_.get());
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normal.Process(decoded_buffer, decoded_length, last_mode_,
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mute_factor_array_.get(), algorithm_buffer);
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mute_factor_array_.get(), algorithm_buffer_);
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if (decoded_length != 0) {
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last_mode_ = kModeNormal;
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}
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@ -1232,12 +1230,11 @@ void NetEqImpl::DoNormal(const int16_t* decoded_buffer, size_t decoded_length,
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}
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void NetEqImpl::DoMerge(int16_t* decoded_buffer, size_t decoded_length,
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AudioDecoder::SpeechType speech_type, bool play_dtmf,
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AudioMultiVector<int16_t>* algorithm_buffer) {
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Merge merge(fs_hz_, algorithm_buffer->Channels(), expand_, sync_buffer_);
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AudioDecoder::SpeechType speech_type, bool play_dtmf) {
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Merge merge(fs_hz_, algorithm_buffer_->Channels(), expand_, sync_buffer_);
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assert(mute_factor_array_.get());
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int new_length = merge.Process(decoded_buffer, decoded_length,
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mute_factor_array_.get(), algorithm_buffer);
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mute_factor_array_.get(), algorithm_buffer_);
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// Update in-call and post-call statistics.
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if (expand_->MuteFactor(0) == 0) {
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@ -1259,13 +1256,12 @@ void NetEqImpl::DoMerge(int16_t* decoded_buffer, size_t decoded_length,
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}
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}
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int NetEqImpl::DoExpand(bool play_dtmf,
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AudioMultiVector<int16_t>* algorithm_buffer) {
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int NetEqImpl::DoExpand(bool play_dtmf) {
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while ((sync_buffer_->FutureLength() - expand_->overlap_length()) <
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static_cast<size_t>(output_size_samples_)) {
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algorithm_buffer->Clear();
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int return_value = expand_->Process(algorithm_buffer);
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int length = algorithm_buffer->Size();
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algorithm_buffer_->Clear();
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int return_value = expand_->Process(algorithm_buffer_);
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int length = algorithm_buffer_->Size();
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// Update in-call and post-call statistics.
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if (expand_->MuteFactor(0) == 0) {
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@ -1282,8 +1278,8 @@ int NetEqImpl::DoExpand(bool play_dtmf,
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return return_value;
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}
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sync_buffer_->PushBack(*algorithm_buffer);
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algorithm_buffer->Clear();
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sync_buffer_->PushBack(*algorithm_buffer_);
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algorithm_buffer_->Clear();
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}
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if (!play_dtmf) {
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dtmf_tone_generator_->Reset();
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@ -1293,11 +1289,10 @@ int NetEqImpl::DoExpand(bool play_dtmf,
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int NetEqImpl::DoAccelerate(int16_t* decoded_buffer, size_t decoded_length,
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AudioDecoder::SpeechType speech_type,
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bool play_dtmf,
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AudioMultiVector<int16_t>* algorithm_buffer) {
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bool play_dtmf) {
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const size_t required_samples = 240 * fs_mult_; // Must have 30 ms.
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int borrowed_samples_per_channel = 0;
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size_t num_channels = algorithm_buffer->Channels();
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size_t num_channels = algorithm_buffer_->Channels();
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size_t decoded_length_per_channel = decoded_length / num_channels;
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if (decoded_length_per_channel < required_samples) {
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// Must move data from the |sync_buffer_| in order to get 30 ms.
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@ -1315,7 +1310,7 @@ int NetEqImpl::DoAccelerate(int16_t* decoded_buffer, size_t decoded_length,
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Accelerate accelerate(fs_hz_, num_channels, *background_noise_);
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Accelerate::ReturnCodes return_code = accelerate.Process(decoded_buffer,
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decoded_length,
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algorithm_buffer,
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algorithm_buffer_,
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&samples_removed);
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stats_.AcceleratedSamples(samples_removed);
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switch (return_code) {
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@ -1336,22 +1331,22 @@ int NetEqImpl::DoAccelerate(int16_t* decoded_buffer, size_t decoded_length,
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if (borrowed_samples_per_channel > 0) {
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// Copy borrowed samples back to the |sync_buffer_|.
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int length = algorithm_buffer->Size();
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int length = algorithm_buffer_->Size();
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if (length < borrowed_samples_per_channel) {
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// This destroys the beginning of the buffer, but will not cause any
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// problems.
