/// <summary>
/// Encode frame with Silk
/// Note: if prefillFlag is set, the input must contain 10 ms of audio, irrespective of what
/// encControl.payloadSize_ms is set to
/// </summary>
/// <param name="psEnc">I/O State</param>
/// <param name="encControl">I Control status</param>
/// <param name="samplesIn">I Speech sample input vector</param>
/// <param name="nSamplesIn">I Number of samples in input vector</param>
/// <param name="psRangeEnc">I/O Compressor data structure</param>
/// <param name="nBytesOut">I/O Number of bytes in payload (input: Max bytes)</param>
/// <param name="prefillFlag">I Flag to indicate prefilling buffers no coding</param>
/// <returns>error code</returns>
internal static int silk_Encode(
SilkEncoder psEnc,
EncControlState encControl,
short[] samplesIn,
int nSamplesIn,
EntropyCoder psRangeEnc,
BoxedValueInt nBytesOut,
int prefillFlag)
{
int ret = SilkError.SILK_NO_ERROR;
int n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0;
int nSamplesToBuffer, nSamplesToBufferMax, nBlocksOf10ms;
int nSamplesFromInput = 0, nSamplesFromInputMax;
int speech_act_thr_for_switch_Q8;
int TargetRate_bps, channelRate_bps, LBRR_symbol, sum;
int[] MStargetRates_bps = new int[2];
short[] buf;
int transition, curr_block, tot_blocks;
nBytesOut.Val = 0;
if (encControl.reducedDependency != 0)
{
psEnc.state_Fxx[0].first_frame_after_reset = 1;
psEnc.state_Fxx[1].first_frame_after_reset = 1;
}
psEnc.state_Fxx[0].nFramesEncoded = psEnc.state_Fxx[1].nFramesEncoded = 0;
/* Check values in encoder control structure */
ret += encControl.check_control_input();
if (ret != SilkError.SILK_NO_ERROR)
{
Inlines.OpusAssert(false);
return(ret);
}
encControl.switchReady = 0;
if (encControl.nChannelsInternal > psEnc.nChannelsInternal)
{
/* Mono . Stereo transition: init state of second channel and stereo state */
ret += SilkEncoder.silk_init_encoder(psEnc.state_Fxx[1]);
Arrays.MemSetShort(psEnc.sStereo.pred_prev_Q13, 0, 2);
Arrays.MemSetShort(psEnc.sStereo.sSide, 0, 2);
psEnc.sStereo.mid_side_amp_Q0[0] = 0;
psEnc.sStereo.mid_side_amp_Q0[1] = 1;
psEnc.sStereo.mid_side_amp_Q0[2] = 0;
psEnc.sStereo.mid_side_amp_Q0[3] = 1;
psEnc.sStereo.width_prev_Q14 = 0;
psEnc.sStereo.smth_width_Q14 = (short)(((int)((1.0f) * ((long)1 << (14)) + 0.5)) /*Inlines.SILK_CONST(1.0f, 14)*/);
if (psEnc.nChannelsAPI == 2)
{
psEnc.state_Fxx[1].resampler_state.Assign(psEnc.state_Fxx[0].resampler_state);
Array.Copy(psEnc.state_Fxx[0].In_HP_State, psEnc.state_Fxx[1].In_HP_State, 2);
}
}
transition = ((encControl.payloadSize_ms != psEnc.state_Fxx[0].PacketSize_ms) || (psEnc.nChannelsInternal != encControl.nChannelsInternal)) ? 1 : 0;
psEnc.nChannelsAPI = encControl.nChannelsAPI;
psEnc.nChannelsInternal = encControl.nChannelsInternal;
nBlocksOf10ms = Inlines.silk_DIV32(100 * nSamplesIn, encControl.API_sampleRate);
tot_blocks = (nBlocksOf10ms > 1) ? nBlocksOf10ms >> 1 : 1;
curr_block = 0;
if (prefillFlag != 0)
{
/* Only accept input length of 10 ms */
if (nBlocksOf10ms != 1)
{
Inlines.OpusAssert(false);
return(SilkError.SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES);
}
/* Reset Encoder */
for (n = 0; n < encControl.nChannelsInternal; n++)
{
ret += SilkEncoder.silk_init_encoder(psEnc.state_Fxx[n]);
Inlines.OpusAssert(ret == SilkError.SILK_NO_ERROR);
}
tmp_payloadSize_ms = encControl.payloadSize_ms;
encControl.payloadSize_ms = 10;
tmp_complexity = encControl.complexity;
encControl.complexity = 0;
for (n = 0; n < encControl.nChannelsInternal; n++)
{
psEnc.state_Fxx[n].controlled_since_last_payload = 0;
psEnc.state_Fxx[n].prefillFlag = 1;
}
}
else
{
/* Only accept input lengths that are a multiple of 10 ms */
if (nBlocksOf10ms * encControl.API_sampleRate != 100 * nSamplesIn || nSamplesIn < 0)
{
Inlines.OpusAssert(false);
return(SilkError.SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES);
}
/* Make sure no more than one packet can be produced */
