boblark
2 years ago
parent
5a9be45938
commit
624f14283e
@ -0,0 +1,724 @@ |
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/* Extended from Audio Library for Teensy which is
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* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com |
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* |
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* Development of this audio library was funded by PJRC.COM, LLC by sales of |
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* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop |
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* open source software by purchasing Teensy or other PJRC products. |
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* |
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* Permission is hereby granted, free of charge, to any person obtaining a copy |
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* of this software and associated documentation files (the "Software"), to deal |
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* in the Software without restriction, including without limitation the rights |
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
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* copies of the Software, and to permit persons to whom the Software is |
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* furnished to do so, subject to the following conditions: |
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* |
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* The above copyright notice, development funding notice, and this permission |
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* notice shall be included in all copies or substantial portions of the Software. |
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* |
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN |
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* THE SOFTWARE. |
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* |
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* Extended by Chip Audette, OpenAudio, Dec 2019 |
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* Converted to F32 and to variable audio block length |
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* The F32 conversion is under the MIT License. Use at your own risk. |
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* |
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* Further extensions to sub multiple WAV sample rates are copyright |
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* (c) 2023 Bob Larkin under the MIT License. |
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*/ |
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#include <Arduino.h> |
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#include "AudioSDPlayer_F32.h" |
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#include "spi_interrupt.h" |
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#define STATE_DIRECT_8BIT_MONO 0 // playing mono at native sample rate
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#define STATE_DIRECT_8BIT_STEREO 1 // playing stereo at native sample rate
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#define STATE_DIRECT_16BIT_MONO 2 // playing mono at native sample rate
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#define STATE_DIRECT_16BIT_STEREO 3 // playing stereo at native sample rate
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#define STATE_CONVERT_8BIT_MONO 4 // playing mono, converting sample rate
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#define STATE_CONVERT_8BIT_STEREO 5 // playing stereo, converting sample rate
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#define STATE_CONVERT_16BIT_MONO 6 // playing mono, converting sample rate
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#define STATE_CONVERT_16BIT_STEREO 7 // playing stereo, converting sample rate
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#define STATE_PARSE1 8 // looking for 20 byte ID header
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#define STATE_PARSE2 9 // looking for 16 byte format header
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#define STATE_PARSE3 10 // looking for 8 byte data header
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#define STATE_PARSE4 11 // ignoring unknown chunk after "fmt "
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#define STATE_PARSE5 12 // ignoring unknown chunk before "fmt "
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#define STATE_PAUSED 13 |
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#define STATE_STOP 14 |
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void AudioSDPlayer_F32::begin(void) |
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{ |
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state = STATE_STOP; |
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state_play = STATE_STOP; |
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data_length = 0; |
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if (block_left_f32) { |
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AudioStream_F32::release(block_left_f32); |
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block_left_f32 = NULL; |
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} |
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if (block_right_f32) { |
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AudioStream_F32::release(block_right_f32); |
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block_right_f32 = NULL; |
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} |
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} |
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bool AudioSDPlayer_F32::play(const char *filename) |
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{ |
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stop(); |
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bool irq = false; |
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if (NVIC_IS_ENABLED(IRQ_SOFTWARE)) { |
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NVIC_DISABLE_IRQ(IRQ_SOFTWARE); |
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irq = true; |
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} |
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#if defined(HAS_KINETIS_SDHC) |
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if (!(SIM_SCGC3 & SIM_SCGC3_SDHC)) AudioStartUsingSPI(); |
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#else |
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AudioStartUsingSPI(); |
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#endif |
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wavfile = SD.open(filename); |
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if (!wavfile) { |
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#if defined(HAS_KINETIS_SDHC) |
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if (!(SIM_SCGC3 & SIM_SCGC3_SDHC)) AudioStopUsingSPI(); |
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#else |
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AudioStopUsingSPI(); |
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#endif |
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if (irq) NVIC_ENABLE_IRQ(IRQ_SOFTWARE); |
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return false; |
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} |
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buffer_length = 0; |
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buffer_offset = 0; |
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state_play = STATE_STOP; |
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data_length = 20; |
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header_offset = 0; |
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state = STATE_PARSE1; |
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if (irq) NVIC_ENABLE_IRQ(IRQ_SOFTWARE); |
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return true; |
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} |
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void AudioSDPlayer_F32::stop(void) |
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{ |
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bool irq = false; |
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if (NVIC_IS_ENABLED(IRQ_SOFTWARE)) { |
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NVIC_DISABLE_IRQ(IRQ_SOFTWARE); |
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irq = true; |
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} |
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if (state != STATE_STOP) { |
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audio_block_f32_t *b1 = block_left_f32; |
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block_left_f32 = NULL; |
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audio_block_f32_t *b2 = block_right_f32; |
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block_right_f32 = NULL; |
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state = STATE_STOP; |
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if (b1) AudioStream_F32::release(b1); |
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if (b2) AudioStream_F32::release(b2); |
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wavfile.close(); |
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#if defined(HAS_KINETIS_SDHC) |
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if (!(SIM_SCGC3 & SIM_SCGC3_SDHC)) AudioStopUsingSPI(); |
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#else |
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AudioStopUsingSPI(); |
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#endif |
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} |
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if (irq) NVIC_ENABLE_IRQ(IRQ_SOFTWARE); |
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} |
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void AudioSDPlayer_F32::togglePlayPause(void) { |
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// take no action if wave header is not parsed OR
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// state is explicitly STATE_STOP
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if(state_play >= 8 || state == STATE_STOP) return; |
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// toggle back and forth between state_play and STATE_PAUSED
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if(state == state_play) { |
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state = STATE_PAUSED; |
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} |
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else if(state == STATE_PAUSED) { |
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state = state_play; |
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} |
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} |
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void AudioSDPlayer_F32::update(void) |
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{ |
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int32_t n; |
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// only update if we're playing and not paused
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if (state == STATE_STOP || state == STATE_PAUSED) return; |
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// allocate the audio blocks to transmit
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block_left_f32 = AudioStream_F32::allocate_f32(); |
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if (block_left_f32 == NULL) return; |
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if (state < 8 && (state & 1) == 1) { |
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// if we're playing stereo, allocate another
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// block for the right channel output
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block_right_f32 = AudioStream_F32::allocate_f32(); |
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if (block_right_f32 == NULL) { |
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AudioStream_F32::release(block_left_f32); |
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return; |
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} |
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} else { |
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// if we're playing mono or just parsing
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// the WAV file header, no right-side block
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block_right_f32 = NULL; |
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} |
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block_offset = 0; |
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// is there buffered data?
