add circuitpython code

1 files changed, 399 insertions, 0 deletions

diff --git a/circuitpython/shared-module/audiomp3/MP3Decoder.c b/circuitpython/shared-module/audiomp3/MP3Decoder.c
new file mode 100644
index 0000000..bbc7e1c
--- /dev/null
+++ b/circuitpython/shared-module/audiomp3/MP3Decoder.c
@@ -0,0 +1,399 @@
+/*
+ * This file is part of the Micro Python project, http://micropython.org/
+ *
+ * The MIT License (MIT)
+ *
+ * Copyright (c) 2018 Scott Shawcroft for Adafruit Industries
+ * Copyright (c) 2019 Jeff Epler for Adafruit Industries
+ *
+ * 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 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.
+ */
+
+#include "shared-bindings/audiomp3/MP3Decoder.h"
+
+#include <stdint.h>
+#include <string.h>
+#include <math.h>
+
+#include "py/mperrno.h"
+#include "py/runtime.h"
+
+#include "shared-module/audiomp3/MP3Decoder.h"
+#include "supervisor/shared/translate.h"
+#include "supervisor/background_callback.h"
+#include "lib/mp3/src/mp3common.h"
+
+#define MAX_BUFFER_LEN (MAX_NSAMP * MAX_NGRAN * MAX_NCHAN * sizeof(int16_t))
+
+/** Fill the input buffer unconditionally.
+ *
+ * Returns true if the input buffer contains any useful data,
+ * false otherwise.  (The input buffer will be padded to the end with
+ * 0 bytes, which do not interfere with MP3 decoding)
+ *
+ * Raises OSError if f_read fails.
+ *
+ * Sets self->eof if any read of the file returns 0 bytes
+ */
+STATIC bool mp3file_update_inbuf_always(audiomp3_mp3file_obj_t *self) {
+    // If we didn't previously reach the end of file, we can try reading now
+    if (!self->eof && self->inbuf_offset != 0) {
+
+        // Move the unconsumed portion of the buffer to the start
+        uint8_t *end_of_buffer = self->inbuf + self->inbuf_length;
+        uint8_t *new_end_of_data = self->inbuf + self->inbuf_length - self->inbuf_offset;
+        memmove(self->inbuf, self->inbuf + self->inbuf_offset,
+            self->inbuf_length - self->inbuf_offset);
+        self->inbuf_offset = 0;
+
+        UINT to_read = end_of_buffer - new_end_of_data;
+        UINT bytes_read = 0;
+        memset(new_end_of_data, 0, to_read);
+        if (f_read(&self->file->fp, new_end_of_data, to_read, &bytes_read) != FR_OK) {
+            self->eof = true;
+            mp_raise_OSError(MP_EIO);
+        }
+
+        if (bytes_read == 0) {
+            self->eof = true;
+        }
+
+        if (to_read != bytes_read) {
+            new_end_of_data += bytes_read;
+            memset(new_end_of_data, 0, end_of_buffer - new_end_of_data);
+        }
+
+    }
+
+    // Return true iff there are at least some useful bytes in the buffer
+    return self->inbuf_offset < self->inbuf_length;
+}
+
+/** Update the inbuf from a background callback.
+ *
+ * This variant is introduced so that at the site of the
+ * add_background_callback_core call, the prototype matches.
+ */
+STATIC void mp3file_update_inbuf_cb(void *self) {
+    mp3file_update_inbuf_always(self);
+}
+
+/** Fill the input buffer if it is less than half full.
+ *
+ * Returns the same as mp3file_update_inbuf_always.