if (1000 * (int)nSamplesIn > encControl.payloadSize_ms * encControl.API_sampleRate)
{
Inlines.OpusAssert(false);
return(SilkError.SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES);
}
}
TargetRate_bps = Inlines.silk_RSHIFT32(encControl.bitRate, encControl.nChannelsInternal - 1);
for (n = 0; n < encControl.nChannelsInternal; n++)
{
/* Force the side channel to the same rate as the mid */
int force_fs_kHz = (n == 1) ? psEnc.state_Fxx[0].fs_kHz : 0;
ret += psEnc.state_Fxx[n].silk_control_encoder(encControl, TargetRate_bps, psEnc.allowBandwidthSwitch, n, force_fs_kHz);
if (ret != SilkError.SILK_NO_ERROR)
{
Inlines.OpusAssert(false);
return(ret);
}
if (psEnc.state_Fxx[n].first_frame_after_reset != 0 || transition != 0)
{
for (i = 0; i < psEnc.state_Fxx[0].nFramesPerPacket; i++)
{
psEnc.state_Fxx[n].LBRR_flags[i] = 0;
}
}
psEnc.state_Fxx[n].inDTX = psEnc.state_Fxx[n].useDTX;
}
Inlines.OpusAssert(encControl.nChannelsInternal == 1 || psEnc.state_Fxx[0].fs_kHz == psEnc.state_Fxx[1].fs_kHz);
/* Input buffering/resampling and encoding */
nSamplesToBufferMax = 10 * nBlocksOf10ms * psEnc.state_Fxx[0].fs_kHz;
nSamplesFromInputMax =
Inlines.silk_DIV32_16(nSamplesToBufferMax *
psEnc.state_Fxx[0].API_fs_Hz,
(short)(psEnc.state_Fxx[0].fs_kHz * 1000));
buf = new short[nSamplesFromInputMax];
int samplesIn_ptr = 0;
while (true)
{
nSamplesToBuffer = psEnc.state_Fxx[0].frame_length - psEnc.state_Fxx[0].inputBufIx;
nSamplesToBuffer = Inlines.silk_min(nSamplesToBuffer, nSamplesToBufferMax);
nSamplesFromInput = Inlines.silk_DIV32_16(nSamplesToBuffer * psEnc.state_Fxx[0].API_fs_Hz, psEnc.state_Fxx[0].fs_kHz * 1000);
/* Resample and write to buffer */
if (encControl.nChannelsAPI == 2 && encControl.nChannelsInternal == 2)
{
int id = psEnc.state_Fxx[0].nFramesEncoded;
for (n = 0; n < nSamplesFromInput; n++)
{
buf[n] = samplesIn[samplesIn_ptr + (2 * n)];
}
/* Making sure to start both resamplers from the same state when switching from mono to stereo */
if (psEnc.nPrevChannelsInternal == 1 && id == 0)
{
//silk_memcpy(&psEnc.state_Fxx[1].resampler_state, &psEnc.state_Fxx[0].resampler_state, sizeof(psEnc.state_Fxx[1].resampler_state));
psEnc.state_Fxx[1].resampler_state.Assign(psEnc.state_Fxx[0].resampler_state);
}
ret += Resampler.silk_resampler(
psEnc.state_Fxx[0].resampler_state,
psEnc.state_Fxx[0].inputBuf,
psEnc.state_Fxx[0].inputBufIx + 2,
buf,
0,
nSamplesFromInput);
psEnc.state_Fxx[0].inputBufIx += nSamplesToBuffer;
nSamplesToBuffer = psEnc.state_Fxx[1].frame_length - psEnc.state_Fxx[1].inputBufIx;
nSamplesToBuffer = Inlines.silk_min(nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc.state_Fxx[1].fs_kHz);
for (n = 0; n < nSamplesFromInput; n++)
{
buf[n] = samplesIn[samplesIn_ptr + (2 * n) + 1];
}
ret += Resampler.silk_resampler(
psEnc.state_Fxx[1].resampler_state,
psEnc.state_Fxx[1].inputBuf,
psEnc.state_Fxx[1].inputBufIx + 2,
buf,
0,
nSamplesFromInput);
psEnc.state_Fxx[1].inputBufIx += nSamplesToBuffer;
}
else if (encControl.nChannelsAPI == 2 && encControl.nChannelsInternal == 1)
{
/* Combine left and right channels before resampling */
for (n = 0; n < nSamplesFromInput; n++)
{
sum = samplesIn[samplesIn_ptr + (2 * n)] + samplesIn[samplesIn_ptr + (2 * n) + 1];
buf[n] = (short)Inlines.silk_RSHIFT_ROUND(sum, 1);
}
ret += Resampler.silk_resampler(
psEnc.state_Fxx[0].resampler_state,
psEnc.state_Fxx[0].inputBuf,
psEnc.state_Fxx[0].inputBufIx + 2,
buf,
0,
nSamplesFromInput);
/* On the first mono frame, average the results for the two resampler states */
if (psEnc.nPrevChannelsInternal == 2 && psEnc.state_Fxx[0].nFramesEncoded == 0)
{
ret += Resampler.silk_resampler(
psEnc.state_Fxx[1].resampler_state,
psEnc.state_Fxx[1].inputBuf,
psEnc.state_Fxx[1].inputBufIx + 2,
buf,
0,
nSamplesFromInput);
for (n = 0; n < psEnc.state_Fxx[0].frame_length; n++)
{
psEnc.state_Fxx[0].inputBuf[psEnc.state_Fxx[0].inputBufIx + n + 2] =