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n = buffer_length - buffer_offset; |
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if (n > 0) { |
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// Have buffered data. consume(n) returns true if audio transmitted.
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if (consume(n)) return; // it was enough to transmit audio
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} |
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// we only get to this point when buffer[512] is empty
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if (state != STATE_STOP && wavfile.available()) { |
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// we can read more data from the file...
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readagain: |
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buffer_length = wavfile.read(buffer, 512); |
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if (buffer_length == 0) goto end; |
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buffer_offset = 0; |
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bool parsing = (state >= 8); |
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bool txok = consume(buffer_length); |
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if (txok) { |
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if (state != STATE_STOP) return; |
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} else { |
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if (state != STATE_STOP) { |
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if (parsing && state < 8) goto readagain; |
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else goto cleanup; |
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} |
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} |
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} |
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end: // end of file reached or other reason to stop
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wavfile.close(); |
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#if defined(HAS_KINETIS_SDHC) |
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if (!(SIM_SCGC3 & SIM_SCGC3_SDHC)) AudioStopUsingSPI(); |
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#else |
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AudioStopUsingSPI(); |
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#endif |
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state_play = STATE_STOP; |
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state = STATE_STOP; |
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cleanup: |
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if (block_left_f32) { |
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if (block_offset > 0) { |
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for (uint32_t i=block_offset; i < audio_block_samples; i++) { |
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block_left_f32->data[i] = 0.0f; |
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} |
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transmit(block_left_f32, 0); |
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if (state < 8 && (state & 1) == 0) { |
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transmit(block_left_f32, 1); |
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} |
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} |
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AudioStream_F32::release(block_left_f32); |
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block_left_f32 = NULL; |
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} |
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if (block_right_f32) { |
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if (block_offset > 0) { |
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for (uint32_t i=block_offset; i < audio_block_samples; i++) { |
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block_right_f32->data[i] = 0.0f; |
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} |
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transmit(block_right_f32, 1); |
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} |
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AudioStream_F32::release(block_right_f32); |
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block_right_f32 = NULL; |
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} |
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} |
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// Consume already buffered WAV file data. Returns true if audio transmitted.
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bool AudioSDPlayer_F32::consume(uint32_t size) { |
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uint32_t len; |
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uint8_t lsb, msb; |
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const uint8_t *p; |
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int16_t val_int16; |
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float32_t rateRatioF; |
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rateRatioF = (float32_t) pSampleSubMultiple->rateRatio; |
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p = buffer + buffer_offset; |
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start: |
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if (size == 0) return false; |
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/*
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Serial.print("AudioSDPlayer_F32 consume, "); |
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Serial.print("size = "); |
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Serial.print(size); |
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Serial.print(", buffer_offset = "); |
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Serial.print(buffer_offset); |
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Serial.print(", data_length = "); |
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Serial.print(data_length); |
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Serial.print(", space = "); |
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Serial.print((audio_block_samples - block_offset) * 2); |
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Serial.print(", state = "); |
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Serial.println(state); |
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*/ |
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switch (state) { |
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// parse wav file header, is this really a .wav file?
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case STATE_PARSE1: |
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len = data_length; |
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if (size < len) len = size; |
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memcpy((uint8_t *)header + header_offset, p, len); |
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header_offset += len; |
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buffer_offset += len; |
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data_length -= len; |
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if (data_length > 0) return false; |
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// parse the header...
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if (header[0] == 0x46464952 && header[2] == 0x45564157) |
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{ |
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//Serial.println("is wav file");
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if (header[3] == 0x20746D66) |
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{ |
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// "fmt " header
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if (header[4] < 16) |
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{ |
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// WAV "fmt " info must be at least 16 bytes
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break; |
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} |
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if (header[4] > sizeof(header)) |
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{ |
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// if such .wav files exist, increasing the
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// size of header[] should accomodate them...
