Remove Channel::UpdatePacketDelay and some member variables
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}
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index c2e0891..d6daffd 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -530,10 +530,6 @@
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 @@
_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 @@
}
}
-// 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 @@
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(¤t_recive_codec) == 0) {
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index adeb1da..1cf49d9 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -466,14 +466,13 @@
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 @@
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;
