API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.
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
diff --git a/webrtc/modules/audio_coding/neteq/automode.c b/webrtc/modules/audio_coding/neteq/automode.c
index edee98e..ea6fa8d 100644
--- a/webrtc/modules/audio_coding/neteq/automode.c
+++ b/webrtc/modules/audio_coding/neteq/automode.c
@@ -216,6 +216,14 @@
streamingMode);
if (tempvar > 0)
{
+ int high_lim_delay;
+ /* Convert the minimum delay from milliseconds to packets in Q8.
+ * |fsHz| is sampling rate in Hertz, and |inst->packetSpeechLenSamp|
+ * is the number of samples per packet (according to the last
+ * decoding).
+ */
+ int32_t minimum_delay_q8 = ((inst->minimum_delay_ms *
+ (fsHz / 1000)) << 8) / inst->packetSpeechLenSamp;
inst->optBufLevel = tempvar;
if (streamingMode != 0)
@@ -224,6 +232,13 @@
inst->maxCSumIatQ8);
}
+ /* The required delay. */
+ inst->required_delay_q8 = inst->optBufLevel;
+
+ // Maintain the target delay.
+ inst->optBufLevel = WEBRTC_SPL_MAX(inst->optBufLevel,
+ minimum_delay_q8);
+
/*********/
/* Limit */
/*********/
@@ -238,8 +253,12 @@
maxBufLen = WEBRTC_SPL_LSHIFT_W32(maxBufLen, 8); /* shift to Q8 */
/* Enforce upper limit; 75% of maxBufLen */
- inst->optBufLevel = WEBRTC_SPL_MIN( inst->optBufLevel,
- (maxBufLen >> 1) + (maxBufLen >> 2) ); /* 1/2 + 1/4 = 75% */
+ /* 1/2 + 1/4 = 75% */
+ high_lim_delay = (maxBufLen >> 1) + (maxBufLen >> 2);
+ inst->optBufLevel = WEBRTC_SPL_MIN(inst->optBufLevel,
+ high_lim_delay);
+ inst->required_delay_q8 = WEBRTC_SPL_MIN(inst->required_delay_q8,
+ high_lim_delay);
}
else
{
@@ -700,6 +719,7 @@
*/
inst->optBufLevel = WEBRTC_SPL_MIN(4,
(maxBufLenPackets >> 1) + (maxBufLenPackets >> 1)); /* 75% of maxBufLenPackets */
+ inst->required_delay_q8 = inst->optBufLevel;
inst->levelFiltFact = 253;
/*
diff --git a/webrtc/modules/audio_coding/neteq/automode.h b/webrtc/modules/audio_coding/neteq/automode.h
index 5996a51..49878c0 100644
--- a/webrtc/modules/audio_coding/neteq/automode.h
+++ b/webrtc/modules/audio_coding/neteq/automode.h
@@ -89,6 +89,12 @@
reached 0 */
int16_t extraDelayMs; /* extra delay for sync with video */
+ int minimum_delay_ms; /* Desired delay, NetEq maintains this amount of
+ delay unless jitter statistics suggests a higher value. */
+ int required_delay_q8; /* Smallest delay required. This is computed
+ according to inter-arrival time and playout mode. It has the same unit
+ as |optBufLevel|. */
+
/* Peak-detection */
/* vector with the latest peak periods (peak spacing in samples) */
uint32_t peakPeriodSamp[NUM_PEAKS];
diff --git a/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h b/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h
index 4eefce0..021704c 100644
--- a/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h
+++ b/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h
@@ -309,6 +309,19 @@
WebRtcNetEQ_RTPInfo* rtp_info,
uint32_t receive_timestamp);
+/*
+ * Set a minimum latency for the jitter buffer. The overall delay is the max of
+ * |minimum_delay_ms| and the latency that is internally computed based on the
+ * inter-arrival times.
+ */
+int WebRtcNetEQ_SetMinimumDelay(void *inst, int minimum_delay_ms);
+
+/*
+ * Get the least required delay in milliseconds given inter-arrival times
+ * and playout mode.
+ */
+int WebRtcNetEQ_GetRequiredDelayMs(const void* inst);
+
#ifdef __cplusplus
}
#endif
diff --git a/webrtc/modules/audio_coding/neteq/mcu_reset.c b/webrtc/modules/audio_coding/neteq/mcu_reset.c
index 3aae4ce..c8a4cd7 100644
--- a/webrtc/modules/audio_coding/neteq/mcu_reset.c
+++ b/webrtc/modules/audio_coding/neteq/mcu_reset.c
@@ -32,7 +32,9 @@
inst->main_inst = NULL;
inst->one_desc = 0;
inst->BufferStat_inst.Automode_inst.extraDelayMs = 0;
+ inst->BufferStat_inst.Automode_inst.minimum_delay_ms = 0;
inst->NetEqPlayoutMode = kPlayoutOn;
+ inst->av_sync = 0;
WebRtcNetEQ_DbReset(&inst->codec_DB_inst);
memset(&inst->PayloadSplit_inst, 0, sizeof(SplitInfo_t));
diff --git a/webrtc/modules/audio_coding/neteq/webrtc_neteq.c b/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
index 31940c8..8347925 100644
--- a/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+++ b/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
@@ -437,6 +437,7 @@
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 @@
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 @@
/* 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 @@
}
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;
+}