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sync_buffer_->ReplaceAtIndex(*algorithm_buffer,
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sync_buffer_->ReplaceAtIndex(*algorithm_buffer_,
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sync_buffer_->Size() -
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borrowed_samples_per_channel);
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sync_buffer_->PushFrontZeros(borrowed_samples_per_channel - length);
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algorithm_buffer->PopFront(length);
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assert(algorithm_buffer->Empty());
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algorithm_buffer_->PopFront(length);
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assert(algorithm_buffer_->Empty());
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} else {
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sync_buffer_->ReplaceAtIndex(*algorithm_buffer,
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sync_buffer_->ReplaceAtIndex(*algorithm_buffer_,
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borrowed_samples_per_channel,
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sync_buffer_->Size() -
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borrowed_samples_per_channel);
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algorithm_buffer->PopFront(borrowed_samples_per_channel);
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algorithm_buffer_->PopFront(borrowed_samples_per_channel);
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}
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}
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@ -1369,10 +1364,9 @@ int NetEqImpl::DoAccelerate(int16_t* decoded_buffer, size_t decoded_length,
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int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
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size_t decoded_length,
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AudioDecoder::SpeechType speech_type,
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bool play_dtmf,
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AudioMultiVector<int16_t>* algorithm_buffer) {
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bool play_dtmf) {
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const size_t required_samples = 240 * fs_mult_; // Must have 30 ms.
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size_t num_channels = algorithm_buffer->Channels();
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size_t num_channels = algorithm_buffer_->Channels();
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int borrowed_samples_per_channel = 0;
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int old_borrowed_samples_per_channel = 0;
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size_t decoded_length_per_channel = decoded_length / num_channels;
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@ -1397,7 +1391,7 @@ int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
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PreemptiveExpand preemptive_expand(fs_hz_, num_channels, *background_noise_);
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PreemptiveExpand::ReturnCodes return_code = preemptive_expand.Process(
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decoded_buffer, decoded_length, old_borrowed_samples_per_channel,
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algorithm_buffer, &samples_added);
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algorithm_buffer_, &samples_added);
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stats_.PreemptiveExpandedSamples(samples_added);
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switch (return_code) {
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case PreemptiveExpand::kSuccess:
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@ -1418,9 +1412,9 @@ int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
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if (borrowed_samples_per_channel > 0) {
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// Copy borrowed samples back to the |sync_buffer_|.
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sync_buffer_->ReplaceAtIndex(
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*algorithm_buffer, borrowed_samples_per_channel,
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*algorithm_buffer_, borrowed_samples_per_channel,
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sync_buffer_->Size() - borrowed_samples_per_channel);
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algorithm_buffer->PopFront(borrowed_samples_per_channel);
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algorithm_buffer_->PopFront(borrowed_samples_per_channel);
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}
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// If last packet was decoded as an inband CNG, set mode to CNG instead.
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@ -1434,8 +1428,7 @@ int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
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return 0;
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}
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int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf,
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AudioMultiVector<int16_t>* algorithm_buffer) {
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int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) {
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if (!packet_list->empty()) {
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// Must have exactly one SID frame at this point.
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assert(packet_list->size() == 1);
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@ -1470,12 +1463,12 @@ int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf,
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if (comfort_noise_->UpdateParameters(packet) ==
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ComfortNoise::kInternalError) {
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LOG_FERR0(LS_WARNING, UpdateParameters);
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algorithm_buffer->Zeros(output_size_samples_);
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algorithm_buffer_->Zeros(output_size_samples_);
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return -comfort_noise_->internal_error_code();
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}
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}
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int cn_return = comfort_noise_->Generate(output_size_samples_,
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algorithm_buffer);
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algorithm_buffer_);
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expand_->Reset();
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last_mode_ = kModeRfc3389Cng;
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if (!play_dtmf) {
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@ -1492,8 +1485,7 @@ int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf,
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return 0;
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}
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void NetEqImpl::DoCodecInternalCng(
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AudioMultiVector<int16_t>* algorithm_buffer) {
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void NetEqImpl::DoCodecInternalCng() {
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int length = 0;
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// TODO(hlundin): Will probably need a longer buffer for multi-channel.