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//Serial.println("WAVEFORMATEXTENSIBLE too long");
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break; |
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} |
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//Serial.println("header ok");
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header_offset = 0; |
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state = STATE_PARSE2; |
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} |
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else |
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{ |
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// first chuck is something other than "fmt "
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//Serial.print("skipping \"");
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//Serial.printf("\" (%08X), ", __builtin_bswap32(header[3]));
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//Serial.print(header[4]);
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//Serial.println(" bytes");
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header_offset = 12; |
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state = STATE_PARSE5; |
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} |
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p += len; |
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size -= len; |
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data_length = header[4]; |
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goto start; |
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} |
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//Serial.println("unknown WAV header");
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break; |
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// check & extract key audio parameters
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case STATE_PARSE2: |
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len = data_length; |
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if (size < len) len = size; |
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memcpy((uint8_t *)header + header_offset, p, len); |
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header_offset += len; |
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buffer_offset += len; |
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data_length -= len; |
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if (data_length > 0) return false; |
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if (parse_format()) |
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{ |
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//Serial.println("audio format ok");
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p += len; |
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size -= len; |
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data_length = 8; |
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header_offset = 0; |
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state = STATE_PARSE3; |
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goto start; |
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} |
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//Serial.println("unknown audio format");
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break; |
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// find the data chunk
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case STATE_PARSE3: // 10
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len = data_length; |
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if (size < len) len = size; |
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memcpy((uint8_t *)header + header_offset, p, len); |
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header_offset += len; |
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buffer_offset += len; |
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data_length -= len; |
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if (data_length > 0) return false; |
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p += len; |
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size -= len; |
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data_length = header[1]; |
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if (header[0] == 0x61746164) |
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{ |
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// TODO: verify offset in file is an even number
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// as required by WAV format. abort if odd. Code
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// below will depend upon this and fail if not even.
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leftover_bytes = 0; |
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state = state_play; |
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if (state & 1) |
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{ |
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// if we're going to start stereo
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// better allocate another output block
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block_right_f32 = AudioStream_F32::allocate_f32(); |
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if (!block_right_f32) return false; |
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} |
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total_length = data_length; |
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} |
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else |
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{ |
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state = STATE_PARSE4; |
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} |
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goto start; |
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// ignore any extra unknown chunks (title & artist info)
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case STATE_PARSE4: // 11
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if (size < data_length) |
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{ |
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data_length -= size; |
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buffer_offset += size; |
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return false; |
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} |
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p += data_length; |
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size -= data_length; |
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buffer_offset += data_length; |
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data_length = 8; |
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header_offset = 0; |
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state = STATE_PARSE3; |
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goto start; |
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// skip past "junk" data before "fmt " header
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case STATE_PARSE5: |
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len = data_length; |
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if (size < len) len = size; |
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buffer_offset += len; |
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data_length -= len; |
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if (data_length > 0) return false; |
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p += len; |
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size -= len; |
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data_length = 8; |
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state = STATE_PARSE1; |
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goto start; |
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// playing mono at native sample rate
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case STATE_DIRECT_8BIT_MONO: |
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return false; |
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// playing stereo at native sample rate
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case STATE_DIRECT_8BIT_STEREO: |
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return false; |
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// Playing Mono at native sample rate ****** 16-BIT MONO ******
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case STATE_DIRECT_16BIT_MONO: |
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if (size > data_length) // End of WAV file
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size = data_length; |
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data_length -= size; |
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while (1) |
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{ |
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if(zerosToSend > 0) |
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{ |
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block_left_f32->data[block_offset++] = 0.0f; // Zeros for interpolation
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zerosToSend--; |
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if (block_offset >= audio_block_samples) |
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{ |
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if(pSampleSubMultiple->numCoeffs > 1 && // i.e., using FIR
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pSampleSubMultiple->firBufferL ) |
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{ |
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arm_fir_f32(&fir_instL, block_left_f32->data, |
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block_left_f32->data, block_left_f32->length); |
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} |
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transmit(block_left_f32, 0); // Mono sends same to L&R
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transmit(block_left_f32, 1); |
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AudioStream_F32::release(block_left_f32); |
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block_left_f32 = NULL; |
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data_length += size; |
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buffer_offset = p - buffer; |
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if (block_right_f32) |
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AudioStream_F32::release(block_right_f32); |
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if (data_length == 0) |
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state = STATE_STOP; |
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return true; |
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} |
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} |
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else // Not zeros, but data
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{ |
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lsb = *p++; // Little endian
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msb = *p++; |
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size -= 2; // 2 bytes per word
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// Convert to F32
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val_int16 = (msb << 8) | lsb; |
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// Scale up by rateRatioF to account for zeros
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block_left_f32->data[block_offset++] = rateRatioF*((float)val_int16)/(32768.0); |
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// For interpolation, each data point is followed by 0.0f's
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zerosToSend = pSampleSubMultiple->rateRatio - 1; // 0, 1, 3, 7
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if (block_offset >= audio_block_samples) |
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{ |
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// The FIR update
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if(pSampleSubMultiple->numCoeffs > 1 && |
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pSampleSubMultiple->firBufferL ) |
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{ |
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arm_fir_f32(&fir_instL, block_left_f32->data, |
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block_left_f32->data, block_left_f32->length); |
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} |
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transmit(block_left_f32, 0); // Mono sends same to L&R
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transmit(block_left_f32, 1); |
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AudioStream_F32::release(block_left_f32); |
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block_left_f32 = NULL; |
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data_length += size; |
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buffer_offset = p - buffer; |
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if (block_right_f32) |
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AudioStream_F32::release(block_right_f32); |
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if (data_length == 0) |
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state = STATE_STOP; |
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return true; |
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} |
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} |
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} // End while(1)
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if (size == 0) |
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{ |
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if (data_length == 0) break; |
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return false; |
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} |
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// End of file reached
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if (block_offset > 0) |
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{ |
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// TODO: fill remainder of last block with zero and transmit
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} |
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state = STATE_STOP; |
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return false; |
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// Playing stereo at native sample rate ****** 16-BIT STEREO ******
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case STATE_DIRECT_16BIT_STEREO: |
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if (size > data_length) |
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size = data_length; |
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data_length -= size; |
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if (leftover_bytes) |
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{ |
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block_left_f32->data[block_offset] = header[0]; |
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//PAH fix problem with left+right channels being swapped
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//RSL Is this actually the CODEC L/R problem?