(short)(Inlines.silk_RSHIFT(psEnc.state_Fxx[0].inputBuf[psEnc.state_Fxx[0].inputBufIx + n + 2]
+ psEnc.state_Fxx[1].inputBuf[psEnc.state_Fxx[1].inputBufIx + n + 2], 1));
}
}
psEnc.state_Fxx[0].inputBufIx += nSamplesToBuffer;
}
else
{
Inlines.OpusAssert(encControl.nChannelsAPI == 1 && encControl.nChannelsInternal == 1);
Array.Copy(samplesIn, samplesIn_ptr, buf, 0, nSamplesFromInput);
ret += Resampler.silk_resampler(
psEnc.state_Fxx[0].resampler_state,
psEnc.state_Fxx[0].inputBuf,
psEnc.state_Fxx[0].inputBufIx + 2,
buf,
0,
nSamplesFromInput);
psEnc.state_Fxx[0].inputBufIx += nSamplesToBuffer;
}
samplesIn_ptr += (nSamplesFromInput * encControl.nChannelsAPI);
nSamplesIn -= nSamplesFromInput;
/* Default */
psEnc.allowBandwidthSwitch = 0;
/* Silk encoder */
if (psEnc.state_Fxx[0].inputBufIx >= psEnc.state_Fxx[0].frame_length)
{
/* Enough data in input buffer, so encode */
Inlines.OpusAssert(psEnc.state_Fxx[0].inputBufIx == psEnc.state_Fxx[0].frame_length);
Inlines.OpusAssert(encControl.nChannelsInternal == 1 || psEnc.state_Fxx[1].inputBufIx == psEnc.state_Fxx[1].frame_length);
/* Deal with LBRR data */
if (psEnc.state_Fxx[0].nFramesEncoded == 0 && prefillFlag == 0)
{
/* Create space at start of payload for VAD and FEC flags */
byte[] iCDF = { 0, 0 };
iCDF[0] = (byte)(256 - Inlines.silk_RSHIFT(256, (psEnc.state_Fxx[0].nFramesPerPacket + 1) * encControl.nChannelsInternal));
psRangeEnc.enc_icdf(0, iCDF, 8);
/* Encode any LBRR data from previous packet */
/* Encode LBRR flags */
for (n = 0; n < encControl.nChannelsInternal; n++)
{
LBRR_symbol = 0;
for (i = 0; i < psEnc.state_Fxx[n].nFramesPerPacket; i++)
{
LBRR_symbol |= Inlines.silk_LSHIFT(psEnc.state_Fxx[n].LBRR_flags[i], i);
}
psEnc.state_Fxx[n].LBRR_flag = (sbyte)(LBRR_symbol > 0 ? 1 : 0);
if (LBRR_symbol != 0 && psEnc.state_Fxx[n].nFramesPerPacket > 1)
{
psRangeEnc.enc_icdf(LBRR_symbol - 1, Tables.silk_LBRR_flags_iCDF_ptr[psEnc.state_Fxx[n].nFramesPerPacket - 2], 8);
}
}
/* Code LBRR indices and excitation signals */
for (i = 0; i < psEnc.state_Fxx[0].nFramesPerPacket; i++)
{
for (n = 0; n < encControl.nChannelsInternal; n++)
{
if (psEnc.state_Fxx[n].LBRR_flags[i] != 0)
{
int condCoding;
if (encControl.nChannelsInternal == 2 && n == 0)
{
Stereo.silk_stereo_encode_pred(psRangeEnc, psEnc.sStereo.predIx[i]);
/* For LBRR data there's no need to code the mid-only flag if the side-channel LBRR flag is set */
if (psEnc.state_Fxx[1].LBRR_flags[i] == 0)
{
Stereo.silk_stereo_encode_mid_only(psRangeEnc, psEnc.sStereo.mid_only_flags[i]);
}
}
/* Use conditional coding if previous frame available */
if (i > 0 && psEnc.state_Fxx[n].LBRR_flags[i - 1] != 0)
{
condCoding = SilkConstants.CODE_CONDITIONALLY;
}
else
{
condCoding = SilkConstants.CODE_INDEPENDENTLY;
}
EncodeIndices.silk_encode_indices(psEnc.state_Fxx[n], psRangeEnc, i, 1, condCoding);
EncodePulses.silk_encode_pulses(psRangeEnc, psEnc.state_Fxx[n].indices_LBRR[i].signalType, psEnc.state_Fxx[n].indices_LBRR[i].quantOffsetType,
psEnc.state_Fxx[n].pulses_LBRR[i], psEnc.state_Fxx[n].frame_length);
}
}
}
/* Reset LBRR flags */
for (n = 0; n < encControl.nChannelsInternal; n++)
{
Arrays.MemSetInt(psEnc.state_Fxx[n].LBRR_flags, 0, SilkConstants.MAX_FRAMES_PER_PACKET);
}
psEnc.nBitsUsedLBRR = psRangeEnc.tell();
}
HPVariableCutoff.silk_HP_variable_cutoff(psEnc.state_Fxx);
/* Total target bits for packet */
nBits = Inlines.silk_DIV32_16(Inlines.silk_MUL(encControl.bitRate, encControl.payloadSize_ms), 1000);
/* Subtract bits used for LBRR */
if (prefillFlag == 0)
{
nBits -= psEnc.nBitsUsedLBRR;
}
/* Divide by number of uncoded frames left in packet */
nBits = Inlines.silk_DIV32_16(nBits, psEnc.state_Fxx[0].nFramesPerPacket);
/* Convert to bits/second */
if (encControl.payloadSize_ms == 10)
{
TargetRate_bps = Inlines.silk_SMULBB(nBits, 100);
}
else
{
TargetRate_bps = Inlines.silk_SMULBB(nBits, 50);
}
