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API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.

Browse Source 
TEST=unit-test, manual, trybots.
R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1384005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org
2013-05-22 20:39:43 +00:00
parent 561990fd73
commit e46c8d3875
19 changed files with 405 additions and 74 deletions
Download Patch File Download Diff File Expand all files Collapse all files

38
webrtc/modules/audio_coding/neteq/webrtc_neteq.c
View File

@ -437,6 +437,7 @@ int WebRtcNetEQ_Init(void *inst, uint16_t fs)
NetEqMainInst->MCUinst.first_packet = 1;
NetEqMainInst->MCUinst.one_desc = 0;
NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs = 0;
NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.minimum_delay_ms = 0;
NetEqMainInst->MCUinst.NoOfExpandCalls = 0;
NetEqMainInst->MCUinst.fs = fs;
@ -529,6 +530,19 @@ int WebRtcNetEQ_SetExtraDelay(void *inst, int DelayInMs)
return (0);
}
int WebRtcNetEQ_SetMinimumDelay(void *inst, int minimum_delay_ms) {
MainInst_t *NetEqMainInst = (MainInst_t*) inst;
if (NetEqMainInst == NULL)
return -1;
if (minimum_delay_ms < 0 || minimum_delay_ms > 10000) {
NetEqMainInst->ErrorCode = -FAULTY_DELAYVALUE;
return -1;
}
NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.minimum_delay_ms =
minimum_delay_ms;
return 0;
}
int WebRtcNetEQ_SetPlayoutMode(void *inst, enum WebRtcNetEQPlayoutMode playoutMode)
{
MainInst_t *NetEqMainInst = (MainInst_t*) inst;
@ -1213,7 +1227,7 @@ int WebRtcNetEQ_GetNetworkStatistics(void *inst, WebRtcNetEQ_NetworkStatistics *
/* Get optimal buffer size */
/***************************/
if (NetEqMainInst->MCUinst.fs != 0 && NetEqMainInst->MCUinst.fs <= WEBRTC_SPL_WORD16_MAX)
if (NetEqMainInst->MCUinst.fs != 0)
{
/* preferredBufferSize = Bopt * packSizeSamples / (fs/1000) */
stats->preferredBufferSize
@ -1693,3 +1707,25 @@ int WebRtcNetEQ_RecInSyncRTP(void* inst, WebRtcNetEQ_RTPInfo* rtp_info,
}
return SYNC_PAYLOAD_LEN_BYTES;
}
int WebRtcNetEQ_GetRequiredDelayMs(const void* inst) {
const MainInst_t* NetEqMainInst = (MainInst_t*)inst;
const AutomodeInst_t* auto_mode = (NetEqMainInst == NULL) ? NULL :
&NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst;
/* Instance sanity */
if (NetEqMainInst == NULL || auto_mode == NULL)
return 0;
if (NetEqMainInst->MCUinst.fs == 0)
return 0; // Sampling rate not initialized.
/* |required_delay_q8| has the unit of packets in Q8 domain, therefore,
* the corresponding delay is
* required_delay_ms = (1000 * required_delay_q8 * samples_per_packet /
* sample_rate_hz) / 256;
*/
return (auto_mode->required_delay_q8 *
((auto_mode->packetSpeechLenSamp * 1000) / NetEqMainInst->MCUinst.fs) +
128) >> 8;
}