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leftover_bytes = 0; |
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// goto right16; // RSL What is the deal???
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} |
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while (1) { |
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if(zerosToSend > 0) |
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{ |
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block_left_f32->data[block_offset] = 0.0f; // Zeros for interpolation
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block_right_f32->data[block_offset++] = 0.0f; |
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zerosToSend--; |
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if (block_offset >= audio_block_samples) |
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{ |
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if(pSampleSubMultiple->numCoeffs > 1 && // i.e., using FIR
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pSampleSubMultiple->firBufferL) |
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{ |
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arm_fir_f32(&fir_instL, block_left_f32->data, |
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block_left_f32->data, block_left_f32->length); |
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arm_fir_f32(&fir_instR, block_right_f32->data, |
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block_right_f32->data, block_right_f32->length); |
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} |
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transmit(block_left_f32, 0); |
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transmit(block_right_f32, 1); |
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AudioStream_F32::release(block_left_f32); |
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block_left_f32 = NULL; |
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data_length += size; |
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buffer_offset = p - buffer; |
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if (block_right_f32) |
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AudioStream_F32::release(block_right_f32); |
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if (data_length == 0) |
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state = STATE_STOP; |
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return true; |
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} |
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} |
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else // Not zeros, but data
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{ |
||||
lsb = *p++; // Little endian
|
||||
msb = *p++; |
||||
size -= 2; |
||||
if (size == 0) |
||||
{ |
||||
if (data_length == 0) break; |
||||
header[0] = (msb << 8) | lsb; |
||||
leftover_bytes = 2; |
||||
return false; |
||||
} |
||||
val_int16 = (int16_t)((msb << 8) | lsb); |
||||
//convert from int16 to float32 spanning +/-1.0
|
||||
// Scale up by rateRatioF to account for zeros
|
||||
block_left_f32->data[block_offset] = rateRatioF*((float)val_int16)/(32768.0); |
||||
|
||||
// right16: See about 15 lines above
|
||||
lsb = *p++; |
||||
msb = *p++; |
||||
size -= 2; |
||||
val_int16 = (int16_t)((msb << 8) | lsb); |
||||
// Convert from int16 to float32 spanning +/-1.0
|
||||
// Scale up by rateRatioF to account for zeros
|
||||
block_right_f32->data[block_offset++] = rateRatioF*((float)val_int16)/(32768.0); |
||||
// For stereo, the number of zeros to send refers to
|
||||
// the number of *pairs* of zeros.
|
||||
// For interpolation, each data point is followed by 0.0f's
|
||||
zerosToSend = pSampleSubMultiple->rateRatio - 1; // 0, 1, 3, 7
|
||||
if (block_offset >= audio_block_samples) |
||||
{ |
||||
if(pSampleSubMultiple->numCoeffs > 1 && // i.e., using FIR
|
||||
pSampleSubMultiple->firBufferL ) |
||||
{ |
||||
arm_fir_f32(&fir_instL, block_left_f32->data, |
||||
block_left_f32->data, block_left_f32->length); |
||||
arm_fir_f32(&fir_instR, block_right_f32->data, |
||||
block_right_f32->data, block_right_f32->length); |
||||
} |
||||
transmit(block_left_f32, 0); |
||||
AudioStream_F32::release(block_left_f32); |
||||
block_left_f32 = NULL; |
||||
transmit(block_right_f32, 1); |
||||
AudioStream_F32::release(block_right_f32); |
||||
block_right_f32 = NULL; |
||||
|
||||
data_length += size; |
||||
buffer_offset = p - buffer; |
||||
if (data_length == 0) state = STATE_STOP; |
||||
return true; |
||||
} |
||||
if (size == 0) |
||||
{ |
||||
if (data_length == 0) break; |
||||
leftover_bytes = 0; |
||||
return false; |
||||
} |
||||
} // Sending data, not zeros
|
||||
// end of file reached
|
||||
} // End while(1)
|
||||
if (block_offset > 0) |
||||
{ |
||||
// TODO: fill remainder of last block with zero and transmit
|
||||
} |
||||
state = STATE_STOP; |
||||
return false; |
||||
|
||||
// playing mono, converting sample rate
|
||||
case STATE_CONVERT_8BIT_MONO : |
||||
return false; |
||||
|
||||
// playing stereo, converting sample rate
|
||||
case STATE_CONVERT_8BIT_STEREO: |
||||
return false; |
||||
|
||||
// playing mono, converting sample rate
|
||||
case STATE_CONVERT_16BIT_MONO: |
||||
return false; |
||||
|
||||
// playing stereo, converting sample rate
|
||||
case STATE_CONVERT_16BIT_STEREO: |
||||
return false; |
||||
|
||||
// ignore any extra data after playing
|
||||
// or anything following any error
|
||||
case STATE_STOP: |
||||
return false; |
||||
|
||||
// this is not supposed to happen!