+ */
+STATIC bool mp3file_update_inbuf_half(audiomp3_mp3file_obj_t *self) {
+    // If buffer is over half full, do nothing
+    if (self->inbuf_offset < self->inbuf_length / 2) {
+        return true;
+    }
+
+    return mp3file_update_inbuf_always(self);
+}
+
+#define READ_PTR(self) (self->inbuf + self->inbuf_offset)
+#define BYTES_LEFT(self) (self->inbuf_length - self->inbuf_offset)
+#define CONSUME(self, n) (self->inbuf_offset += n)
+
+// http://id3.org/d3v2.3.0
+// http://id3.org/id3v2.3.0
+STATIC void mp3file_skip_id3v2(audiomp3_mp3file_obj_t *self) {
+    mp3file_update_inbuf_half(self);
+    if (BYTES_LEFT(self) < 10) {
+        return;
+    }
+    uint8_t *data = READ_PTR(self);
+    if (!(
+        data[0] == 'I' &&
+        data[1] == 'D' &&
+        data[2] == '3' &&
+        data[3] != 0xff &&
+        data[4] != 0xff &&
+        (data[5] & 0x1f) == 0 &&
+        (data[6] & 0x80) == 0 &&
+        (data[7] & 0x80) == 0 &&
+        (data[8] & 0x80) == 0 &&
+        (data[9] & 0x80) == 0)) {
+        return;
+    }
+    uint32_t size = (data[6] << 21) | (data[7] << 14) | (data[8] << 7) | (data[9]);
+    size += 10; // size excludes the "header" (but not the "extended header")
+    // First, deduct from size whatever is left in buffer
+    uint32_t to_consume = MIN(size, BYTES_LEFT(self));
+    CONSUME(self, to_consume);
+    size -= to_consume;
+
+    // Next, seek in the file after the header
+    f_lseek(&self->file->fp, f_tell(&self->file->fp) + size);
+    return;
+}
+
+/* If a sync word can be found, advance to it and return true.  Otherwise,
+ * return false.
+ */
+STATIC bool mp3file_find_sync_word(audiomp3_mp3file_obj_t *self) {
+    do {
+        mp3file_update_inbuf_half(self);
+        int offset = MP3FindSyncWord(READ_PTR(self), BYTES_LEFT(self));
+        if (offset >= 0) {
+            CONSUME(self, offset);
+            mp3file_update_inbuf_half(self);
+            return true;
+        }
+        CONSUME(self, MAX(0, BYTES_LEFT(self) - 16));
+    } while (!self->eof);
+    return false;
+}
+
+STATIC bool mp3file_get_next_frame_info(audiomp3_mp3file_obj_t *self, MP3FrameInfo *fi) {
+    int err;
+    do {
+        err = MP3GetNextFrameInfo(self->decoder, fi, READ_PTR(self));
+        if (err == ERR_MP3_NONE) {
+            break;
+        }
+        CONSUME(self, 1);
+        mp3file_find_sync_word(self);
+    } while (!self->eof);
+    return err == ERR_MP3_NONE;
+}
+
+void common_hal_audiomp3_mp3file_construct(audiomp3_mp3file_obj_t *self,
+    pyb_file_obj_t *file,
+    uint8_t *buffer,
+    size_t buffer_size) {
+    // XXX Adafruit_MP3 uses a 2kB input buffer and two 4kB output buffers.
+    // for a whopping total of 10kB buffers (+mp3 decoder state and frame buffer)
+    // At 44kHz, that's 23ms of output audio data.
+    //
+    // We will choose a slightly different allocation strategy for the output:
+    // Make sure the buffers are sized exactly to match (a multiple of) the
+    // frame size; this is typically 2304 * 2 bytes, so a little bit bigger
+    // than the two 4kB output buffers, except that the alignment allows to
+    // never allocate that extra frame buffer.