/* Subtract fraction of bits in excess of target in previous frames and packets */
TargetRate_bps -= Inlines.silk_DIV32_16(Inlines.silk_MUL(psEnc.nBitsExceeded, 1000), TuningParameters.BITRESERVOIR_DECAY_TIME_MS);
if (prefillFlag == 0 && psEnc.state_Fxx[0].nFramesEncoded > 0)
{
/* Compare actual vs target bits so far in this packet */
int bitsBalance = psRangeEnc.tell() - psEnc.nBitsUsedLBRR - nBits * psEnc.state_Fxx[0].nFramesEncoded;
TargetRate_bps -= Inlines.silk_DIV32_16(Inlines.silk_MUL(bitsBalance, 1000), TuningParameters.BITRESERVOIR_DECAY_TIME_MS);
}
/* Never exceed input bitrate */
TargetRate_bps = Inlines.silk_LIMIT(TargetRate_bps, encControl.bitRate, 5000);
/* Convert Left/Right to Mid/Side */
if (encControl.nChannelsInternal == 2)
{
BoxedValueSbyte midOnlyFlagBoxed = new BoxedValueSbyte(psEnc.sStereo.mid_only_flags[psEnc.state_Fxx[0].nFramesEncoded]);
Stereo.silk_stereo_LR_to_MS(psEnc.sStereo,
psEnc.state_Fxx[0].inputBuf,
2,
psEnc.state_Fxx[1].inputBuf,
2,
psEnc.sStereo.predIx[psEnc.state_Fxx[0].nFramesEncoded],
midOnlyFlagBoxed,
MStargetRates_bps,
TargetRate_bps,
psEnc.state_Fxx[0].speech_activity_Q8,
encControl.toMono,
psEnc.state_Fxx[0].fs_kHz,
psEnc.state_Fxx[0].frame_length);
psEnc.sStereo.mid_only_flags[psEnc.state_Fxx[0].nFramesEncoded] = midOnlyFlagBoxed.Val;
if (midOnlyFlagBoxed.Val == 0)
{
/* Reset side channel encoder memory for first frame with side coding */
if (psEnc.prev_decode_only_middle == 1)
{
psEnc.state_Fxx[1].sShape.Reset();
psEnc.state_Fxx[1].sPrefilt.Reset();
psEnc.state_Fxx[1].sNSQ.Reset();
Arrays.MemSetShort(psEnc.state_Fxx[1].prev_NLSFq_Q15, 0, SilkConstants.MAX_LPC_ORDER);
Arrays.MemSetInt(psEnc.state_Fxx[1].sLP.In_LP_State, 0, 2);
psEnc.state_Fxx[1].prevLag = 100;
psEnc.state_Fxx[1].sNSQ.lagPrev = 100;
psEnc.state_Fxx[1].sShape.LastGainIndex = 10;
psEnc.state_Fxx[1].prevSignalType = SilkConstants.TYPE_NO_VOICE_ACTIVITY;
psEnc.state_Fxx[1].sNSQ.prev_gain_Q16 = 65536;
psEnc.state_Fxx[1].first_frame_after_reset = 1;
}
psEnc.state_Fxx[1].silk_encode_do_VAD();
}
else
{
psEnc.state_Fxx[1].VAD_flags[psEnc.state_Fxx[0].nFramesEncoded] = 0;
}
if (prefillFlag == 0)
{
Stereo.silk_stereo_encode_pred(psRangeEnc, psEnc.sStereo.predIx[psEnc.state_Fxx[0].nFramesEncoded]);
if (psEnc.state_Fxx[1].VAD_flags[psEnc.state_Fxx[0].nFramesEncoded] == 0)
{
Stereo.silk_stereo_encode_mid_only(psRangeEnc, psEnc.sStereo.mid_only_flags[psEnc.state_Fxx[0].nFramesEncoded]);
}
}
}
else
{
/* Buffering */
Array.Copy(psEnc.sStereo.sMid, psEnc.state_Fxx[0].inputBuf, 2);
Array.Copy(psEnc.state_Fxx[0].inputBuf, psEnc.state_Fxx[0].frame_length, psEnc.sStereo.sMid, 0, 2);
}
psEnc.state_Fxx[0].silk_encode_do_VAD();
/* Encode */
for (n = 0; n < encControl.nChannelsInternal; n++)
{
int maxBits, useCBR;
/* Handling rate constraints */
maxBits = encControl.maxBits;
if (tot_blocks == 2 && curr_block == 0)
{
maxBits = maxBits * 3 / 5;
}
else if (tot_blocks == 3)
{
if (curr_block == 0)
{
maxBits = maxBits * 2 / 5;
}
else if (curr_block == 1)
{
maxBits = maxBits * 3 / 4;
}
}
useCBR = (encControl.useCBR != 0 && curr_block == tot_blocks - 1) ? 1 : 0;
if (encControl.nChannelsInternal == 1)
{
channelRate_bps = TargetRate_bps;
}
else
{
channelRate_bps = MStargetRates_bps[n];
if (n == 0 && MStargetRates_bps[1] > 0)
{
useCBR = 0;
/* Give mid up to 1/2 of the max bits for that frame */
maxBits -= encControl.maxBits / (tot_blocks * 2);
}
}
if (channelRate_bps > 0)
{
int condCoding;
psEnc.state_Fxx[n].silk_control_SNR(channelRate_bps);
/* Use independent coding if no previous frame available */
if (psEnc.state_Fxx[0].nFramesEncoded - n <= 0)
{
condCoding = SilkConstants.CODE_INDEPENDENTLY;
}
else if (n > 0 && psEnc.prev_decode_only_middle != 0)
{
/* If we skipped a side frame in this packet, we don't
* need LTP scaling; the LTP state is well-defined. */
condCoding = SilkConstants.CODE_INDEPENDENTLY_NO_LTP_SCALING;
}
else
{