|
||||
//default:
|
||||
//Serial.println("AudioSDPlayer_F32, unknown state");
|
||||
} |
||||
state_play = STATE_STOP; |
||||
state = STATE_STOP; |
||||
return false; |
||||
} |
||||
|
||||
|
||||
bool AudioSDPlayer_F32::parse_format(void) { |
||||
uint8_t num = 0; |
||||
uint16_t format; |
||||
uint16_t channels; |
||||
uint32_t rate, b2m; |
||||
uint16_t bits; |
||||
|
||||
format = header[0]; |
||||
currentWavData.audio_format = header[0]; // uint16_t
|
||||
//Serial.print(" format = ");
|
||||
//Serial.println(format);
|
||||
if (format != 1) return false; |
||||
|
||||
rate = header[1]; |
||||
currentWavData.sample_rate = header[1]; // uint32_t
|
||||
Serial.print("WAV file sample rate = "); Serial.println(rate); |
||||
|
||||
// b2m is used to determine playing time. We base it on the WAV
|
||||
// file meta data. It is allowed to be played at a different rate
|
||||
// but all we do is to make the info available via the
|
||||
// struct currentWavData The INO needs to deal with differences.
|
||||
// 4294967296000.0 = 2^32 * 1000
|
||||
b2m = (uint32_t)((double)4294967296000.0 / (double)rate); |
||||
|
||||
channels = header[0] >> 16; |
||||
currentWavData.num_channels = header[0] >> 16; // uint16_t
|
||||
//Serial.print(" channels = ");
|
||||
//Serial.println(channels);
|
||||
if (channels == 1) { } |
||||
else if (channels == 2) |
||||
{ |
||||
b2m >>= 1; // Divide b2m by 2
|
||||
num |= 1; |
||||
} |
||||
else |
||||
return false; |
||||
|
||||
bits = header[3] >> 16; |
||||
currentWavData.bits = header[3] >> 16; // uint16_t
|
||||
//Serial.print(" bits = ");
|
||||
//Serial.println(bits);
|
||||
if (bits == 8) { } |
||||
else if (bits == 16) |
||||
{ |
||||
b2m >>= 1; // Again divide b2m by 2
|
||||
num |= 2; |
||||
} |
||||
else {return false;} |
||||
|
||||
bytes2millis = b2m; // Transfer to global
|
||||
Serial.print(" bytes2millis = "); Serial.println(b2m); |
||||
// we're not checking the byte rate and block align fields
|
||||
// if they're not the expected values, all we could do is
|
||||
// return false. Do any real wav files have unexpected
|
||||
// values in these other fields?
|
||||
state_play = num; |
||||
return true; |
||||
} |
||||
|
||||
uint32_t AudioSDPlayer_F32::updateBytes2Millis(void) { |
||||
double b2m; |
||||
|
||||
//account for sample rate
|
||||
b2m = ((double)4294967296000.0 / ((double)sample_rate_Hz)); |
||||
//account for channels
|
||||
b2m = b2m / ((double)channels); |
||||
//account for bits per second
|
||||
if (bits == 16) |
||||
b2m = b2m / 2; |
||||
else if (bits == 24) |
||||
b2m = b2m / 3; //can we handle 24 bits? I don't think that we can.
