Re-enable verbose logging in NetEq4.
Using neteq4_speed_test there no complexity penalty is observed when verbose
logging is enabled.
BUG=2317
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2293004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4841 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/decision_logic.cc b/webrtc/modules/audio_coding/neteq4/decision_logic.cc
index 58accfd..04b886a 100644
--- a/webrtc/modules/audio_coding/neteq4/decision_logic.cc
+++ b/webrtc/modules/audio_coding/neteq4/decision_logic.cc
@@ -128,7 +128,7 @@
const int cur_size_samples =
samples_left + packet_buffer_.NumSamplesInBuffer(decoder_database_,
decoder_frame_length);
- NETEQ_LOG_VERBOSE << "Buffers: " << packet_buffer_.NumPacketsInBuffer() <<
+ LOG(LS_VERBOSE) << "Buffers: " << packet_buffer_.NumPacketsInBuffer() <<
" packets * " << decoder_frame_length << " samples/packet + " <<
samples_left << " samples in sync buffer = " << cur_size_samples;
diff --git a/webrtc/modules/audio_coding/neteq4/defines.h b/webrtc/modules/audio_coding/neteq4/defines.h
index 67b7cde..b6f9eb2 100644
--- a/webrtc/modules/audio_coding/neteq4/defines.h
+++ b/webrtc/modules/audio_coding/neteq4/defines.h
@@ -47,11 +47,5 @@
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_
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
index d872b80..8e8ffe2 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
@@ -124,7 +124,7 @@
int length_bytes,
uint32_t receive_timestamp) {
CriticalSectionScoped lock(crit_sect_.get());
- NETEQ_LOG_VERBOSE << "InsertPacket: ts=" << rtp_header.header.timestamp <<
+ LOG(LS_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 <<
@@ -143,10 +143,10 @@
int* samples_per_channel, int* num_channels,
NetEqOutputType* type) {
CriticalSectionScoped lock(crit_sect_.get());
- NETEQ_LOG_VERBOSE << "GetAudio";
+ LOG(LS_VERBOSE) << "GetAudio";
int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
num_channels);
- NETEQ_LOG_VERBOSE << "Produced " << *samples_per_channel <<
+ LOG(LS_VERBOSE) << "Produced " << *samples_per_channel <<
" samples/channel for " << *num_channels << " channel(s)";
if (error != 0) {
LOG_FERR1(LS_WARNING, GetAudioInternal, error);
@@ -623,7 +623,7 @@
last_mode_ = kModeError;
return return_value;
}
- NETEQ_LOG_VERBOSE << "GetDecision returned operation=" << operation <<
+ LOG(LS_VERBOSE) << "GetDecision returned operation=" << operation <<
" and " << packet_list.size() << " packet(s)";
AudioDecoder::SpeechType speech_type;
@@ -735,7 +735,7 @@
sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
output));
*num_channels = static_cast<int>(sync_buffer_->Channels());
- NETEQ_LOG_VERBOSE << "Sync buffer (" << *num_channels << " channel(s)):" <<
+ LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" <<
" insert " << algorithm_buffer_->Size() << " samples, extract " <<
samples_from_sync << " samples";
if (samples_from_sync != output_size_samples_) {
@@ -1162,7 +1162,7 @@
int16_t decode_length;
if (!packet->primary) {
// This is a redundant payload; call the special decoder method.
- NETEQ_LOG_VERBOSE << "Decoding packet (redundant):" <<
+ LOG(LS_VERBOSE) << "Decoding packet (redundant):" <<
" ts=" << packet->header.timestamp <<
", sn=" << packet->header.sequenceNumber <<
", pt=" << static_cast<int>(packet->header.payloadType) <<
@@ -1172,7 +1172,7 @@
packet->payload, packet->payload_length,
&decoded_buffer_[*decoded_length], speech_type);
} else {
- NETEQ_LOG_VERBOSE << "Decoding packet: ts=" << packet->header.timestamp <<
+ LOG(LS_VERBOSE) << "Decoding packet: ts=" << packet->header.timestamp <<
", sn=" << packet->header.sequenceNumber <<
", pt=" << static_cast<int>(packet->header.payloadType) <<
", ssrc=" << packet->header.ssrc <<
@@ -1190,7 +1190,7 @@
// Update |decoder_frame_length_| with number of samples per channel.
decoder_frame_length_ = decode_length /
static_cast<int>(decoder->channels());
- NETEQ_LOG_VERBOSE << "Decoded " << decode_length << " samples (" <<
+ LOG(LS_VERBOSE) << "Decoded " << decode_length << " samples (" <<
decoder->channels() << " channel(s) -> " << decoder_frame_length_ <<
" samples per channel)";
} else if (decode_length < 0) {
diff --git a/webrtc/modules/audio_coding/neteq4/timestamp_scaler.cc b/webrtc/modules/audio_coding/neteq4/timestamp_scaler.cc
index 423edee..b2b5b40 100644
--- a/webrtc/modules/audio_coding/neteq4/timestamp_scaler.cc
+++ b/webrtc/modules/audio_coding/neteq4/timestamp_scaler.cc
@@ -85,7 +85,7 @@
assert(denominator_ > 0); // Should not be possible.
external_ref_ = external_timestamp;
internal_ref_ += (external_diff * numerator_) / denominator_;
- NETEQ_LOG_VERBOSE << "Converting timestamp: " << external_timestamp <<
+ LOG(LS_VERBOSE) << "Converting timestamp: " << external_timestamp <<
" -> " << internal_ref_;
return internal_ref_;
} else {