condCoding = SilkConstants.CODE_CONDITIONALLY;
}
ret += psEnc.state_Fxx[n].silk_encode_frame(nBytesOut, psRangeEnc, condCoding, maxBits, useCBR);
Inlines.OpusAssert(ret == SilkError.SILK_NO_ERROR);
}
psEnc.state_Fxx[n].controlled_since_last_payload = 0;
psEnc.state_Fxx[n].inputBufIx = 0;
psEnc.state_Fxx[n].nFramesEncoded++;
}
psEnc.prev_decode_only_middle = psEnc.sStereo.mid_only_flags[psEnc.state_Fxx[0].nFramesEncoded - 1];
/* Insert VAD and FEC flags at beginning of bitstream */
if (nBytesOut.Val > 0 && psEnc.state_Fxx[0].nFramesEncoded == psEnc.state_Fxx[0].nFramesPerPacket)
{
flags = 0;
for (n = 0; n < encControl.nChannelsInternal; n++)
{
for (i = 0; i < psEnc.state_Fxx[n].nFramesPerPacket; i++)
{
flags = Inlines.silk_LSHIFT(flags, 1);
flags |= (int)psEnc.state_Fxx[n].VAD_flags[i];
}
flags = Inlines.silk_LSHIFT(flags, 1);
flags |= (int)psEnc.state_Fxx[n].LBRR_flag;
}
if (prefillFlag == 0)
{
psRangeEnc.enc_patch_initial_bits((uint)flags, (uint)((psEnc.state_Fxx[0].nFramesPerPacket + 1) * encControl.nChannelsInternal));
}
/* Return zero bytes if all channels DTXed */
if (psEnc.state_Fxx[0].inDTX != 0 && (encControl.nChannelsInternal == 1 || psEnc.state_Fxx[1].inDTX != 0))
{
nBytesOut.Val = 0;
}
psEnc.nBitsExceeded += nBytesOut.Val * 8;
psEnc.nBitsExceeded -= Inlines.silk_DIV32_16(Inlines.silk_MUL(encControl.bitRate, encControl.payloadSize_ms), 1000);
psEnc.nBitsExceeded = Inlines.silk_LIMIT(psEnc.nBitsExceeded, 0, 10000);
/* Update flag indicating if bandwidth switching is allowed */
speech_act_thr_for_switch_Q8 = Inlines.silk_SMLAWB(((int)((TuningParameters.SPEECH_ACTIVITY_DTX_THRES) * ((long)1 << (8)) + 0.5)) /*Inlines.SILK_CONST(TuningParameters.SPEECH_ACTIVITY_DTX_THRES, 8)*/,
((int)(((1 - TuningParameters.SPEECH_ACTIVITY_DTX_THRES) / TuningParameters.MAX_BANDWIDTH_SWITCH_DELAY_MS) * ((long)1 << (16 + 8)) + 0.5)) /*Inlines.SILK_CONST((1 - TuningParameters.SPEECH_ACTIVITY_DTX_THRES) / TuningParameters.MAX_BANDWIDTH_SWITCH_DELAY_MS, 16 + 8)*/,
psEnc.timeSinceSwitchAllowed_ms);
if (psEnc.state_Fxx[0].speech_activity_Q8 < speech_act_thr_for_switch_Q8)
{
psEnc.allowBandwidthSwitch = 1;
psEnc.timeSinceSwitchAllowed_ms = 0;
}
else
{
psEnc.allowBandwidthSwitch = 0;
psEnc.timeSinceSwitchAllowed_ms += encControl.payloadSize_ms;
}
}
if (nSamplesIn == 0)
{
break;
}
}
else
{
break;
}
curr_block++;
}
psEnc.nPrevChannelsInternal = encControl.nChannelsInternal;
encControl.allowBandwidthSwitch = psEnc.allowBandwidthSwitch;
encControl.inWBmodeWithoutVariableLP = (psEnc.state_Fxx[0].fs_kHz == 16 && psEnc.state_Fxx[0].sLP.mode == 0) ? 1 : 0;
encControl.internalSampleRate = Inlines.silk_SMULBB(psEnc.state_Fxx[0].fs_kHz, 1000);
encControl.stereoWidth_Q14 = encControl.toMono != 0 ? 0 : psEnc.sStereo.smth_width_Q14;
if (prefillFlag != 0)
{
encControl.payloadSize_ms = tmp_payloadSize_ms;
encControl.complexity = tmp_complexity;
for (n = 0; n < encControl.nChannelsInternal; n++)
{
psEnc.state_Fxx[n].controlled_since_last_payload = 0;
psEnc.state_Fxx[n].prefillFlag = 0;
}
}
return(ret);
}
/* Decode a frame */
internal static int silk_Decode( /* O Returns error code */
SilkDecoder psDec, /* I/O State */
DecControlState decControl, /* I/O Control Structure */
int lostFlag, /* I 0: no loss, 1 loss, 2 decode fec */
int newPacketFlag, /* I Indicates first decoder call for this packet */
EntropyCoder psRangeDec, /* I/O Compressor data structure */
short[] samplesOut, /* O Decoded output speech vector */
int samplesOut_ptr,
out int nSamplesOut /* O Number of samples decoded */
)
{
int i, n, decode_only_middle = 0, ret = SilkError.SILK_NO_ERROR;
int LBRR_symbol;
BoxedValueInt nSamplesOutDec = new BoxedValueInt();
short[] samplesOut_tmp;
int[] samplesOut_tmp_ptrs = new int[2];
short[] samplesOut1_tmp_storage1;
short[] samplesOut1_tmp_storage2;
short[] samplesOut2_tmp;
int[] MS_pred_Q13 = new int[] { 0, 0 };
short[] resample_out;