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int16_t decoded_buffer[kMaxFrameSize];
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@ -1506,13 +1498,12 @@ void NetEqImpl::DoCodecInternalCng(
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Normal normal(fs_hz_, decoder_database_.get(), *background_noise_, expand_);
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assert(mute_factor_array_.get());
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normal.Process(decoded_buffer, length, last_mode_, mute_factor_array_.get(),
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algorithm_buffer);
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algorithm_buffer_);
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last_mode_ = kModeCodecInternalCng;
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expand_->Reset();
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}
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int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf,
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AudioMultiVector<int16_t>* algorithm_buffer) {
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int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) {
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// This block of the code and the block further down, handling |dtmf_switch|
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// are commented out. Otherwise playing out-of-band DTMF would fail in VoE
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// test, DtmfTest.ManualSuccessfullySendsOutOfBandTelephoneEvents. This is
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@ -1540,11 +1531,11 @@ int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf,
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if (dtmf_return_value == 0) {
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// Generate DTMF signal.
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dtmf_return_value = dtmf_tone_generator_->Generate(output_size_samples_,
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algorithm_buffer);
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algorithm_buffer_);
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}
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if (dtmf_return_value < 0) {
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algorithm_buffer->Zeros(output_size_samples_);
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algorithm_buffer_->Zeros(output_size_samples_);
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return dtmf_return_value;
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}
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@ -1562,10 +1553,10 @@ int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf,
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// int required_length = sync_buffer_->FutureLength();
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// assert(dtmf_tone_generator_->initialized());
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// dtmf_return_value = dtmf_tone_generator_->Generate(required_length,
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// algorithm_buffer);
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// assert((size_t) required_length == algorithm_buffer->Size());
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// algorithm_buffer_);
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// assert((size_t) required_length == algorithm_buffer_->Size());
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// if (dtmf_return_value < 0) {
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// algorithm_buffer->Zeros(output_size_samples_);
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// algorithm_buffer_->Zeros(output_size_samples_);
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// return dtmf_return_value;
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// }
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//
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@ -1573,13 +1564,13 @@ int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf,
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// // data.
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// // TODO(hlundin): It seems that this overwriting has gone lost.
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// // Not adapted for multi-channel yet.
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// assert(algorithm_buffer->Channels() == 1);
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// if (algorithm_buffer->Channels() != 1) {
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// assert(algorithm_buffer_->Channels() == 1);
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// if (algorithm_buffer_->Channels() != 1) {
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// LOG(LS_WARNING) << "DTMF not supported for more than one channel";
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// return kStereoNotSupported;
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// }
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// // Shuffle the remaining data to the beginning of algorithm buffer.
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// algorithm_buffer->PopFront(sync_buffer_->FutureLength());
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// algorithm_buffer_->PopFront(sync_buffer_->FutureLength());
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// }
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sync_buffer_->IncreaseEndTimestamp(output_size_samples_);
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@ -1591,8 +1582,7 @@ int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf,
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return 0;
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}
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void NetEqImpl::DoAlternativePlc(bool increase_timestamp,
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AudioMultiVector<int16_t>* algorithm_buffer) {
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void NetEqImpl::DoAlternativePlc(bool increase_timestamp) {
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AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
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int length;
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if (decoder && decoder->HasDecodePlc()) {
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@ -1601,14 +1591,14 @@ void NetEqImpl::DoAlternativePlc(bool increase_timestamp,
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int16_t decoded_buffer[kMaxFrameSize];
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length = decoder->DecodePlc(1, decoded_buffer);
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if (length > 0) {
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algorithm_buffer->PushBackInterleaved(decoded_buffer, length);
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algorithm_buffer_->PushBackInterleaved(decoded_buffer, length);
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} else {
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length = 0;
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}
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} else {
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// Do simple zero-stuffing.
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length = output_size_samples_;
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algorithm_buffer->Zeros(length);
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algorithm_buffer_->Zeros(length);
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// By not advancing the timestamp, NetEq inserts samples.
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stats_.AddZeros(length);
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}
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@ -1756,6 +1746,12 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
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assert(vad_.get()); // Cannot be NULL here.
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vad_->Init();
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|
||||
// 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_;
|
||||
|
||||
@ -222,49 +222,42 @@ class NetEqImpl : public webrtc::NetEq {
|
||||
|
||||
// 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 @@ class NetEqImpl : public webrtc::NetEq {
|
||||
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_;
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user