Merge pull request #2445 from jepler/mp3-jeplayer-fixes

tannewt · web-flow · commit a10cd8580c32 · 2020-01-08T12:50:13.000-08:00
Fixes for JEplayer
diff --git a/ports/atmel-samd/peripherals b/ports/atmel-samd/peripherals
@@ -1 +1 @@
-Subproject commit 4c0deecf889da0074c1dbc9a5e2d24cb7c7a31c6
+Subproject commit b89811f22a24ac350079ceaf0cdf0e62aa03f4f4
diff --git a/py/circuitpy_defns.mk b/py/circuitpy_defns.mk
@@ -323,7 +323,7 @@ SRC_SHARED_MODULE_ALL = \
 	audiomixer/Mixer.c \
 	audiomixer/MixerVoice.c \
 	audiomp3/__init__.c \
-	audiomp3/MP3File.c \
+	audiomp3/MP3Decoder.c \
 	bitbangio/I2C.c \
 	bitbangio/OneWire.c \
 	bitbangio/SPI.c \
diff --git a/shared-bindings/audiomp3/MP3Decoder.c b/shared-bindings/audiomp3/MP3Decoder.c
@@ -30,18 +30,16 @@
 #include "lib/utils/context_manager_helpers.h"
 #include "py/objproperty.h"
 #include "py/runtime.h"
-#include "shared-bindings/audiomp3/MP3File.h"
+#include "shared-bindings/audiomp3/MP3Decoder.h"
 #include "shared-bindings/util.h"
 #include "supervisor/shared/translate.h"
 
 //| .. currentmodule:: audiomp3
 //|
-//| :class:`MP3` -- Load a mp3 file for audio playback
-//| ========================================================
+//| :class:`MP3Decoder` -- Load a mp3 file for audio playback
+//| =========================================================
 //|
-//| A .mp3 file prepped for audio playback. Only mono and stereo files are supported. Samples must
-//| be 8 bit unsigned or 16 bit signed. If a buffer is provided, it will be used instead of allocating
-//| an internal buffer.
+//| An object that decodes MP3 files for playback on an audio device.
 //|
 //| .. class:: MP3(file[, buffer])
 //|
@@ -63,7 +61,7 @@
 //|     speaker_enable.switch_to_output(value=True)
 //|
 //|     data = open("cplay-16bit-16khz-64kbps.mp3", "rb")
-//|     mp3 = audiomp3.MP3File(data)
+//|     mp3 = audiomp3.MP3Decoder(data)
 //|     a = audioio.AudioOut(board.A0)
 //|
 //|     print("playing")
@@ -129,6 +127,37 @@ STATIC mp_obj_t audiomp3_mp3file_obj___exit__(size_t n_args, const mp_obj_t *arg
 }
 STATIC MP_DEFINE_CONST_FUN_OBJ_VAR_BETWEEN(audiomp3_mp3file___exit___obj, 4, 4, audiomp3_mp3file_obj___exit__);
 
+//|   .. attribute:: file
+//|
+//|     File to play back.
+//|
+STATIC mp_obj_t audiomp3_mp3file_obj_get_file(mp_obj_t self_in) {
+    audiomp3_mp3file_obj_t *self = MP_OBJ_TO_PTR(self_in);
+    check_for_deinit(self);
+    return self->file;
+}
+MP_DEFINE_CONST_FUN_OBJ_1(audiomp3_mp3file_get_file_obj, audiomp3_mp3file_obj_get_file);
+
+STATIC mp_obj_t audiomp3_mp3file_obj_set_file(mp_obj_t self_in, mp_obj_t file) {
+    audiomp3_mp3file_obj_t *self = MP_OBJ_TO_PTR(self_in);
+    check_for_deinit(self);
+    if (!MP_OBJ_IS_TYPE(file, &mp_type_fileio)) {
+        mp_raise_TypeError(translate("file must be a file opened in byte mode"));
+    }
+    common_hal_audiomp3_mp3file_set_file(self, file);
+    return mp_const_none;
+}
+MP_DEFINE_CONST_FUN_OBJ_2(audiomp3_mp3file_set_file_obj, audiomp3_mp3file_obj_set_file);
+
+const mp_obj_property_t audiomp3_mp3file_file_obj = {
+    .base.type = &mp_type_property,
+    .proxy = {(mp_obj_t)&audiomp3_mp3file_get_file_obj,
+              (mp_obj_t)&audiomp3_mp3file_set_file_obj,
+              (mp_obj_t)&mp_const_none_obj},
+};
+
+
+
 //|   .. attribute:: sample_rate
 //|
 //|     32 bit value that dictates how quickly samples are loaded into the DAC
@@ -193,6 +222,24 @@ const mp_obj_property_t audiomp3_mp3file_channel_count_obj = {
               (mp_obj_t)&mp_const_none_obj},
 };
 