int resample_out_ptr;
SilkChannelDecoder[] channel_state = psDec.channel_state;
int has_side;
int stereo_to_mono;
int delay_stack_alloc;
nSamplesOut = 0;
Inlines.OpusAssert(decControl.nChannelsInternal == 1 || decControl.nChannelsInternal == 2);
/**********************************/
/* Test if first frame in payload */
/**********************************/
if (newPacketFlag != 0)
{
for (n = 0; n < decControl.nChannelsInternal; n++)
{
channel_state[n].nFramesDecoded = 0; /* Used to count frames in packet */
}
}
/* If Mono . Stereo transition in bitstream: init state of second channel */
if (decControl.nChannelsInternal > psDec.nChannelsInternal)
{
ret += channel_state[1].silk_init_decoder();
}
stereo_to_mono = (decControl.nChannelsInternal == 1 && psDec.nChannelsInternal == 2 &&
(decControl.internalSampleRate == 1000 * channel_state[0].fs_kHz)) ? 1 : 0;
if (channel_state[0].nFramesDecoded == 0)
{
for (n = 0; n < decControl.nChannelsInternal; n++)
{
int fs_kHz_dec;
if (decControl.payloadSize_ms == 0)
{
/* Assuming packet loss, use 10 ms */
channel_state[n].nFramesPerPacket = 1;
channel_state[n].nb_subfr = 2;
}
else if (decControl.payloadSize_ms == 10)
{
channel_state[n].nFramesPerPacket = 1;
channel_state[n].nb_subfr = 2;
}
else if (decControl.payloadSize_ms == 20)
{
channel_state[n].nFramesPerPacket = 1;
channel_state[n].nb_subfr = 4;
}
else if (decControl.payloadSize_ms == 40)
{
channel_state[n].nFramesPerPacket = 2;
channel_state[n].nb_subfr = 4;
}
else if (decControl.payloadSize_ms == 60)
{
channel_state[n].nFramesPerPacket = 3;
channel_state[n].nb_subfr = 4;
}
else
{
Inlines.OpusAssert(false);
return(SilkError.SILK_DEC_INVALID_FRAME_SIZE);
}
fs_kHz_dec = (decControl.internalSampleRate >> 10) + 1;
if (fs_kHz_dec != 8 && fs_kHz_dec != 12 && fs_kHz_dec != 16)
{
Inlines.OpusAssert(false);
return(SilkError.SILK_DEC_INVALID_SAMPLING_FREQUENCY);
}
ret += channel_state[n].silk_decoder_set_fs(fs_kHz_dec, decControl.API_sampleRate);
}
}
if (decControl.nChannelsAPI == 2 && decControl.nChannelsInternal == 2 && (psDec.nChannelsAPI == 1 || psDec.nChannelsInternal == 1))
{
Arrays.MemSetShort(psDec.sStereo.pred_prev_Q13, 0, 2);
Arrays.MemSetShort(psDec.sStereo.sSide, 0, 2);
channel_state[1].resampler_state.Assign(channel_state[0].resampler_state);
}
psDec.nChannelsAPI = decControl.nChannelsAPI;
psDec.nChannelsInternal = decControl.nChannelsInternal;
if (decControl.API_sampleRate > (int)SilkConstants.MAX_API_FS_KHZ * 1000 || decControl.API_sampleRate < 8000)
{
ret = SilkError.SILK_DEC_INVALID_SAMPLING_FREQUENCY;
return(ret);
}
if (lostFlag != DecoderAPIFlag.FLAG_PACKET_LOST && channel_state[0].nFramesDecoded == 0)
{
/* First decoder call for this payload */
/* Decode VAD flags and LBRR flag */
for (n = 0; n < decControl.nChannelsInternal; n++)
{
for (i = 0; i < channel_state[n].nFramesPerPacket; i++)
{
channel_state[n].VAD_flags[i] = psRangeDec.dec_bit_logp(1);
}
channel_state[n].LBRR_flag = psRangeDec.dec_bit_logp(1);
}
/* Decode LBRR flags */
for (n = 0; n < decControl.nChannelsInternal; n++)
{
Arrays.MemSetInt(channel_state[n].LBRR_flags, 0, SilkConstants.MAX_FRAMES_PER_PACKET);
if (channel_state[n].LBRR_flag != 0)
{
if (channel_state[n].nFramesPerPacket == 1)
{
channel_state[n].LBRR_flags[0] = 1;
}
else
{
LBRR_symbol = psRangeDec.dec_icdf(Tables.silk_LBRR_flags_iCDF_ptr[channel_state[n].nFramesPerPacket - 2], 8) + 1;
for (i = 0; i < channel_state[n].nFramesPerPacket; i++)
{
channel_state[n].LBRR_flags[i] = Inlines.silk_RSHIFT(LBRR_symbol, i) & 1;
}
}
}
}
if (lostFlag == DecoderAPIFlag.FLAG_DECODE_NORMAL)
{
/* Regular decoding: skip all LBRR data */
for (i = 0; i < channel_state[0].nFramesPerPacket; i++)
{
for (n = 0; n < decControl.nChannelsInternal; n++)
{
if (channel_state[n].LBRR_flags[i] != 0)
{
short[] pulses = new short[SilkConstants.MAX_FRAME_LENGTH];
int condCoding;
if (decControl.nChannelsInternal == 2 && n == 0)
