Removing deprecated NetEQ APIs
Removing WebRtcNetEQ_GetPreferredBufferSize and WebRtcNetEQ_GetCurrentDelay and all dependent APIs. Review URL: http://webrtc-codereview.appspot.com/289006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1063 4adac7df-926f-26a2-2b94-8c16560cd09d
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@@ -1188,9 +1188,28 @@ int WebRtcNetEQ_GetNetworkStatistics(void *inst, WebRtcNetEQ_NetworkStatistics *
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/* Get buffer size */
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/*******************/
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if (WebRtcNetEQ_GetCurrentDelay((void *) inst, &(stats->currentBufferSize)) != 0)
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if (NetEqMainInst->MCUinst.fs != 0)
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{
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return (-1);
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WebRtc_Word32 temp32;
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/* Query packet buffer for number of samples. */
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temp32 = WebRtcNetEQ_PacketBufferGetSize(
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&NetEqMainInst->MCUinst.PacketBuffer_inst);
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/* Divide by sample rate.
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* Calculate temp32 * 1000 / fs to get result in ms. */
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stats->currentBufferSize = (WebRtc_UWord16)
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WebRtcSpl_DivU32U16(temp32 * 1000, NetEqMainInst->MCUinst.fs);
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/* Add number of samples yet to play in sync buffer. */
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temp32 = (WebRtc_Word32) (NetEqMainInst->DSPinst.endPosition -
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NetEqMainInst->DSPinst.curPosition);
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stats->currentBufferSize += (WebRtc_UWord16)
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WebRtcSpl_DivU32U16(temp32 * 1000, NetEqMainInst->MCUinst.fs);
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}
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else
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{
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/* Sample rate not initialized. */
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stats->currentBufferSize = 0;
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}
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/***************************/
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@@ -1512,81 +1531,6 @@ int WebRtcNetEQ_GetNetworkStatistics(void *inst, WebRtcNetEQ_NetworkStatistics *
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return (0);
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}
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/* Get the optimal buffer size calculated for the current network conditions. */
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int WebRtcNetEQ_GetPreferredBufferSize(void *inst, WebRtc_UWord16 *preferredBufferSize)
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{
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MainInst_t *NetEqMainInst = (MainInst_t*) inst;
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/***************************/
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/* Get optimal buffer size */
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/***************************/
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if (NetEqMainInst->MCUinst.fs != 0 && NetEqMainInst->MCUinst.fs <= WEBRTC_SPL_WORD16_MAX)
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{
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/* preferredBufferSize = Bopt * packSizeSamples / (fs/1000) */
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*preferredBufferSize
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= (WebRtc_UWord16) WEBRTC_SPL_MUL_16_16(
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(WebRtc_Word16) ((NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.optBufLevel) >> 8), /* optimal buffer level in packets shifted to Q0 */
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WebRtcSpl_DivW32W16ResW16(
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(WebRtc_Word32) NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.packetSpeechLenSamp, /* samples per packet */
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WebRtcSpl_DivW32W16ResW16( (WebRtc_Word32) NetEqMainInst->MCUinst.fs, (WebRtc_Word16) 1000 ) /* samples per ms */
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) );
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/* add extra delay */
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if (NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs > 0)
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{
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*preferredBufferSize
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+= NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs;
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}
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}
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else
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{
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/* sample rate not initialized */
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*preferredBufferSize = 0;
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}
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return (0);
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}
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/* Get the current buffer size in ms. Return value is 0 if ok, -1 if error. */
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int WebRtcNetEQ_GetCurrentDelay(const void *inst, WebRtc_UWord16 *currentDelayMs)
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{
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WebRtc_Word32 temp32;
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const MainInst_t *NetEqMainInst = (const MainInst_t*) inst;
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/* Instance sanity */
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if (NetEqMainInst == NULL) return (-1);
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/*******************/
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/* Get buffer size */
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/*******************/
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if (NetEqMainInst->MCUinst.fs != 0 && NetEqMainInst->MCUinst.fs <= WEBRTC_SPL_WORD16_MAX)
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{
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/* query packet buffer for number of samples */
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temp32 = WebRtcNetEQ_PacketBufferGetSize(&NetEqMainInst->MCUinst.PacketBuffer_inst);
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/* divide by sample rate */
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*currentDelayMs = WebRtcSpl_DivW32W16ResW16(temp32 * 1000, /* multiply by 1000 to get ms */
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(WebRtc_Word16) NetEqMainInst->MCUinst.fs); /* divide by fs in samples per second */
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/* add number of samples yet to play in sync buffer */
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temp32 = (WebRtc_Word32) (NetEqMainInst->DSPinst.endPosition
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- NetEqMainInst->DSPinst.curPosition);
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*currentDelayMs += WebRtcSpl_DivW32W16ResW16(temp32 * 1000, /* multiply by 1000 to get ms */
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(WebRtc_Word16) NetEqMainInst->MCUinst.fs); /* divide by fs in samples per second */
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}
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else
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{
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/* sample rate not initialized */
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*currentDelayMs = 0;
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}
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return 0;
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}
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/****************************************************************************
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* WebRtcNetEQ_SetVADInstance(...)
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*
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