|
||||
// if 8-bits, fall through
|
||||
return bytes2millis = (uint32_t)b2m; |
||||
} |
||||
|
||||
bool AudioSDPlayer_F32::isPlaying(void) { |
||||
uint8_t s = *(volatile uint8_t *)&state; |
||||
return (s < 8); |
||||
} |
||||
|
||||
bool AudioSDPlayer_F32::isPaused(void) { |
||||
uint8_t s = *(volatile uint8_t *)&state; |
||||
return (s == STATE_PAUSED); |
||||
} |
||||
|
||||
bool AudioSDPlayer_F32::isStopped(void) { |
||||
uint8_t s = *(volatile uint8_t *)&state; |
||||
return (s == STATE_STOP); |
||||
} |
||||
|
||||
uint32_t AudioSDPlayer_F32::positionMillis(void) { |
||||
uint8_t s = *(volatile uint8_t *)&state; |
||||
if (s >= 8 && s != STATE_PAUSED) return 0; |
||||
uint32_t tlength = *(volatile uint32_t *)&total_length; |
||||
uint32_t dlength = *(volatile uint32_t *)&data_length; |
||||
uint32_t offset = tlength - dlength; |
||||
uint32_t b2m = *(volatile uint32_t *)&bytes2millis; |
||||
return ((uint64_t)offset * b2m) >> 32; |
||||
} |
||||
|
||||
uint32_t AudioSDPlayer_F32::lengthMillis(void) { |
||||
uint8_t s = *(volatile uint8_t *)&state; |
||||
if (s >= 8 && s != STATE_PAUSED) return 0; |
||||
uint32_t tlength = *(volatile uint32_t *)&total_length; |
||||
uint32_t b2m = *(volatile uint32_t *)&bytes2millis; |
||||
return ((uint64_t)tlength * b2m) >> 32; |
||||
} |
||||
@ -0,0 +1,227 @@ |
||||
/* *** AudioSDPlayer_F32.h ***
|
||||
* |
||||
* Audio Library for Teensy 3.X |
||||
* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.comaudio_block_samples |
||||
* |
||||
* Development of this audio library was funded by PJRC.COM, LLC by sales of |
||||
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop |
||||
* open source software by purchasing Teensy or other PJRC products. |
||||
* |
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy |
||||
* of this software and associated documentation files (the "Software"), to deal |
||||
* in the Software without restriction, including without limitation the rights |
||||
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
||||
* copies of the Software, and to permit persons to whom the Software is |
||||
* furnished to do so, subject to the following conditions: |
||||
* |
||||
* The above copyright notice, development funding notice, and this permission |
||||
* notice shall be included in all copies or substantial portions of the Software. |
||||
* |
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
||||
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN |
||||
* THE SOFTWARE. |
||||
*/ |
||||
/*
|
||||
* Extended by Chip Audette, OpenAudio, Dec 2019 |
||||
* Converted to F32 and to variable audio block length |
||||
* The F32 conversion is under the MIT License. Use at your own risk. |
||||
*/ |
||||
|
||||
/* WAV File Format
|
||||
Bytes Meaning |
||||
1 - 4 “RIFF” Marks the file as a riff file. Characters are each 1 byte long. |
||||
5 - 8 File size (integer) Size of the overall file - 8 bytes, in bytes |
||||
(32-bit integer). |
||||
9 -12 “WAVE” File Type Header. For our purposes, it always equals “WAVE”. |
||||
13-16 “fmt " Format chunk marker. Includes trailing null |
||||
17-20 Length of format data as listed above, e.g., 16 |
||||
21-22 Type of format (1 is PCM) - 2 byte integer |
||||
23-24 Number of Channels - 2 byte integer, e.g. 2 |
||||
25-28 Sample Rate - 32 byte integer. Common values are 44100 (CD), |
||||
48000 (DAT). Sample Rate = Number of Samples per second, or Hertz. |
||||
29-32 176400 (Sample Rate * BitsPerSample * Channels) / 8. |
||||
33-34 4 (BitsPerSample * Channels) / 8.1 - 8 bit mono2 - 8 bit |
||||
stereo/16 bit mono4 - 16 bit stereo |
||||
35-36 16 Bits per sample or other |
||||
37-40 "data" Marks the beginning of the data section. |
||||
41-44 File size (data) Size of the data section. |
||||
* |
||||
Sample of WAV file start: |
||||
00000000 52494646 66EA6903 57415645 666D7420 RIFFf.i.WAVEfmt |
||||
00000010 10000000 01000200 44AC0000 10B10200 ........D....... |
||||
00000020 04001000 4C495354 3A000000 494E464F ....LIST:...INFO |
||||
00000030 494E414D 14000000 49205761 6E742054 INAM....I Want T |
||||
00000040 6F20436F 6D65204F 76657200 49415254 o Come Over.IART |
||||
00000050 12000000 4D656C69 73736120 45746865 ....Melissa Ethe |
||||
00000060 72696467 65006461 746100EA 69030100 ridge.data..i... |
||||
00000070 FEFF0300 FCFF0400 FDFF0200 0000FEFF ................ |
||||
00000080 0300FDFF 0200FFFF 00000100 FEFF0300 ................ |
||||
00000090 FDFF0300 FDFF0200 FFFF0100 0000FFFF ................ |
||||
*/ |
||||
|
||||
/* *** SAMPLE RATES ***
|
||||
* In the case of WAV files, there is a specified sample rate that is part |
||||
* of the header data. The file is a stream of numbers. If these are |
||||
* turned into voltages and played at the specified rate, everything will |
||||
* sound correct. For shorthand, we will call this the WAV rate. |
||||
* |
||||
* There is also a Teensy Sample Rate for the Audio system, set by things |
||||
* like the hardware clock for I2S. If this sample rate is the same as |
||||
* the WAV rate, life is simple and we process and output a sample |
||||
* with the WAV rate being achieved. A second case is for the WAV rate |
||||
* to be an integer sub-multiple of the Teensy Sample Rate. This would |
||||
* allow the wave file to run with a 12 ksps sample rate and the Teensy |
||||
* Sample Rate to be 48, or 96, ksps. There is a data interpolator |
||||
* included after the the data has been read from the file. This requires |
||||
* specification of the sub-multiple integer and a FIR filter to complete |
||||
* the interpolation operation. The structure sampleSubMultiple, |
||||
* provided by the .INO, communicates this design information from the INO. |
||||
* |
||||
* The third case is to have the Wave rate and the Teensy Sample rate |
||||
* related by a rational fraction. This would require a rate changet |
||||
* consisting of both a decimator and an interpolator. None of that is |
||||
* included in this class. |
||||
*/ |
||||
|
||||
#ifndef AudioSDPlayer_F32_h_ |
||||
#define AudioSDPlayer_F32_h_ |
||||
|
||||
#include "Arduino.h" |
||||
#include "AudioSettings_F32.h" |
||||
#include "AudioStream_F32.h" |
||||
|
||||
#include <SdFat.h> //included in Teensy install as of Teensyduino 1.54-bete3 |
||||
|
||||
// This communicates the info for running slow WAV file sample rates.
