REVISION
Definition in file CODEC_playHdl.c.
#include "../SYSTEM/sys_hardware.h"
#include "../SYSTEM/sys_hiddenfeature.h"
#include "codec.h"
#include "codec_assert.h"
#include "../AUDIO/audio.h"
#include "../FMXMIT/xmit_inc.h"
Include dependency graph for CODEC_playHdl.c:
Go to the source code of this file.
Functions | |
| void | CODEC_playPcmPut (int *pcmBuf, int len) |
| Function serves as the normal playback PCM data put handle. | |
| void | CODEC_myfiPcmPut (int *pcmBuf, int len) |
| Function serves as the MyFi playback PCM data put handle. | |
| void | CODEC_linkitPcmPut (int *pcmBuf, int len) |
| Function serves as the linkIt mode PCM data put handle. | |
| void | CODEC_linkitPcmGet (int *pcmBuf, int len) |
| Function serves as the linkIt mode PCM data get handle. | |
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Function serves as the linkIt mode PCM data get handle.
Definition at line 367 of file CODEC_playHdl.c. Referenced by AUDIO_start(). 00368 {
00369 leftChannel = pcmBuf;
00370 rightChannel = pcmBuf + len/2;
00371 }
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Function serves as the linkIt mode PCM data put handle.
Definition at line 305 of file CODEC_playHdl.c. Referenced by AUDIO_start(). 00306 {
00307 int i;
00308
00309 /*if(SYS_HF_myfiTest())
00310 {
00311 st_AUDIO_OSCILLATOR LeftTestTone;
00312 st_AUDIO_OSCILLATOR RightTestTone;
00313
00314 // set up oscillators for the test tones
00315 // 500 Hz, 95% full-scale sine wave (to prevent clipping due to
00316 // rounding)
00317 LeftTestTone.Coeff =
00318 RightTestTone.Coeff = 32753;
00319 LeftTestTone.Tap1 =
00320 RightTestTone.Tap1 = 887;
00321 LeftTestTone.Tap2 =
00322 RightTestTone.Tap2 = 0;
00323
00324 LeftTestTone.pOutput = leftChannel;
00325 RightTestTone.pOutput = rightChannel;
00326 AUDIO_toneGen(&LeftTestTone);
00327 AUDIO_toneGen(&RightTestTone);
00328 }*/
00329
00330 StereoMod.pSum = leftChannel;
00331 StereoMod.pDiff = rightChannel;
00332 StereoMod.OutputLeft = (int *)pcmBuf;
00333 StereoMod.OutputRight = (int *)pcmBuf + 64;
00334
00335 //if(SYS_HF_myfiTest())
00336 {
00337 FIRPremphasisLeft.pOutput = leftChannel;
00338 FIRPremphasisRight.pOutput = rightChannel;
00339 }
00340
00341 for (i = 0; i < PCM_CHANNEL_BUFLEN; i++)
00342 {
00343 //XMIT_Energy(&EnergyInputLeft);
00344 //XMIT_Energy(&EnergyInputRight);
00345
00346 CarrierGen(&Carrier);
00347
00348 //if(SYS_HF_myfiTest())
00349 {
00350 XMIT_preemphasisFIR(&FIRPremphasisLeft);
00351 XMIT_preemphasisFIR(&FIRPremphasisRight);
00352 }
00353
00354 XMIT_Modulator(&StereoMod);
00355 }
00356 }
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Function serves as the MyFi playback PCM data put handle.
Definition at line 144 of file CODEC_playHdl.c. References st_PCM::decode, st_PCM::flag, GEN_zeroOut(), and st_PCM::sampleRate. Referenced by AUDIO_start(). 00145 {
00146 int i;
00147 int * pData;
00148 int * pDstLeft;
00149 int * pDstRight;
00150
00151 if (XMIT_CompressBuffer( &EncoderBuffer ))
00152 // if the input buffer has less than the required number of samples
00153 // and the output buffer has less than 64 samples ready to go, we need
00154 // to decode more MP3 and perform SRC on it.
00155 // For the input buffer, there are always 11 extra samples left over
00156 // from the previous frame for the taps of the FIR filter.
