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Mainly hlundin's patch.

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Review URL: https://webrtc-codereview.appspot.com/1052004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3405 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
niklas.enbom@webrtc.org
2013-01-24 18:53:43 +00:00
parent 4782911572
commit 05e7bfeeea
12 changed files with 277 additions and 16 deletions
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11
webrtc/modules/audio_coding/neteq/bufstats_decision.c
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@ -38,11 +38,11 @@ WebRtc_UWord16 WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, WebRtc_Word16
int currentDelayMs;
WebRtc_Word32 currSizeSamples = cur_size;
WebRtc_Word16 extraDelayPacketsQ8 = 0;
int extraDelayPacketsQ8 = 0;
/* Avoid overflow if the buffer size should be really large (cur_size is limited 256ms) */
WebRtc_Word32 curr_sizeQ7 = WEBRTC_SPL_LSHIFT_W32(cur_size, 4);
WebRtc_UWord16 level_limit_hi, level_limit_lo;
int level_limit_hi, level_limit_lo;
inst->Automode_inst.prevTimeScale &= (prevPlayMode == MODE_SUCCESS_ACCELERATE
|| prevPlayMode == MODE_LOWEN_ACCELERATE || prevPlayMode == MODE_SUCCESS_PREEMPTIVE
@ -167,10 +167,11 @@ WebRtc_UWord16 WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, WebRtc_Word16
if (inst->Automode_inst.extraDelayMs > 0 && inst->Automode_inst.packetSpeechLenSamp
> 0)
{
extraDelayPacketsQ8 = WebRtcSpl_DivW32W16ResW16(
(WEBRTC_SPL_MUL(inst->Automode_inst.extraDelayMs, 8 * fs_mult) << 8),
inst->Automode_inst.packetSpeechLenSamp);
/* (extra delay in samples in Q8) */
extraDelayPacketsQ8 =
((inst->Automode_inst.extraDelayMs * 8 * fs_mult) << 8) /
inst->Automode_inst.packetSpeechLenSamp;
}
/* Check if needed packet is available */