|
||||
// This one is declared in the .INO
|
||||
struct subMult { |
||||
uint16_t rateRatio; // Should be 1 for no rate change, else 2, 4, 8
|
||||
uint16_t numCoeffs; // FIR filter
|
||||
float32_t* firCoeffs; // FIR Filter Coeffs
|
||||
float32_t* firBufferL; // pointer to 127 + numCoeffs float32_t, left ch
|
||||
float32_t* firBufferR; // pointer to 127 + numCoeffs float32_t, right ch
|
||||
}; |
||||
|
||||
// This communicates the important parameters of the WAV file. This is
|
||||
// declared in AudioSDPlayer_F32 to provide data to the .INO.
|
||||
struct wavData { |
||||
uint16_t audio_format; // Should be 1 for PCM
|
||||
uint16_t num_channels; // 1 for mono, 2 for stereo
|
||||
uint32_t sample_rate; // 44100, 48000, etc
|
||||
uint16_t bits; // Number of bits per sample
|
||||
}; |
||||
|
||||
class AudioSDPlayer_F32 : public AudioStream_F32 |
||||
{ |
||||
//GUI: inputs:0, outputs:2 //this line used for automatic generation of GUI nodes
|
||||
public: |
||||
|
||||
AudioSDPlayer_F32(void) : |
||||
AudioStream_F32(0, NULL), block_left_f32(NULL), block_right_f32(NULL) |
||||
{ |
||||
begin(); |
||||
} |
||||
|
||||
AudioSDPlayer_F32(const AudioSettings_F32 &settings) : |
||||
AudioStream_F32(0, NULL), block_left_f32(NULL), block_right_f32(NULL) |
||||
{ |
||||
setSampleRate_Hz(settings.sample_rate_Hz); |
||||
//setBlockSize(settings.audio_block_samples); // Always 128
|
||||
begin(); |
||||
} |
||||
|
||||
void begin(void); //begins SD card
|
||||
bool play(const char *filename); |
||||
void stop(void); |
||||
void togglePlayPause(void); |
||||
bool isPaused(void); |
||||
bool isStopped(void); |
||||
bool isPlaying(void); |
||||
uint32_t positionMillis(void); |
||||
uint32_t lengthMillis(void); |
||||
|
||||
// Required when WAV file is at a sub-multiple rate of audio sampling rate
|
||||
void setSubMult(subMult* pSampleSubMultipleStruct) { |
||||
if(pSampleSubMultipleStruct->rateRatio == 1 || |
||||
pSampleSubMultipleStruct->rateRatio == 2 || |
||||
pSampleSubMultipleStruct->rateRatio == 4 || |
||||
pSampleSubMultipleStruct->rateRatio == 8) |
||||
{ |
||||
pSampleSubMultiple = pSampleSubMultipleStruct; |
||||
if(pSampleSubMultiple->numCoeffs > 1 && |
||||
pSampleSubMultiple->firBufferL ) |
||||
{ |
||||
arm_fir_init_f32(&fir_instL, |
||||
pSampleSubMultiple->numCoeffs, |
||||
(float32_t *)pSampleSubMultiple->firCoeffs, |
||||
(float32_t *)pSampleSubMultiple->firBufferL, |
||||
(uint32_t)audio_block_samples); |
||||
arm_fir_init_f32(&fir_instR, |
||||
pSampleSubMultiple->numCoeffs, |
||||
(float32_t *)pSampleSubMultiple->firCoeffs, |
||||
(float32_t *)pSampleSubMultiple->firBufferR, |
||||
(uint32_t)audio_block_samples); |
||||
} |
||||
} |
||||
else |
||||
Serial.println("Illegal sub-division multiple for WAV rate."); |
||||
} |
||||
|
||||
// Provides basic meta-data about WAV file.