+
+    self->inbuf_length = 2048;
+    self->inbuf_offset = self->inbuf_length;
+    self->inbuf = m_malloc(self->inbuf_length, false);
+    if (self->inbuf == NULL) {
+        common_hal_audiomp3_mp3file_deinit(self);
+        mp_raise_msg(&mp_type_MemoryError,
+            translate("Couldn't allocate input buffer"));
+    }
+    self->decoder = MP3InitDecoder();
+    if (self->decoder == NULL) {
+        common_hal_audiomp3_mp3file_deinit(self);
+        mp_raise_msg(&mp_type_MemoryError,
+            translate("Couldn't allocate decoder"));
+    }
+
+    if ((intptr_t)buffer & 1) {
+        buffer += 1;
+        buffer_size -= 1;
+    }
+    if (buffer_size >= 2 * MAX_BUFFER_LEN) {
+        self->buffers[0] = (int16_t *)(void *)buffer;
+        self->buffers[1] = (int16_t *)(void *)(buffer + MAX_BUFFER_LEN);
+    } else {
+        self->buffers[0] = m_malloc(MAX_BUFFER_LEN, false);
+        if (self->buffers[0] == NULL) {
+            common_hal_audiomp3_mp3file_deinit(self);
+            mp_raise_msg(&mp_type_MemoryError,
+                translate("Couldn't allocate first buffer"));
+        }
+
+        self->buffers[1] = m_malloc(MAX_BUFFER_LEN, false);
+        if (self->buffers[1] == NULL) {
+            common_hal_audiomp3_mp3file_deinit(self);
+            mp_raise_msg(&mp_type_MemoryError,
+                translate("Couldn't allocate second buffer"));
+        }
+    }
+
+    common_hal_audiomp3_mp3file_set_file(self, file);
+}
+
+void common_hal_audiomp3_mp3file_set_file(audiomp3_mp3file_obj_t *self, pyb_file_obj_t *file) {
+    background_callback_begin_critical_section();
+
+    self->file = file;
+    f_lseek(&self->file->fp, 0);
+    self->inbuf_offset = self->inbuf_length;
+    self->eof = 0;
+    self->other_channel = -1;
+    mp3file_update_inbuf_half(self);
+    mp3file_find_sync_word(self);
+    // It **SHOULD** not be necessary to do this; the buffer should be filled
+    // with fresh content before it is returned by get_buffer().  The fact that
+    // this is necessary to avoid a glitch at the start of playback of a second
+    // track using the same decoder object means there's still a bug in
+    // get_buffer() that I didn't understand.
+    memset(self->buffers[0], 0, MAX_BUFFER_LEN);
+    memset(self->buffers[1], 0, MAX_BUFFER_LEN);
+    MP3FrameInfo fi;
+    bool result = mp3file_get_next_frame_info(self, &fi);
+    background_callback_end_critical_section();
+    if (!result) {
+        mp_raise_msg(&mp_type_RuntimeError,
+            translate("Failed to parse MP3 file"));
+    }
+
+    self->sample_rate = fi.samprate;
+    self->channel_count = fi.nChans;
+    self->frame_buffer_size = fi.outputSamps * sizeof(int16_t);
+    self->len = 2 * self->frame_buffer_size;
+    self->samples_decoded = 0;
+}
+
+void common_hal_audiomp3_mp3file_deinit(audiomp3_mp3file_obj_t *self) {
+    MP3FreeDecoder(self->decoder);
+    self->decoder = NULL;
+    self->inbuf = NULL;
+    self->buffers[0] = NULL;
+    self->buffers[1] = NULL;
+    self->file = NULL;
+    self->samples_decoded = 0;
+}
+
+bool common_hal_audiomp3_mp3file_deinited(audiomp3_mp3file_obj_t *self) {
+    return self->buffers[0] == NULL;
+}
+
+uint32_t common_hal_audiomp3_mp3file_get_sample_rate(audiomp3_mp3file_obj_t *self) {
+    return self->sample_rate;
+}
+
+void common_hal_audiomp3_mp3file_set_sample_rate(audiomp3_mp3file_obj_t *self,
+    uint32_t sample_rate) {
+    self->sample_rate = sample_rate;
+}
+
+uint8_t common_hal_audiomp3_mp3file_get_bits_per_sample(audiomp3_mp3file_obj_t *self) {
+    return 16;
+}
+
