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Allow NetEQ to use real packet durations.

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This is a copy of http://review.webrtc.org/864014/

This adds a FuncDurationEst to each codec instance which estimates
the duration of a packet given the packet contents and the duration
of the previous packet. By default, this simply returns the
duration of the previous packet (which is what is currently assumed
to be the duration of all future packets). This patch also provides
an initial implementation of this function for G.711 which returns
the actual number of samples in the packet.

BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/935016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3129 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
2012-11-19 08:02:55 +00:00
parent 3662aa38f3
commit 5ac387c4d1
12 changed files with 230 additions and 43 deletions
Download Patch File Download Diff File Expand all files Collapse all files

37
webrtc/modules/audio_coding/neteq/signal_mcu.c
View File

@ -31,6 +31,27 @@ extern FILE *delay_fid2; /* file pointer to delay log file */
#endif
/*
* Update the frame size, if we can.
*/
static int WebRtcNetEQ_UpdatePackSizeSamples(MCUInst_t* inst, int buffer_pos,
int payload_type,
int pack_size_samples) {
if (buffer_pos >= 0) {
int codec_pos;
codec_pos = WebRtcNetEQ_DbGetCodec(&inst->codec_DB_inst, payload_type);
if (codec_pos >= 0) {
codec_pos = inst->codec_DB_inst.position[codec_pos];
if (codec_pos >= 0) {
return WebRtcNetEQ_PacketBufferGetPacketSize(
&inst->PacketBuffer_inst, buffer_pos,
&inst->codec_DB_inst, codec_pos, pack_size_samples);
}
}
}
return pack_size_samples;
}
/*
* Signals the MCU that DSP status data is available.
*/
@ -223,7 +244,8 @@ int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
}
/* Check packet buffer */
w32_bufsize = WebRtcNetEQ_PacketBufferGetSize(&inst->PacketBuffer_inst);
w32_bufsize = WebRtcNetEQ_PacketBufferGetSize(&inst->PacketBuffer_inst,
&inst->codec_DB_inst);
if (dspInfo.lastMode == MODE_SUCCESS_ACCELERATE || dspInfo.lastMode
== MODE_LOWEN_ACCELERATE || dspInfo.lastMode == MODE_SUCCESS_PREEMPTIVE
@ -254,6 +276,10 @@ int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
}
#endif
/* Update the frame size, if we can. */
inst->PacketBuffer_inst.packSizeSamples =
WebRtcNetEQ_UpdatePackSizeSamples(inst, i_bufferpos, payloadType,
inst->PacketBuffer_inst.packSizeSamples);
/* Update statistics and make decision */
uw16_instr = WebRtcNetEQ_BufstatsDecision(&inst->BufferStat_inst,
inst->PacketBuffer_inst.packSizeSamples, w32_bufsize, dspInfo.playedOutTS,
@ -350,7 +376,9 @@ int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
}
/* Set the packet size by guessing */
inst->PacketBuffer_inst.packSizeSamples = inst->timestampsPerCall * 3;
inst->PacketBuffer_inst.packSizeSamples =
WebRtcNetEQ_UpdatePackSizeSamples(inst, i_bufferpos, payloadType,
inst->timestampsPerCall * 3);
WebRtcNetEQ_ResetAutomode(&(inst->BufferStat_inst.Automode_inst),
inst->PacketBuffer_inst.maxInsertPositions);
@ -695,7 +723,10 @@ int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
}
prevSeqNo = inst->PacketBuffer_inst.seqNumber[i_bufferpos];
}
/* Update the frame size, if we can. */
inst->PacketBuffer_inst.packSizeSamples =
WebRtcNetEQ_UpdatePackSizeSamples(inst, i_bufferpos,
payloadType, inst->PacketBuffer_inst.packSizeSamples);
}
while ((totalTS < wantedNoOfTimeStamps) && (nextSeqNoAvail == 1));
}