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Disable all LS_VERBOSE logging in NetEq4

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This reduces exectution time of NetEqDecodingTest.TestBitExactness
with almost 30% and reduces the allocation count (from valgrind)
with almost 50% for the same test.

An issue has been created to re-enable logs when logging performance
is improved; see https://code.google.com/p/webrtc/issues/detail?id=2317.

BUG=1363
TEST=out/Release/modules_unittests --gtest_filter=NetEqDecodingTest.TestBitExactness
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2136004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4652 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2013-09-02 09:41:06 +00:00
parent c487c6abb0
commit b3e905cd91
4 changed files with 17 additions and 10 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/audio_coding/neteq4/decision_logic.cc
View File

@ -127,7 +127,7 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
const int cur_size_samples =
samples_left + packet_buffer_.NumSamplesInBuffer(decoder_database_,
decoder_frame_length);
LOG(LS_VERBOSE) << "Buffers: " << packet_buffer_.NumPacketsInBuffer() <<
NETEQ_LOG_VERBOSE << "Buffers: " << packet_buffer_.NumPacketsInBuffer() <<
" packets * " << decoder_frame_length << " samples/packet + " <<
samples_left << " samples in sync buffer = " << cur_size_samples;

6
webrtc/modules/audio_coding/neteq4/defines.h
View File

@ -47,5 +47,11 @@ enum Modes {
kModeUndefined = -1
};
#ifdef NETEQ4_VERBOSE_LOGGING
#define NETEQ_LOG_VERBOSE LOG(LS_VERBOSE)
#else
#define NETEQ_LOG_VERBOSE while(false)LOG(LS_VERBOSE)
#endif
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DEFINES_H_

16
webrtc/modules/audio_coding/neteq4/neteq_impl.cc
View File

@ -126,7 +126,7 @@ int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
int length_bytes,
uint32_t receive_timestamp) {
CriticalSectionScoped lock(crit_sect_);
LOG(LS_VERBOSE) << "InsertPacket: ts=" << rtp_header.header.timestamp <<
NETEQ_LOG_VERBOSE << "InsertPacket: ts=" << rtp_header.header.timestamp <<
", sn=" << rtp_header.header.sequenceNumber <<
", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
", ssrc=" << rtp_header.header.ssrc <<
@ -145,10 +145,10 @@ int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
int* samples_per_channel, int* num_channels,
NetEqOutputType* type) {
CriticalSectionScoped lock(crit_sect_);
LOG(LS_VERBOSE) << "GetAudio";
NETEQ_LOG_VERBOSE << "GetAudio";
int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
num_channels);
LOG(LS_VERBOSE) << "Produced " << *samples_per_channel <<
NETEQ_LOG_VERBOSE << "Produced " << *samples_per_channel <<
" samples/channel for " << *num_channels << " channel(s)";
if (error != 0) {
LOG_FERR1(LS_WARNING, GetAudioInternal, error);
@ -608,7 +608,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
last_mode_ = kModeError;
return return_value;
}
LOG(LS_VERBOSE) << "GetDecision returned operation=" << operation <<
NETEQ_LOG_VERBOSE << "GetDecision returned operation=" << operation <<
" and " << packet_list.size() << " packet(s)";
AudioDecoder::SpeechType speech_type;
@ -718,7 +718,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
int samples_from_sync = sync_buffer_->GetNextAudioInterleaved(
num_output_samples_per_channel, output);
*num_channels = sync_buffer_->Channels();
LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" <<
NETEQ_LOG_VERBOSE << "Sync buffer (" << *num_channels << " channel(s)):" <<
" insert " << algorithm_buffer_->Size() << " samples, extract " <<
samples_from_sync << " samples";
if (samples_from_sync != output_size_samples_) {
@ -1145,7 +1145,7 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
int16_t decode_length;
if (!packet->primary) {
// This is a redundant payload; call the special decoder method.
LOG(LS_VERBOSE) << "Decoding packet (redundant):" <<
NETEQ_LOG_VERBOSE << "Decoding packet (redundant):" <<
" ts=" << packet->header.timestamp <<
", sn=" << packet->header.sequenceNumber <<
", pt=" << static_cast<int>(packet->header.payloadType) <<
@ -1155,7 +1155,7 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
packet->payload, packet->payload_length,
&decoded_buffer_[*decoded_length], speech_type);
} else {
LOG(LS_VERBOSE) << "Decoding packet: ts=" << packet->header.timestamp <<
NETEQ_LOG_VERBOSE << "Decoding packet: ts=" << packet->header.timestamp <<
", sn=" << packet->header.sequenceNumber <<
", pt=" << static_cast<int>(packet->header.payloadType) <<
", ssrc=" << packet->header.ssrc <<
@ -1172,7 +1172,7 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
*decoded_length += decode_length;
// Update |decoder_frame_length_| with number of samples per channel.
decoder_frame_length_ = decode_length / decoder->channels();
LOG(LS_VERBOSE) << "Decoded " << decode_length << " samples (" <<
NETEQ_LOG_VERBOSE << "Decoded " << decode_length << " samples (" <<
decoder->channels() << " channel(s) -> " << decoder_frame_length_ <<
" samples per channel)";
} else if (decode_length < 0) {

3
webrtc/modules/audio_coding/neteq4/timestamp_scaler.cc
View File

@ -11,6 +11,7 @@
#include "webrtc/modules/audio_coding/neteq4/timestamp_scaler.h"
#include "webrtc/modules/audio_coding/neteq4/decoder_database.h"
#include "webrtc/modules/audio_coding/neteq4/defines.h"
#include "webrtc/system_wrappers/interface/logging.h"
namespace webrtc {
@ -84,7 +85,7 @@ uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
assert(denominator_ > 0); // Should not be possible.
external_ref_ = external_timestamp;
internal_ref_ += (external_diff * numerator_) / denominator_;
LOG(LS_VERBOSE) << "Converting timestamp: " << external_timestamp <<
NETEQ_LOG_VERBOSE << "Converting timestamp: " << external_timestamp <<
" -> " << internal_ref_;
return internal_ref_;
} else {