{
Stereo.silk_stereo_decode_pred(psRangeDec, MS_pred_Q13);
if (channel_state[1].LBRR_flags[i] == 0)
{
BoxedValueInt decodeOnlyMiddleBoxed = new BoxedValueInt(decode_only_middle);
Stereo.silk_stereo_decode_mid_only(psRangeDec, decodeOnlyMiddleBoxed);
decode_only_middle = decodeOnlyMiddleBoxed.Val;
}
}
/* Use conditional coding if previous frame available */
if (i > 0 && (channel_state[n].LBRR_flags[i - 1] != 0))
{
condCoding = SilkConstants.CODE_CONDITIONALLY;
}
else
{
condCoding = SilkConstants.CODE_INDEPENDENTLY;
}
DecodeIndices.silk_decode_indices(channel_state[n], psRangeDec, i, 1, condCoding);
DecodePulses.silk_decode_pulses(psRangeDec, pulses, channel_state[n].indices.signalType,
channel_state[n].indices.quantOffsetType, channel_state[n].frame_length);
}
}
}
}
}
/* Get MS predictor index */
if (decControl.nChannelsInternal == 2)
{
if (lostFlag == DecoderAPIFlag.FLAG_DECODE_NORMAL ||
(lostFlag == DecoderAPIFlag.FLAG_DECODE_LBRR && channel_state[0].LBRR_flags[channel_state[0].nFramesDecoded] == 1))
{
Stereo.silk_stereo_decode_pred(psRangeDec, MS_pred_Q13);
/* For LBRR data, decode mid-only flag only if side-channel's LBRR flag is false */
if ((lostFlag == DecoderAPIFlag.FLAG_DECODE_NORMAL && channel_state[1].VAD_flags[channel_state[0].nFramesDecoded] == 0) ||
(lostFlag == DecoderAPIFlag.FLAG_DECODE_LBRR && channel_state[1].LBRR_flags[channel_state[0].nFramesDecoded] == 0))
{
BoxedValueInt decodeOnlyMiddleBoxed = new BoxedValueInt(decode_only_middle);
Stereo.silk_stereo_decode_mid_only(psRangeDec, decodeOnlyMiddleBoxed);
decode_only_middle = decodeOnlyMiddleBoxed.Val;
}
else
{
decode_only_middle = 0;
}
}
else
{
for (n = 0; n < 2; n++)
{
MS_pred_Q13[n] = psDec.sStereo.pred_prev_Q13[n];
}
}
}
/* Reset side channel decoder prediction memory for first frame with side coding */
if (decControl.nChannelsInternal == 2 && decode_only_middle == 0 && psDec.prev_decode_only_middle == 1)
{
Arrays.MemSetShort(psDec.channel_state[1].outBuf, 0, SilkConstants.MAX_FRAME_LENGTH + 2 * SilkConstants.MAX_SUB_FRAME_LENGTH);
Arrays.MemSetInt(psDec.channel_state[1].sLPC_Q14_buf, 0, SilkConstants.MAX_LPC_ORDER);
psDec.channel_state[1].lagPrev = 100;
psDec.channel_state[1].LastGainIndex = 10;
psDec.channel_state[1].prevSignalType = SilkConstants.TYPE_NO_VOICE_ACTIVITY;
psDec.channel_state[1].first_frame_after_reset = 1;
}
/* Check if the temp buffer fits into the output PCM buffer. If it fits,
* we can delay allocating the temp buffer until after the SILK peak stack
* usage. We need to use a < and not a <= because of the two extra samples. */
delay_stack_alloc = (decControl.internalSampleRate * decControl.nChannelsInternal
< decControl.API_sampleRate * decControl.nChannelsAPI) ? 1 : 0;
if (delay_stack_alloc != 0)
{
samplesOut_tmp = samplesOut;
samplesOut_tmp_ptrs[0] = samplesOut_ptr;
samplesOut_tmp_ptrs[1] = samplesOut_ptr + channel_state[0].frame_length + 2;
}
else
{
samplesOut1_tmp_storage1 = new short[decControl.nChannelsInternal * (channel_state[0].frame_length + 2)];
samplesOut_tmp = samplesOut1_tmp_storage1;
samplesOut_tmp_ptrs[0] = 0;
samplesOut_tmp_ptrs[1] = channel_state[0].frame_length + 2;
}
if (lostFlag == DecoderAPIFlag.FLAG_DECODE_NORMAL)
{
has_side = (decode_only_middle == 0) ? 1 : 0;
}
else
{
has_side = (psDec.prev_decode_only_middle == 0 ||
(decControl.nChannelsInternal == 2 &&
lostFlag == DecoderAPIFlag.FLAG_DECODE_LBRR &&
channel_state[1].LBRR_flags[channel_state[1].nFramesDecoded] == 1)) ? 1 : 0;
}
/* Call decoder for one frame */
for (n = 0; n < decControl.nChannelsInternal; n++)
{
if (n == 0 || (has_side != 0))
{
int FrameIndex;
int condCoding;
FrameIndex = channel_state[0].nFramesDecoded - n;
/* Use independent coding if no previous frame available */
if (FrameIndex <= 0)
{
condCoding = SilkConstants.CODE_INDEPENDENTLY;
}
else if (lostFlag == DecoderAPIFlag.FLAG_DECODE_LBRR)
{