+uint8_t common_hal_audiomp3_mp3file_get_channel_count(audiomp3_mp3file_obj_t *self) {
+    return self->channel_count;
+}
+
+void audiomp3_mp3file_reset_buffer(audiomp3_mp3file_obj_t *self,
+    bool single_channel_output,
+    uint8_t channel) {
+    if (single_channel_output && channel == 1) {
+        return;
+    }
+    // We don't reset the buffer index in case we're looping and we have an odd number of buffer
+    // loads
+    background_callback_begin_critical_section();
+    f_lseek(&self->file->fp, 0);
+    self->inbuf_offset = self->inbuf_length;
+    self->eof = 0;
+    self->samples_decoded = 0;
+    self->other_channel = -1;
+    mp3file_update_inbuf_half(self);
+    mp3file_skip_id3v2(self);
+    mp3file_find_sync_word(self);
+    background_callback_end_critical_section();
+}
+
+audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t *self,
+    bool single_channel_output,
+    uint8_t channel,
+    uint8_t **bufptr,
+    uint32_t *buffer_length) {
+    if (!self->inbuf) {
+        *buffer_length = 0;
+        return GET_BUFFER_ERROR;
+    }
+    if (!single_channel_output) {
+        channel = 0;
+    }
+
+    *buffer_length = self->frame_buffer_size;
+
+    if (channel == self->other_channel) {
+        *bufptr = (uint8_t *)(self->buffers[self->other_buffer_index] + channel);
+        self->other_channel = -1;
+        self->samples_decoded += *buffer_length / sizeof(int16_t);
+        return GET_BUFFER_MORE_DATA;
+    }
+
+
+    self->buffer_index = !self->buffer_index;
+    self->other_channel = 1 - channel;
+    self->other_buffer_index = self->buffer_index;
+    int16_t *buffer = (int16_t *)(void *)self->buffers[self->buffer_index];
+    *bufptr = (uint8_t *)buffer;
+
+    mp3file_skip_id3v2(self);
+    if (!mp3file_find_sync_word(self)) {
+        *buffer_length = 0;
+        return self->eof ? GET_BUFFER_DONE : GET_BUFFER_ERROR;
+    }
+    int bytes_left = BYTES_LEFT(self);
+    uint8_t *inbuf = READ_PTR(self);
+    int err = MP3Decode(self->decoder, &inbuf, &bytes_left, buffer, 0);
+    CONSUME(self, BYTES_LEFT(self) - bytes_left);
+
+    if (err) {
+        *buffer_length = 0;
+        return GET_BUFFER_DONE;
+    }
+
+    self->samples_decoded += *buffer_length / sizeof(int16_t);
+
+    mp3file_skip_id3v2(self);
+    int result = mp3file_find_sync_word(self) ? GET_BUFFER_MORE_DATA : GET_BUFFER_DONE;
+
+    if (self->inbuf_offset >= 512) {
+        background_callback_add(
+            &self->inbuf_fill_cb,
+            mp3file_update_inbuf_cb,
+            self);
+    }
+
+    return result;
+}
+
+void audiomp3_mp3file_get_buffer_structure(audiomp3_mp3file_obj_t *self, bool single_channel_output,
+    bool *single_buffer, bool *samples_signed,
+    uint32_t *max_buffer_length, uint8_t *spacing) {
+    *single_buffer = false;
+    *samples_signed = true;
+    *max_buffer_length = self->frame_buffer_size;
+    if (single_channel_output) {
+        *spacing = self->channel_count;
+    } else {
+        *spacing = 1;
+    }
+}
+
+float common_hal_audiomp3_mp3file_get_rms_level(audiomp3_mp3file_obj_t *self) {
+    float sumsq = 0.f;
+    // Assumes no DC component to the audio.  Is that a safe assumption?
+    int16_t *buffer = (int16_t *)(void *)self->buffers[self->buffer_index];
+    for (size_t i = 0; i < self->frame_buffer_size / sizeof(int16_t); i++) {
+        sumsq += (float)buffer[i] * buffer[i];
+    }
+    return sqrtf(sumsq) / (self->frame_buffer_size / sizeof(int16_t));
+}
+
+uint32_t common_hal_audiomp3_mp3file_get_samples_decoded(audiomp3_mp3file_obj_t *self) {
+    return self->samples_decoded;
+}