00157 {
00158 pDstLeft = EncoderBuffer.pSumInput;
00159 pDstRight = EncoderBuffer.pDiffInput;
00160
00161 /* Setup sample rate conversion output buffer. */
00162 SRC_setOutput(&LeftSRC, pDstLeft);
00163 SRC_setOutput(&RightSRC, pDstRight);
00164
00165 /* check for a change in sample rate and reset SRC if needed */
00166 if (Pcm.flag&SAMPLE_RATE_CHANGED)
00167 {
00168 /* set sample rate */
00169 SRC_setInputRate(&LeftSRC, Pcm.sampleRate);
00170 SRC_setInputRate(&RightSRC, Pcm.sampleRate);
00171 PCM_samplerateNotChanged();
00172 }
00173
00174 /* loop over sample rate converter until output buffers are filled */
00175 while (SRC_isOutputBufferFull(&LeftSRC) == 0)
00176 {
00177 /* get another input block if needed */
00178 if (SRC_isInputBufferEmpty(&LeftSRC))
00179 {
00180 pData = Pcm.decode();
00181
00182 if( NULL == pData)
00183 {
00184 // Decoder data is not ready,
00185 // Zero sending out.
00186 GEN_zeroOut((UINT16*)pcmBuf, len);
00187 return;
00188 }
00189 else
00190 {
00191 SRC_setInput(&LeftSRC, pData);
00192 SRC_setInput(&RightSRC, pData + SRC_inputBufferSize());
00193 }
00194 }
00195
00196 /* run the left and right channels */
00197 SRC_apply(&LeftSRC);
00198 SRC_apply(&RightSRC);
00199 }
00200
00201 EncoderBuffer.Samples += PCM_CHANNEL_BUFLEN;
00202 }
00203
00204 // by the time we get here, we should have at least 26 samples (+11 previous samples)
00205 // available in the input buffers.
00206 DifSRCData.Input = EncoderBuffer.pDiffOutput;
00207 SumSRCData.Input = EncoderBuffer.pSumOutput;
00208 DifSRCData.Output = SRCBuffer.pDiffInput;
00209 SumSRCData.Output = SRCBuffer.pSumInput;
00210
00211 while (SRCBuffer.Samples < PCM_CHANNEL_BUFLEN)
00212 /* If there are not enough samples after the SRC, then
00213 * we need to run the SRC until there are enough samples */
00214 {
00215 /* SRC Automatically updates the input and output pointers. */
00216 XMIT_SRC(&SumSRCData);
00217 XMIT_SRC(&DifSRCData);
00218 SRCBuffer.Samples += XMIT_OUTPUT;
00219 EncoderBuffer.Samples -= XMIT_SAMPLES;
00220 }
00221
00222 /* Update pointers */
00223 EncoderBuffer.pDiffOutput = DifSRCData.Input;
00224 EncoderBuffer.pSumOutput = SumSRCData.Input;
00225 SRCBuffer.pDiffInput = DifSRCData.Output;
00226 SRCBuffer.pSumInput = SumSRCData.Output;
00227 /* Now, enough up-sampled data is in the SRC Output buffers.
00228 * we can not process the data and put it in the output buffer.
00229 * Then, we need to compress the SRC Output Buffers. */
00230 EnergyInputLeft.pInput = StereoMod.pSum = SRCBuffer.pSumOutput;
00231 EnergyInputRight.pInput = StereoMod.pDiff = SRCBuffer.pDiffOutput;
00232
00233 SRCBuffer.pSumOutput += PCM_CHANNEL_BUFLEN;
00234 SRCBuffer.pDiffOutput += PCM_CHANNEL_BUFLEN;
00235 SRCBuffer.Samples -= PCM_CHANNEL_BUFLEN;
00236
00237 pDstLeft = pcmBuf;
00238 pDstRight = pcmBuf + len/2;
00239
00240 //' Now, let's take this data and put it into the EncoderBuffers
00241 //' Apply all filters (15kHz and pre-emphasis here)
00242 //' Also apply all summing of the left and right channels
00243 //' and the difference of the left and right channels.
00244 StereoMod.OutputRight = pDstRight;
00245 StereoMod.OutputLeft = pDstLeft;
00246
00247 #ifdef MIXER_SUPPORT
00248 {
00249 int ii,
00250 *pL,
00251 *pR;
00252
00253 pL = StereoMod.pDiff;
00254 pR = StereoMod.pSum;
00255 for (ii = 0; ii < PCM_CHANNEL_BUFLEN; ii++)
00256 {
00257 *pL = (*pL+*leftChannel)/2;
00258 *pR = (*pR+*rightChannel)/2;
00259 pL++;
00260 pR++;
00261 leftChannel++;
00262 rightChannel++;
00263 }
00264 }
00265 #endif
00266
00267
00268 for (i = 0; i < PCM_CHANNEL_BUFLEN; i++)
00269 {
00270 // Not enought mips for this.