condCoding = (channel_state[n].LBRR_flags[FrameIndex - 1] != 0) ? SilkConstants.CODE_CONDITIONALLY : SilkConstants.CODE_INDEPENDENTLY;
}
else if (n > 0 && (psDec.prev_decode_only_middle != 0))
{
/* If we skipped a side frame in this packet, we don't
* need LTP scaling; the LTP state is well-defined. */
condCoding = SilkConstants.CODE_INDEPENDENTLY_NO_LTP_SCALING;
}
else
{
condCoding = SilkConstants.CODE_CONDITIONALLY;
}
ret += channel_state[n].silk_decode_frame(psRangeDec, samplesOut_tmp, samplesOut_tmp_ptrs[n] + 2, nSamplesOutDec, lostFlag, condCoding);
}
else
{
Arrays.MemSetWithOffset <short>(samplesOut_tmp, 0, samplesOut_tmp_ptrs[n] + 2, nSamplesOutDec.Val);
}
channel_state[n].nFramesDecoded++;
}
if (decControl.nChannelsAPI == 2 && decControl.nChannelsInternal == 2)
{
/* Convert Mid/Side to Left/Right */
Stereo.silk_stereo_MS_to_LR(psDec.sStereo, samplesOut_tmp, samplesOut_tmp_ptrs[0], samplesOut_tmp, samplesOut_tmp_ptrs[1], MS_pred_Q13, channel_state[0].fs_kHz, nSamplesOutDec.Val);
}
else
{
/* Buffering */
Array.Copy(psDec.sStereo.sMid, 0, samplesOut_tmp, samplesOut_tmp_ptrs[0], 2);
Array.Copy(samplesOut_tmp, samplesOut_tmp_ptrs[0] + nSamplesOutDec.Val, psDec.sStereo.sMid, 0, 2);
}
/* Number of output samples */
nSamplesOut = Inlines.silk_DIV32(nSamplesOutDec.Val * decControl.API_sampleRate, Inlines.silk_SMULBB(channel_state[0].fs_kHz, 1000));
/* Set up pointers to temp buffers */
if (decControl.nChannelsAPI == 2)
{
samplesOut2_tmp = new short[nSamplesOut];
resample_out = samplesOut2_tmp;
resample_out_ptr = 0;
}
else
{
resample_out = samplesOut;
resample_out_ptr = samplesOut_ptr;
}
if (delay_stack_alloc != 0)
{
samplesOut1_tmp_storage2 = new short[decControl.nChannelsInternal * (channel_state[0].frame_length + 2)];
Array.Copy(samplesOut, samplesOut_ptr, samplesOut1_tmp_storage2, 0, decControl.nChannelsInternal * (channel_state[0].frame_length + 2));
samplesOut_tmp = samplesOut1_tmp_storage2;
samplesOut_tmp_ptrs[0] = 0;
samplesOut_tmp_ptrs[1] = channel_state[0].frame_length + 2;
}
for (n = 0; n < Inlines.silk_min(decControl.nChannelsAPI, decControl.nChannelsInternal); n++)
{
/* Resample decoded signal to API_sampleRate */
ret += Resampler.silk_resampler(channel_state[n].resampler_state, resample_out, resample_out_ptr, samplesOut_tmp, samplesOut_tmp_ptrs[n] + 1, nSamplesOutDec.Val);
/* Interleave if stereo output and stereo stream */
if (decControl.nChannelsAPI == 2)
{
int nptr = samplesOut_ptr + n;
for (i = 0; i < nSamplesOut; i++)
{
samplesOut[nptr + 2 * i] = resample_out[resample_out_ptr + i];
}
}
}
/* Create two channel output from mono stream */
if (decControl.nChannelsAPI == 2 && decControl.nChannelsInternal == 1)
{
if (stereo_to_mono != 0)
{
/* Resample right channel for newly collapsed stereo just in case
* we weren't doing collapsing when switching to mono */
ret += Resampler.silk_resampler(channel_state[1].resampler_state, resample_out, resample_out_ptr, samplesOut_tmp, samplesOut_tmp_ptrs[0] + 1, nSamplesOutDec.Val);
for (i = 0; i < nSamplesOut; i++)
{
samplesOut[samplesOut_ptr + 1 + 2 * i] = resample_out[resample_out_ptr + i];
}
}
else
{
for (i = 0; i < nSamplesOut; i++)
{
samplesOut[samplesOut_ptr + 1 + 2 * i] = samplesOut[samplesOut_ptr + 2 * i];
}
}
}
/* Export pitch lag, measured at 48 kHz sampling rate */
if (channel_state[0].prevSignalType == SilkConstants.TYPE_VOICED)
{
int[] mult_tab = { 6, 4, 3 };
decControl.prevPitchLag = channel_state[0].lagPrev * mult_tab[(channel_state[0].fs_kHz - 8) >> 2];
}
else
{
decControl.prevPitchLag = 0;
}
if (lostFlag == DecoderAPIFlag.FLAG_PACKET_LOST)
{
/* On packet loss, remove the gain clamping to prevent having the energy "bounce back"
* if we lose packets when the energy is going down */
for (i = 0; i < psDec.nChannelsInternal; i++)
{
psDec.channel_state[i].LastGainIndex = 10;
}
}
else
{
psDec.prev_decode_only_middle = decode_only_middle;
}
return(ret);
}
