From 20a68c6ab36f42c37a3aa6b9fa223a7837a5733b Mon Sep 17 00:00:00 2001 From: Chip Audette Date: Sun, 19 Feb 2017 09:42:34 -0500 Subject: [PATCH] Revise inputI2S and outputI2S for block size --- input_i2s_f32.cpp | 53 ++++++++++++----- input_i2s_f32.h | 10 +++- output_i2s_f32.cpp | 140 +++++++++++++++++++++++++++++++++++++++------ output_i2s_f32.h | 15 ++++- 4 files changed, 181 insertions(+), 37 deletions(-) diff --git a/input_i2s_f32.cpp b/input_i2s_f32.cpp index 7b62d67..54c9ebd 100644 --- a/input_i2s_f32.cpp +++ b/input_i2s_f32.cpp @@ -36,6 +36,11 @@ bool AudioInputI2S_F32::update_responsibility = false; DMAChannel AudioInputI2S_F32::dma(false); +float AudioInputI2S_F32::sample_rate_Hz = AUDIO_SAMPLE_RATE; +int AudioInputI2S_F32::audio_block_samples = AUDIO_BLOCK_SAMPLES; + +#define I2S_BUFFER_TO_USE_BYTES (AudioOutputI2S_F32::audio_block_samples*sizeof(i2s_rx_buffer[0])) + void AudioInputI2S_F32::begin(void) { dma.begin(true); // Allocate the DMA channel first @@ -44,6 +49,8 @@ void AudioInputI2S_F32::begin(void) //block_right_1st = NULL; // TODO: should we set & clear the I2S_RCSR_SR bit here? + AudioOutputI2S_F32::sample_rate_Hz = sample_rate_Hz; + AudioOutputI2S_F32::audio_block_samples = audio_block_samples; AudioOutputI2S_F32::config_i2s(); CORE_PIN13_CONFIG = PORT_PCR_MUX(4); // pin 13, PTC5, I2S0_RXD0 @@ -55,9 +62,12 @@ void AudioInputI2S_F32::begin(void) dma.TCD->SLAST = 0; dma.TCD->DADDR = i2s_rx_buffer; dma.TCD->DOFF = 2; - dma.TCD->CITER_ELINKNO = sizeof(i2s_rx_buffer) / 2; - dma.TCD->DLASTSGA = -sizeof(i2s_rx_buffer); - dma.TCD->BITER_ELINKNO = sizeof(i2s_rx_buffer) / 2; + //dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; //original + dma.TCD->CITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2; + //dma.TCD->DLASTSGA = -sizeof(i2s_rx_buffer); //original + dma.TCD->DLASTSGA = -I2S_BUFFER_TO_USE_BYTES; + //dma.TCD->BITER_ELINKNO = sizeof(i2s_rx_buffer) / 2; //original + dma.TCD->BITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2; dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR; #endif dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_RX); @@ -67,7 +77,9 @@ void AudioInputI2S_F32::begin(void) I2S0_RCSR |= I2S_RCSR_RE | I2S_RCSR_BCE | I2S_RCSR_FRDE | I2S_RCSR_FR; I2S0_TCSR |= I2S_TCSR_TE | I2S_TCSR_BCE; // TX clock enable, because sync'd to TX dma.attachInterrupt(isr); -} + + +}; void AudioInputI2S_F32::isr(void) { @@ -82,26 +94,33 @@ void AudioInputI2S_F32::isr(void) #endif dma.clearInterrupt(); - if (daddr < (uint32_t)i2s_rx_buffer + sizeof(i2s_rx_buffer) / 2) { + //if (daddr < (uint32_t)i2s_rx_buffer + sizeof(i2s_rx_buffer) / 2) { + if (daddr < (uint32_t)i2s_rx_buffer + I2S_BUFFER_TO_USE_BYTES / 2) { + // DMA is receiving to the first half of the buffer // need to remove data from the second half - src = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES/2]; - end = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES]; + //src = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES/2]; //original + //end = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES]; //original + src = (int16_t *)&i2s_rx_buffer[audio_block_samples/2]; + end = (int16_t *)&i2s_rx_buffer[audio_block_samples]; if (AudioInputI2S_F32::update_responsibility) AudioStream_F32::update_all(); } else { // DMA is receiving to the second half of the buffer // need to remove data from the first half src = (int16_t *)&i2s_rx_buffer[0]; - end = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES/2]; + //end = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES/2]; //original + end = (int16_t *)&i2s_rx_buffer[audio_block_samples/2]; } left = AudioInputI2S_F32::block_left; right = AudioInputI2S_F32::block_right; if (left != NULL && right != NULL) { offset = AudioInputI2S_F32::block_offset; - if (offset <= AUDIO_BLOCK_SAMPLES/2) { + //if (offset <= AUDIO_BLOCK_SAMPLES/2) { //original + if (offset <= ((uint32_t) audio_block_samples/2)) { dest_left = &(left->data[offset]); dest_right = &(right->data[offset]); - AudioInputI2S_F32::block_offset = offset + AUDIO_BLOCK_SAMPLES/2; + //AudioInputI2S_F32::block_offset = offset + AUDIO_BLOCK_SAMPLES/2; //original + AudioInputI2S_F32::block_offset = offset + audio_block_samples/2; do { //n = *src++; //*dest_left++ = (int16_t)n; @@ -114,7 +133,10 @@ void AudioInputI2S_F32::isr(void) //digitalWriteFast(3, LOW); } - +#define I16_TO_F32_NORM_FACTOR (3.051757812500000E-05) //which is 1/32768 +void AudioInputI2S_F32::convert_i16_to_f32( int16_t *p_i16, float32_t *p_f32, int len) { + for (int i=0; i= AUDIO_BLOCK_SAMPLES) { + //if (block_offset >= AUDIO_BLOCK_SAMPLES) { //original + if (block_offset >= audio_block_samples) { // the DMA filled 2 blocks, so grab them and get the // 2 new blocks to the DMA, as quickly as possible out_left = block_left; @@ -153,9 +176,9 @@ void AudioInputI2S_F32::update(void) } } if (out_left_f32 != NULL) { - //convert to f32 - arm_q15_to_float((q15_t *)out_left->data, (float32_t *)out_left_f32->data, AUDIO_BLOCK_SAMPLES); - arm_q15_to_float((q15_t *)out_right->data, (float32_t *)out_right_f32->data, AUDIO_BLOCK_SAMPLES); + //convert int16 to float 32 + convert_i16_to_f32(out_left->data, out_left_f32->data, audio_block_samples); + convert_i16_to_f32(out_right->data, out_right_f32->data, audio_block_samples); //transmit the f32 data! AudioStream_F32::transmit(out_left_f32,0); diff --git a/input_i2s_f32.h b/input_i2s_f32.h index 9720f33..b60e5ae 100644 --- a/input_i2s_f32.h +++ b/input_i2s_f32.h @@ -36,9 +36,15 @@ class AudioInputI2S_F32 : public AudioStream_F32 { //GUI: inputs:0, outputs:2 //this line used for automatic generation of GUI nodes public: - AudioInputI2S_F32(void) : AudioStream_F32(0, NULL) { begin(); } + AudioInputI2S_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) { + sample_rate_Hz = settings.sample_rate_Hz; + audio_block_samples = settings.audio_block_samples; + begin(); + } virtual void update(void); + static void convert_i16_to_f32( int16_t *p_i16, float32_t *p_f32, int len) ; void begin(void); + //friend class AudioOutputI2S_F32; protected: AudioInputI2S_F32(int dummy): AudioStream_F32(0, NULL) {} // to be used only inside AudioInputI2Sslave !! static bool update_responsibility; @@ -47,6 +53,8 @@ protected: private: static audio_block_t *block_left; static audio_block_t *block_right; + static float sample_rate_Hz; + static int audio_block_samples; static uint16_t block_offset; }; diff --git a/output_i2s_f32.cpp b/output_i2s_f32.cpp index 97fa342..1ff85f2 100644 --- a/output_i2s_f32.cpp +++ b/output_i2s_f32.cpp @@ -25,9 +25,54 @@ */ #include "output_i2s_f32.h" -#include "memcpy_audio.h" +//#include "input_i2s_f32.h" +//include "memcpy_audio.h" +//#include "memcpy_interleave.h" #include + +//Here's the function to change the sample rate of the system (via changing the clocking of the I2S bus) +//https://forum.pjrc.com/threads/38753-Discussion-about-a-simple-way-to-change-the-sample-rate?p=121365&viewfull=1#post121365 +float setI2SFreq(const float freq_Hz) { + int freq = (int)freq_Hz; + typedef struct { + uint8_t mult; + uint16_t div; + } __attribute__((__packed__)) tmclk; + + const int numfreqs = 16; + const int samplefreqs[numfreqs] = { 2000, 8000, 11025, 16000, 22050, 24000, 32000, 44100, 44117.64706 , 48000, 88200, 44117.64706 * 2, 96000, 176400, 44117.64706 * 4, 192000}; + +#if (F_PLL==16000000) + const tmclk clkArr[numfreqs] = {{4, 125}, {16, 125}, {148, 839}, {32, 125}, {145, 411}, {48, 125}, {64, 125}, {151, 214}, {12, 17}, {96, 125}, {151, 107}, {24, 17}, {192, 125}, {127, 45}, {48, 17}, {255, 83} }; +#elif (F_PLL==72000000) + const tmclk clkArr[numfreqs] = {{832, 1125}, {32, 1125}, {49, 1250}, {64, 1125}, {49, 625}, {32, 375}, {128, 1125}, {98, 625}, {8, 51}, {64, 375}, {196, 625}, {16, 51}, {128, 375}, {249, 397}, {32, 51}, {185, 271} }; +#elif (F_PLL==96000000) + const tmclk clkArr[numfreqs] = {{2, 375},{8, 375}, {73, 2483}, {16, 375}, {147, 2500}, {8, 125}, {32, 375}, {147, 1250}, {2, 17}, {16, 125}, {147, 625}, {4, 17}, {32, 125}, {151, 321}, {8, 17}, {64, 125} }; +#elif (F_PLL==120000000) + const tmclk clkArr[numfreqs] = {{8, 1875},{32, 1875}, {89, 3784}, {64, 1875}, {147, 3125}, {32, 625}, {128, 1875}, {205, 2179}, {8, 85}, {64, 625}, {89, 473}, {16, 85}, {128, 625}, {178, 473}, {32, 85}, {145, 354} }; +#elif (F_PLL==144000000) + const tmclk clkArr[numfreqs] = {{4, 1125},{16, 1125}, {49, 2500}, {32, 1125}, {49, 1250}, {16, 375}, {64, 1125}, {49, 625}, {4, 51}, {32, 375}, {98, 625}, {8, 51}, {64, 375}, {196, 625}, {16, 51}, {128, 375} }; +#elif (F_PLL==180000000) + const tmclk clkArr[numfreqs] = {{23, 8086}, {46, 4043}, {49, 3125}, {73, 3208}, {98, 3125}, {37, 1084}, {183, 4021}, {196, 3125}, {16, 255}, {128, 1875}, {107, 853}, {32, 255}, {219, 1604}, {214, 853}, {64, 255}, {219, 802} }; +#elif (F_PLL==192000000) + const tmclk clkArr[numfreqs] = {{1, 375}, {4, 375}, {37, 2517}, {8, 375}, {73, 2483}, {4, 125}, {16, 375}, {147, 2500}, {1, 17}, {8, 125}, {147, 1250}, {2, 17}, {16, 125}, {147, 625}, {4, 17}, {32, 125} }; +#elif (F_PLL==216000000) + const tmclk clkArr[numfreqs] = {{8, 3375}, {32, 3375}, {49, 3750}, {64, 3375}, {49, 1875}, {32, 1125}, {128, 3375}, {98, 1875}, {8, 153}, {64, 1125}, {196, 1875}, {16, 153}, {128, 1125}, {226, 1081}, {32, 153}, {147, 646} }; +#elif (F_PLL==240000000) + const tmclk clkArr[numfreqs] = {{4, 1875}, {16, 1875}, {29, 2466}, {32, 1875}, {89, 3784}, {16, 625}, {64, 1875}, {147, 3125}, {4, 85}, {32, 625}, {205, 2179}, {8, 85}, {64, 625}, {89, 473}, {16, 85}, {128, 625} }; +#endif + + for (int f = 0; f < numfreqs; f++) { + if ( freq == samplefreqs[f] ) { + while (I2S0_MCR & I2S_MCR_DUF) ; + I2S0_MDR = I2S_MDR_FRACT((clkArr[f].mult - 1)) | I2S_MDR_DIVIDE((clkArr[f].div - 1)); + return (float)(F_PLL / 256 * clkArr[f].mult / clkArr[f].div); + } + } + return 0.0f; +} + audio_block_t * AudioOutputI2S_F32::block_left_1st = NULL; audio_block_t * AudioOutputI2S_F32::block_right_1st = NULL; audio_block_t * AudioOutputI2S_F32::block_left_2nd = NULL; @@ -35,9 +80,15 @@ audio_block_t * AudioOutputI2S_F32::block_right_2nd = NULL; uint16_t AudioOutputI2S_F32::block_left_offset = 0; uint16_t AudioOutputI2S_F32::block_right_offset = 0; bool AudioOutputI2S_F32::update_responsibility = false; -DMAMEM static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES]; +DMAMEM static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES]; //local audio_block_samples should be no larger than global AUDIO_BLOCK_SAMPLES DMAChannel AudioOutputI2S_F32::dma(false); +float AudioOutputI2S_F32::sample_rate_Hz = AUDIO_SAMPLE_RATE; +int AudioOutputI2S_F32::audio_block_samples = AUDIO_BLOCK_SAMPLES; + +#define I2S_BUFFER_TO_USE_BYTES (AudioOutputI2S_F32::audio_block_samples*sizeof(i2s_tx_buffer[0])) + + void AudioOutputI2S_F32::begin(void) { dma.begin(true); // Allocate the DMA channel first @@ -47,6 +98,7 @@ void AudioOutputI2S_F32::begin(void) // TODO: should we set & clear the I2S_TCSR_SR bit here? config_i2s(); + CORE_PIN22_CONFIG = PORT_PCR_MUX(6); // pin 22, PTC1, I2S0_TXD0 #if defined(KINETISK) @@ -54,12 +106,15 @@ void AudioOutputI2S_F32::begin(void) dma.TCD->SOFF = 2; dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1); dma.TCD->NBYTES_MLNO = 2; - dma.TCD->SLAST = -sizeof(i2s_tx_buffer); + //dma.TCD->SLAST = -sizeof(i2s_tx_buffer); //original + dma.TCD->SLAST = -I2S_BUFFER_TO_USE_BYTES; dma.TCD->DADDR = &I2S0_TDR0; dma.TCD->DOFF = 0; - dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; + //dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; //original + dma.TCD->CITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2; dma.TCD->DLASTSGA = 0; - dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; + //dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; //original + dma.TCD->BITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2; dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR; #endif dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_TX); @@ -69,6 +124,13 @@ void AudioOutputI2S_F32::begin(void) I2S0_TCSR = I2S_TCSR_SR; I2S0_TCSR = I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE; dma.attachInterrupt(isr); + + // change the I2S frequencies to make the requested sample rate + setI2SFreq(AudioOutputI2S_F32::sample_rate_Hz); + + enabled = 1; + + //AudioInputI2S_F32::begin_guts(); } @@ -81,10 +143,12 @@ void AudioOutputI2S_F32::isr(void) saddr = (uint32_t)(dma.TCD->SADDR); dma.clearInterrupt(); - if (saddr < (uint32_t)i2s_tx_buffer + sizeof(i2s_tx_buffer) / 2) { + //if (saddr < (uint32_t)i2s_tx_buffer + sizeof(i2s_tx_buffer) / 2) { //original + if (saddr < (uint32_t)i2s_tx_buffer + I2S_BUFFER_TO_USE_BYTES / 2) { // DMA is transmitting the first half of the buffer // so we must fill the second half - dest = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2]; + //dest = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2]; //original + dest = (int16_t *)&i2s_tx_buffer[audio_block_samples/2]; if (AudioOutputI2S_F32::update_responsibility) AudioStream_F32::update_all(); } else { // DMA is transmitting the second half of the buffer @@ -97,6 +161,7 @@ void AudioOutputI2S_F32::isr(void) offsetL = AudioOutputI2S_F32::block_left_offset; offsetR = AudioOutputI2S_F32::block_right_offset; + /* Original if (blockL && blockR) { memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR); offsetL += AUDIO_BLOCK_SAMPLES / 2; @@ -111,8 +176,34 @@ void AudioOutputI2S_F32::isr(void) memset(dest,0,AUDIO_BLOCK_SAMPLES * 2); return; } + */ + + int16_t *d = dest; + if (blockL && blockR) { + //memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR); + //memcpy_tointerleaveLRwLen(dest, blockL->data + offsetL, blockR->data + offsetR, audio_block_samples/2); + int16_t *pL = blockL->data + offsetL; + int16_t *pR = blockR->data + offsetR; + for (int i=0; i < audio_block_samples/2; i++) { *d++ = *pL++; *d++ = *pR++; } //interleave + offsetL += audio_block_samples / 2; + offsetR += audio_block_samples / 2; + } else if (blockL) { + //memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR); + int16_t *pL = blockL->data + offsetL; + for (int i=0; i < audio_block_samples / 2 * 2; i+=2) { *(d+i) = *pL++; } //interleave + offsetL += audio_block_samples / 2; + } else if (blockR) { + int16_t *pR = blockR->data + offsetR; + for (int i=0; i < audio_block_samples /2 * 2; i+=2) { *(d+i) = *pR++; } //interleave + offsetR += audio_block_samples / 2; + } else { + //memset(dest,0,AUDIO_BLOCK_SAMPLES * 2); + memset(dest,0,audio_block_samples * 2); + return; + } - if (offsetL < AUDIO_BLOCK_SAMPLES) { + //if (offsetL < AUDIO_BLOCK_SAMPLES) { //original + if (offsetL < (uint16_t)audio_block_samples) { AudioOutputI2S_F32::block_left_offset = offsetL; } else { AudioOutputI2S_F32::block_left_offset = 0; @@ -120,7 +211,8 @@ void AudioOutputI2S_F32::isr(void) AudioOutputI2S_F32::block_left_1st = AudioOutputI2S_F32::block_left_2nd; AudioOutputI2S_F32::block_left_2nd = NULL; } - if (offsetR < AUDIO_BLOCK_SAMPLES) { + //if (offsetR < AUDIO_BLOCK_SAMPLES) { + if (offsetR < (uint16_t)audio_block_samples) { AudioOutputI2S_F32::block_right_offset = offsetR; } else { AudioOutputI2S_F32::block_right_offset = 0; @@ -199,8 +291,9 @@ void AudioOutputI2S_F32::isr(void) #endif } - - +void AudioOutputI2S_F32::convert_f32_to_i16(float32_t *p_f32, int16_t *p_i16, int len) { + for (int i=0; ilength != audio_block_samples) { + Serial.print("AudioOutputI2S_F32: *** WARNING ***: audio_block says len = "); + Serial.print(block_f32->length); + Serial.print(", but I2S settings want it to be = "); + Serial.println(audio_block_samples); + } + //Serial.print("AudioOutputI2S_F32: audio_block_samples = "); + //Serial.println(audio_block_samples); //convert F32 to Int16 block = AudioStream::allocate(); - arm_float_to_q15((float32_t *)(block_f32->data),(q15_t *)(block->data), AUDIO_BLOCK_SAMPLES); + convert_f32_to_i16(block_f32->data, block->data, audio_block_samples); AudioStream_F32::release(block_f32); //now process the data blocks @@ -242,7 +343,7 @@ void AudioOutputI2S_F32::update(void) if (block_f32) { //convert F32 to Int16 block = AudioStream::allocate(); - arm_float_to_q15((float32_t *)(block_f32->data),(q15_t *)(block->data), AUDIO_BLOCK_SAMPLES); + convert_f32_to_i16(block_f32->data, block->data, audio_block_samples); AudioStream_F32::release(block_f32); __disable_irq(); @@ -305,11 +406,11 @@ void AudioOutputI2S_F32::update(void) #endif #ifndef MCLK_SRC -#if F_CPU >= 20000000 - #define MCLK_SRC 3 // the PLL -#else - #define MCLK_SRC 0 // system clock -#endif + #if (F_CPU >= 20000000) + #define MCLK_SRC 3 // the PLL + #else + #define MCLK_SRC 0 // system clock + #endif #endif void AudioOutputI2S_F32::config_i2s(void) @@ -432,4 +533,5 @@ void AudioOutputI2Sslave::config_i2s(void) CORE_PIN9_CONFIG = PORT_PCR_MUX(6); // pin 9, PTC3, I2S0_TX_BCLK CORE_PIN11_CONFIG = PORT_PCR_MUX(6); // pin 11, PTC6, I2S0_MCLK } -*/ \ No newline at end of file +*/ + diff --git a/output_i2s_f32.h b/output_i2s_f32.h index bed67a0..1f4f536 100644 --- a/output_i2s_f32.h +++ b/output_i2s_f32.h @@ -32,16 +32,24 @@ #include "AudioStream.h" #include "DMAChannel.h" + class AudioOutputI2S_F32 : public AudioStream_F32 { //GUI: inputs:2, outputs:0 //this line used for automatic generation of GUI node public: - AudioOutputI2S_F32(void) : AudioStream_F32(2, inputQueueArray) { begin(); } + AudioOutputI2S_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray) + { + sample_rate_Hz = settings.sample_rate_Hz; + audio_block_samples = settings.audio_block_samples; + begin(); + } virtual void update(void); void begin(void); friend class AudioInputI2S_F32; + static void convert_f32_to_i16( float32_t *p_f32, int16_t *p_i16, int len) ; + protected: - AudioOutputI2S_F32(int dummy): AudioStream_F32(2, inputQueueArray) {} // to be used only inside AudioOutputI2Sslave !! + //AudioOutputI2S_F32(const AudioSettings &settings): AudioStream_F32(2, inputQueueArray) {} // to be used only inside AudioOutputI2Sslave !! static void config_i2s(void); static audio_block_t *block_left_1st; static audio_block_t *block_right_1st; @@ -54,6 +62,9 @@ private: static uint16_t block_left_offset; static uint16_t block_right_offset; audio_block_f32_t *inputQueueArray[2]; + static float sample_rate_Hz; + static int audio_block_samples; + volatile uint8_t enabled = 1; };