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Remove Channel::UpdatePacketDelay and some member variables

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The method is no longer used, since the jitter buffer delay is
obtained directly from AudioCodingModule instead of being calculated
and smoothed in VoiceEngine. Deleting a few obsolete member variables
as well.

BUG=webrtc:6237

Review-Url: https://codereview.webrtc.org/2290253002
Cr-Commit-Position: refs/heads/master@{#14007}
This commit is contained in:
henrik.lundin
2016-08-31 14:09:51 -07:00
committed by Commit bot
parent 5a601d909f
commit b3e30010de
2 changed files with 2 additions and 76 deletions
Download Patch File Download Diff File Expand all files Collapse all files

72
webrtc/voice_engine/channel.cc
View File

@ -530,10 +530,6 @@ int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData,
return -1;
}
// Update the packet delay.
UpdatePacketDelay(rtpHeader->header.timestamp,
rtpHeader->header.sequenceNumber);
int64_t round_trip_time = 0;
_rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time, NULL, NULL,
NULL);
@ -880,9 +876,6 @@ Channel::Channel(int32_t channelId,
_lastPayloadType(0),
_includeAudioLevelIndication(false),
_outputSpeechType(AudioFrame::kNormalSpeech),
_average_jitter_buffer_delay_us(0),
_previousTimestamp(0),
_recPacketDelayMs(20),
_RxVadDetection(false),
_rxAgcIsEnabled(false),
_rxNsIsEnabled(false),
@ -3385,69 +3378,6 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) {
}
}
// Called for incoming RTP packets after successful RTP header parsing.
// TODO(henrik.lundin): Clean out this method. With the introduction of
// AudioCoding::FilteredCurrentDelayMs() most (if not all) of this method can
// be deleted, along with a few member variables. (WebRTC issue 6237.)
void Channel::UpdatePacketDelay(uint32_t rtp_timestamp,
uint16_t sequence_number) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::UpdatePacketDelay(timestamp=%lu, sequenceNumber=%u)",
rtp_timestamp, sequence_number);
// Get frequency of last received payload
int rtp_receive_frequency = GetPlayoutFrequency();
// |jitter_buffer_playout_timestamp_| updated in UpdatePlayoutTimestamp for
// every incoming packet. May be empty if no valid playout timestamp is
// available.
// If |rtp_timestamp| is newer than |jitter_buffer_playout_timestamp_|, the
// resulting difference is positive and will be used. When the inverse is
// true (can happen when a network glitch causes a packet to arrive late,
// and during long comfort noise periods with clock drift), or when
// |jitter_buffer_playout_timestamp_| has no value, the difference is not
// changed from the initial 0.
uint32_t timestamp_diff_ms = 0;
if (jitter_buffer_playout_timestamp_ &&
IsNewerTimestamp(rtp_timestamp, *jitter_buffer_playout_timestamp_)) {
timestamp_diff_ms = (rtp_timestamp - *jitter_buffer_playout_timestamp_) /
(rtp_receive_frequency / 1000);
if (timestamp_diff_ms > (2 * kVoiceEngineMaxMinPlayoutDelayMs)) {
// Diff is too large; set it to zero instead.
timestamp_diff_ms = 0;
}
}
uint16_t packet_delay_ms =
(rtp_timestamp - _previousTimestamp) / (rtp_receive_frequency / 1000);
_previousTimestamp = rtp_timestamp;
if (timestamp_diff_ms == 0)
return;
{
rtc::CritScope lock(&video_sync_lock_);
if (packet_delay_ms >= 10 && packet_delay_ms <= 60) {
_recPacketDelayMs = packet_delay_ms;
}
if (_average_jitter_buffer_delay_us == 0) {
_average_jitter_buffer_delay_us = timestamp_diff_ms * 1000;
return;
}
// Filter average delay value using exponential filter (alpha is
// 7/8). We derive 1000 *_average_jitter_buffer_delay_us here (reduces
// risk of rounding error) and compensate for it in GetDelayEstimate()
// later.
_average_jitter_buffer_delay_us =
(_average_jitter_buffer_delay_us * 7 + 1000 * timestamp_diff_ms + 500) /
8;
}
}
void Channel::RegisterReceiveCodecsToRTPModule() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterReceiveCodecsToRTPModule()");
@ -3491,7 +3421,7 @@ int Channel::SetSendRtpHeaderExtension(bool enable,
return error;
}
int32_t Channel::GetPlayoutFrequency() {
int32_t Channel::GetPlayoutFrequency() const {
int32_t playout_frequency = audio_coding_->PlayoutFrequency();
CodecInst current_recive_codec;
if (audio_coding_->ReceiveCodec(&current_recive_codec) == 0) {

6
webrtc/voice_engine/channel.h
View File

@ -466,14 +466,13 @@ class Channel
int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
void UpdatePlayoutTimestamp(bool rtcp);
void UpdatePacketDelay(uint32_t timestamp, uint16_t sequenceNumber);
void RegisterReceiveCodecsToRTPModule();
int SetSendRtpHeaderExtension(bool enable,
RTPExtensionType type,
unsigned char id);
int32_t GetPlayoutFrequency();
int32_t GetPlayoutFrequency() const;
int64_t GetRTT(bool allow_associate_channel) const;
rtc::CriticalSection _fileCritSect;
@ -565,9 +564,6 @@ class Channel
AudioFrame::SpeechType _outputSpeechType;
// VoEVideoSync
rtc::CriticalSection video_sync_lock_;
uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_);
uint32_t _previousTimestamp;
uint16_t _recPacketDelayMs GUARDED_BY(video_sync_lock_);
// VoEAudioProcessing
bool _RxVadDetection;
bool _rxAgcIsEnabled;