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NetEq: Remove overly verbose logging

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This change removes all LS_VERBOSE logs that will print once every
packet or more often.

TBR=pbos@webrtc.org
BUG=webrtc:5227

Review URL: https://codereview.webrtc.org/1461903004

Cr-Commit-Position: refs/heads/master@{#10733}
This commit is contained in:
henrik.lundin 2015-11-20 11:57:05 -08:00 committed by Commit bot
parent 5def7b9fde
commit 672304a654
3 changed files with 0 additions and 45 deletions
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3
webrtc/modules/audio_coding/neteq/decision_logic.cc
View File

@ -128,9 +128,6 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
const size_t cur_size_samples =
samples_left + packet_buffer_.NumSamplesInBuffer(decoder_database_,
decoder_frame_length);
LOG(LS_VERBOSE) << "Buffers: " << packet_buffer_.NumPacketsInBuffer() <<
" packets * " << decoder_frame_length << " samples/packet + " <<
samples_left << " samples in sync buffer = " << cur_size_samples;
prev_time_scale_ = prev_time_scale_ &&
(prev_mode == kModeAccelerateSuccess ||

40
webrtc/modules/audio_coding/neteq/neteq_impl.cc
View File

@ -124,11 +124,6 @@ int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp) {
CriticalSectionScoped lock(crit_sect_.get());
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
<< ", len=" << payload.size();
int error =
InsertPacketInternal(rtp_header, payload, receive_timestamp, false);
if (error != 0) {
@ -141,12 +136,6 @@ int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
uint32_t receive_timestamp) {
CriticalSectionScoped lock(crit_sect_.get());
LOG(LS_VERBOSE) << "InsertPacket-Sync: ts="
<< rtp_header.header.timestamp <<
", sn=" << rtp_header.header.sequenceNumber <<
", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
", ssrc=" << rtp_header.header.ssrc;
const uint8_t kSyncPayload[] = { 's', 'y', 'n', 'c' };
int error =
InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true);
@ -162,11 +151,8 @@ int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
size_t* samples_per_channel, int* num_channels,
NetEqOutputType* type) {
CriticalSectionScoped lock(crit_sect_.get());
LOG(LS_VERBOSE) << "GetAudio";
int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
num_channels);
LOG(LS_VERBOSE) << "Produced " << *samples_per_channel <<
" samples/channel for " << *num_channels << " channel(s)";
if (error != 0) {
error_code_ = error;
return kFail;
@ -751,8 +737,6 @@ int NetEqImpl::GetAudioInternal(size_t max_length,
last_mode_ = kModeError;
return return_value;
}
LOG(LS_VERBOSE) << "GetDecision returned operation=" << operation <<
" and " << packet_list.size() << " packet(s)";
AudioDecoder::SpeechType speech_type;
int length = 0;
@ -867,9 +851,6 @@ int NetEqImpl::GetAudioInternal(size_t max_length,
sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
output);
*num_channels = static_cast<int>(sync_buffer_->Channels());
LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" <<
" insert " << algorithm_buffer_->Size() << " samples, extract " <<
samples_from_sync << " samples";
if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
// The sync buffer should always contain |overlap_length| samples, but now
// too many samples have been extracted. Reinstall the |overlap_length|
@ -1327,7 +1308,6 @@ int NetEqImpl::DecodeCng(AudioDecoder* decoder, int* decoded_length,
&decoded_buffer_[*decoded_length], speech_type);
if (length > 0) {
*decoded_length += length;
LOG(LS_VERBOSE) << "Decoded " << length << " CNG samples";
} else {
// Error.
LOG(LS_WARNING) << "Failed to decode CNG";
@ -1367,34 +1347,17 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation,
int decode_length;
if (packet->sync_packet) {
// Decode to silence with the same frame size as the last decode.
LOG(LS_VERBOSE) << "Decoding sync-packet: " <<
" ts=" << packet->header.timestamp <<
", sn=" << packet->header.sequenceNumber <<
", pt=" << static_cast<int>(packet->header.payloadType) <<
", ssrc=" << packet->header.ssrc <<
", len=" << packet->payload_length;
memset(&decoded_buffer_[*decoded_length], 0,
decoder_frame_length_ * decoder->Channels() *
sizeof(decoded_buffer_[0]));
decode_length = rtc::checked_cast<int>(decoder_frame_length_);
} else if (!packet->primary) {
// This is a redundant payload; call the special decoder method.
LOG(LS_VERBOSE) << "Decoding packet (redundant):" <<
" ts=" << packet->header.timestamp <<
", sn=" << packet->header.sequenceNumber <<
", pt=" << static_cast<int>(packet->header.payloadType) <<
", ssrc=" << packet->header.ssrc <<
", len=" << packet->payload_length;
decode_length = decoder->DecodeRedundant(
packet->payload, packet->payload_length, fs_hz_,
(decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
&decoded_buffer_[*decoded_length], speech_type);
} else {
LOG(LS_VERBOSE) << "Decoding packet: ts=" << packet->header.timestamp <<
", sn=" << packet->header.sequenceNumber <<
", pt=" << static_cast<int>(packet->header.payloadType) <<
", ssrc=" << packet->header.ssrc <<
", len=" << packet->payload_length;
decode_length =
decoder->Decode(
packet->payload, packet->payload_length, fs_hz_,
@ -1410,9 +1373,6 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation,
// Update |decoder_frame_length_| with number of samples per channel.
decoder_frame_length_ =
static_cast<size_t>(decode_length) / decoder->Channels();
LOG(LS_VERBOSE) << "Decoded " << decode_length << " samples ("
<< decoder->Channels() << " channel(s) -> "
<< decoder_frame_length_ << " samples per channel)";
} else if (decode_length < 0) {
// Error.
LOG(LS_WARNING) << "Decode " << decode_length << " " << payload_length;

2
webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
View File

@ -87,8 +87,6 @@ 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 <<
" -> " << internal_ref_;
return internal_ref_;
} else {
// No scaling.