00271 //XMIT_Energy(&EnergyInputLeft);
00272 //XMIT_Energy(&EnergyInputRight);
00273 #if 0 // 11/13/02
00274 FIRPremphasisLeftTaps[0] = *pSrcLeft;
00275 FIRPremphasisRightTaps[0] = *pSrcRight;
00276 *pSrcLeft++ = XMIT_PreemphasisFIR(&FIRPremphasisLeft);
00277 *pSrcRight++ = XMIT_PreemphasisFIR(&FIRPremphasisRight);
00278 #endif
00279
00280 CarrierGen(&Carrier);
00281 //XMIT_SumDiff(&StereoMod);
00282 XMIT_Modulator(&StereoMod);
00283 }
00284
00285 XMIT_CompressBuffer(&SRCBuffer);
00286
00287 /*remove the EQ apply for MIPs limit*/
00288 /* Let's apply EQ here. */
00289 /*
00290 EQ_apply(&LeftEQ, pDstLeft, pDstLeft);
00291 EQ_apply(&RightEQ, pDstRight, pDstRight);
00292 */
00293 }
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Function serves as the normal playback PCM data put handle.
Definition at line 50 of file CODEC_playHdl.c. References AUDIO_speedIsOutputBufferFull(), st_PCM::decode, st_PCM::flag, GEN_zeroOut(), and st_PCM::sampleRate. Referenced by AUDIO_start(). 00051 {
00052 int * pData;
00053 int * pRight = pcmBuf+len/2;
00054
00055 #ifdef DJ_SUPPORT
00056 // loop over speed control.
00057 while(1)
00058 #endif
00059 {
00060 #ifdef DJ_SUPPORT
00061 if(AUDIO_speedIsOutputBufferFull()) break;
00062 #endif
00063
00064 /* Setup sample rate conversion output buffer. */
00065 // This could be opted out in the future, to have PCM put to a buffer at fixed
00066 // position.
00067 SRC_setOutput(&LeftSRC, pcmBuf);
00068 SRC_setOutput(&RightSRC, pRight);
00069
00070 /* check for a change in sample rate and reset SRC if needed */
00071 if (Pcm.flag&SAMPLE_RATE_CHANGED)
00072 {
00073 /* set sample rate */
00074 SRC_setInputRate(&LeftSRC, Pcm.sampleRate);
00075 SRC_setInputRate(&RightSRC, Pcm.sampleRate);
00076 PCM_samplerateNotChanged();
00077 }
00078
00079 /* loop over sample rate converter until output buffers are filled */
00080 while (SRC_isOutputBufferFull(&LeftSRC) == 0)
00081 {
00082 /* get another input block if needed */
00083 if (SRC_isInputBufferEmpty(&LeftSRC))
00084 {
00085 pData = Pcm.decode();
00086
00087 if( NULL == pData)
00088 {
00089 // Decoder data is not ready,
00090 // Zero sending out.
00091 GEN_zeroOut((UINT16*)pcmBuf, len);
00092 return;
00093 }
00094 else
00095 {
00096 SRC_setInput(&LeftSRC, pData);
00097 SRC_setInput(&RightSRC, pData + SRC_inputBufferSize());
00098 }
00099 }
00100
00101 /* run the left and right channels */
00102 SRC_apply(&LeftSRC);
00103 SRC_apply(&RightSRC);
00104 }
00105
00106 #ifdef DJ_SUPPORT
00107 if(AUDIO_speedFIR(pcmBuf, pRight)) break;
00108 #endif
00109
00110 }
00111
00112 #ifdef DJ_SUPPORT
00113 AUDIO_speedGetSample(pcmBuf, pRight);
00114 #endif
00115
00116 EQ_apply(&LeftEQ, pcmBuf, pcmBuf);
00117 EQ_apply(&RightEQ, pRight, pRight);
00118
00119 #ifdef MIXER_SUPPORT
00120 {
00121 unsigned short ii;
00122 for (ii = 0; ii < PCM_CHANNEL_BUFLEN; ii++)
00123 {
00124 *pcmBuf = (*pcmBuf+*leftChannel)/2;
00125 *pRight = (*pRight+*rightChannel)/2;
00126 pcmBuf++;
00127 pRight++;
00128 leftChannel++;
00129 rightChannel++;
00130 }
00131 }
00132 #endif
00133 }
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1.3.9.1