|
||||
wavData* getCurrentWavData(void) { |
||||
return ¤tWavData; // Pointer to structure
|
||||
} |
||||
|
||||
float32_t setSampleRate_Hz(float32_t fs_Hz) { |
||||
sample_rate_Hz = fs_Hz; |
||||
updateBytes2Millis(); |
||||
return sample_rate_Hz; |
||||
} |
||||
|
||||
virtual void update(void); |
||||
|
||||
private: |
||||
File wavfile; |
||||
struct subMult* pSampleSubMultiple = &nEqOneTemp; |
||||
// Next is a dummy structure to divide by 1 when no INO structure
|
||||
struct subMult nEqOneTemp = {1, 0, NULL, NULL, NULL}; |
||||
arm_fir_instance_f32 fir_instL; |
||||
arm_fir_instance_f32 fir_instR; |
||||
struct wavData currentWavData = {1, 2, 44100, 16}; |
||||
bool consume(uint32_t size); |
||||
bool parse_format(void); |
||||
uint32_t header[10]; // temporary storage of wav header data
|
||||
uint32_t data_length; // number of bytes remaining in current section
|
||||
uint32_t total_length; // number of audio data bytes in file
|
||||
uint16_t channels = 1; //number of audio channels
|
||||
uint16_t bits = 16; // number of bits per sample
|
||||
uint32_t bytes2millis; |
||||
// Variables for audio library storage, float32_t
|
||||
audio_block_f32_t *block_left_f32 = NULL; |
||||
audio_block_f32_t *block_right_f32 = NULL; |
||||
uint16_t block_offset; // how much data is in block_left & block_right
|
||||
// Variables for buffering the WAV file read, uint8_t
|
||||
uint8_t buffer[512]; // buffer one block of SD file data
|
||||
uint16_t buffer_offset; // where we're at consuming "buffer"
|
||||
uint16_t buffer_length; // how many data bytes are in "buffer" (512 until last read)
|
||||
uint8_t header_offset; // number of bytes in header[]
|
||||
// Variables to control the WAV file reading
|
||||
uint8_t state; |
||||
uint8_t state_play; |
||||
uint8_t leftover_bytes; |
||||
// Variables for WAV file sampled at a sub rate from audio process
|
||||
uint8_t zerosToSend = 0; |
||||
|
||||
static unsigned long update_counter; |
||||
float sample_rate_Hz = ((float)AUDIO_SAMPLE_RATE_EXACT); |
||||
uint16_t audio_block_samples = AUDIO_BLOCK_SAMPLES; |
||||
|
||||
uint32_t updateBytes2Millis(void); |
||||
//int32_t pctr = 0;
|
||||
}; |
||||
|
||||
#endif |
||||
@ -0,0 +1,80 @@ |
||||
/*
|
||||
* SDWavPlayer |
||||
* |
||||
* Created: Chip Audette, OpenAudio, Dec 2019 |
||||
* Based On: WaveFilePlayer from Paul Stoffregen, PJRC, Teensy |
||||
* |
||||
* Play back a WAV file through the Typman. |
||||
* |
||||
* For access to WAV files, please visit https://www.pjrc.com/teensy/td_libs_AudioDataFiles.html.
|
||||
* |
||||
*/ |
||||
|
||||
#include "OpenAudio_ArduinoLibrary.h" |
||||
#include "AudioSDPlayer_F32.h" |
||||
|
||||
//set the sample rate and block size
|
||||
const float sample_rate_Hz = 44100.0f; |
||||
const int audio_block_samples = 128; // Must be 128 for SD recording.
|
||||
AudioSettings_F32 audio_settings(sample_rate_Hz, audio_block_samples); |
||||
|
||||
//create audio objects
|
||||
AudioSDPlayer_F32 audioSDPlayer(audio_settings); |
||||
AudioOutputI2S_F32 audioOutput(audio_settings); |
||||
//Tympan myTympan(TympanRev::E); //do TympanRev::D or TympanRev::E
|
||||
|
||||
//create audio connections
|
||||
AudioConnection_F32 patchCord1(audioSDPlayer, 0, audioOutput, 0); |
||||
AudioConnection_F32 patchCord2(audioSDPlayer, 1, audioOutput, 1); |
||||
AudioControlSGTL5000 sgtl5000_1; |
||||
|
||||
|
||||
// Use these with the Teensy 4.x Rev D Audio Shield
|
||||
#define SDCARD_CS_PIN 10 |
||||
#define SDCARD_MOSI_PIN 11 |
||||
#define SDCARD_SCK_PIN 13 |
||||
|
||||
void setup() { |
||||
Serial.begin(300); delay(1000); |
||||
Serial.print("### SDWavPlayer ###"); |
||||
Serial.print("Sample Rate (Hz): "); Serial.println(audio_settings.sample_rate_Hz); |
||||
Serial.print("Audio Block Size (samples): "); Serial.println(audio_settings.audio_block_samples); |
||||
|
||||
// Audio connections require memory to work.
|
||||
AudioMemory_F32(20, audio_settings); |
||||
|
||||
sgtl5000_1.enable(); |
||||
|
||||
SPI.setMOSI(SDCARD_MOSI_PIN); |
||||
SPI.setSCK(SDCARD_SCK_PIN); |
||||
if (!(SD.begin(SDCARD_CS_PIN))) { |
||||
// stop here, but print a message repetitively
|
||||
while (1) { |
||||
Serial.println("*** Unable to access the SD card ***"); |
||||
delay(1000); |
||||
} |
||||
} |
||||
//prepare SD player
|
||||
audioSDPlayer.begin(); |
||||
|
||||
//finish setup
|
||||
delay(2000); //stall a second
|
||||
Serial.println("Setup complete."); |
||||
} |
||||
|
||||
unsigned long end_millis = 0; |
||||
String filename = "SDTEST1.WAV";// filenames are always uppercase 8.3 format
|
||||
void loop() { |
||||
|
||||
/*
|
||||
//service the audio player
|
||||
if (!audioSDPlayer.isPlaying()) { //wait until previous play is done
|
||||
//start playing audio
|
||||
Serial.print("Starting audio player: "); |
||||
Serial.println(filename); |
||||
audioSDPlayer.play(filename); |
||||
} |
||||
*/ |
||||
|
||||
delay(500); |
||||
} |
||||
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Reference in new issue