+//|   .. attribute:: rms_level
+//|
+//|     The RMS audio level of a recently played moment of audio. (read only)
+//|
+STATIC mp_obj_t audiomp3_mp3file_obj_get_rms_level(mp_obj_t self_in) {
+    audiomp3_mp3file_obj_t *self = MP_OBJ_TO_PTR(self_in);
+    check_for_deinit(self);
+    return mp_obj_new_float(common_hal_audiomp3_mp3file_get_rms_level(self));
+}
+MP_DEFINE_CONST_FUN_OBJ_1(audiomp3_mp3file_get_rms_level_obj, audiomp3_mp3file_obj_get_rms_level);
+
+const mp_obj_property_t audiomp3_mp3file_rms_level_obj = {
+    .base.type = &mp_type_property,
+    .proxy = {(mp_obj_t)&audiomp3_mp3file_get_rms_level_obj,
+              (mp_obj_t)&mp_const_none_obj,
+              (mp_obj_t)&mp_const_none_obj},
+};
+
 
 STATIC const mp_rom_map_elem_t audiomp3_mp3file_locals_dict_table[] = {
     // Methods
@@ -201,9 +248,11 @@ STATIC const mp_rom_map_elem_t audiomp3_mp3file_locals_dict_table[] = {
     { MP_ROM_QSTR(MP_QSTR___exit__), MP_ROM_PTR(&audiomp3_mp3file___exit___obj) },
 
     // Properties
+    { MP_ROM_QSTR(MP_QSTR_file), MP_ROM_PTR(&audiomp3_mp3file_file_obj) },
     { MP_ROM_QSTR(MP_QSTR_sample_rate), MP_ROM_PTR(&audiomp3_mp3file_sample_rate_obj) },
     { MP_ROM_QSTR(MP_QSTR_bits_per_sample), MP_ROM_PTR(&audiomp3_mp3file_bits_per_sample_obj) },
     { MP_ROM_QSTR(MP_QSTR_channel_count), MP_ROM_PTR(&audiomp3_mp3file_channel_count_obj) },
+    { MP_ROM_QSTR(MP_QSTR_rms_level), MP_ROM_PTR(&audiomp3_mp3file_rms_level_obj) },
 };
 STATIC MP_DEFINE_CONST_DICT(audiomp3_mp3file_locals_dict, audiomp3_mp3file_locals_dict_table);
 
@@ -219,7 +268,7 @@ STATIC const audiosample_p_t audiomp3_mp3file_proto = {
 
 const mp_obj_type_t audiomp3_mp3file_type = {
     { &mp_type_type },
-    .name = MP_QSTR_MP3File,
+    .name = MP_QSTR_MP3Decoder,
     .make_new = audiomp3_mp3file_make_new,
     .locals_dict = (mp_obj_dict_t*)&audiomp3_mp3file_locals_dict,
     .protocol = &audiomp3_mp3file_proto,
diff --git a/shared-bindings/audiomp3/MP3Decoder.h b/shared-bindings/audiomp3/MP3Decoder.h
@@ -31,18 +31,20 @@
 #include "py/obj.h"
 #include "extmod/vfs_fat.h"
 
-#include "shared-module/audiomp3/MP3File.h"
+#include "shared-module/audiomp3/MP3Decoder.h"
 
 extern const mp_obj_type_t audiomp3_mp3file_type;
 
 void common_hal_audiomp3_mp3file_construct(audiomp3_mp3file_obj_t* self,
     pyb_file_obj_t* file, uint8_t *buffer, size_t buffer_size);
 
+void common_hal_audiomp3_mp3file_set_file(audiomp3_mp3file_obj_t* self, pyb_file_obj_t* file);
 void common_hal_audiomp3_mp3file_deinit(audiomp3_mp3file_obj_t* self);
 bool common_hal_audiomp3_mp3file_deinited(audiomp3_mp3file_obj_t* self);
 uint32_t common_hal_audiomp3_mp3file_get_sample_rate(audiomp3_mp3file_obj_t* self);
 void common_hal_audiomp3_mp3file_set_sample_rate(audiomp3_mp3file_obj_t* self, uint32_t sample_rate);
 uint8_t common_hal_audiomp3_mp3file_get_bits_per_sample(audiomp3_mp3file_obj_t* self);
 uint8_t common_hal_audiomp3_mp3file_get_channel_count(audiomp3_mp3file_obj_t* self);
+float common_hal_audiomp3_mp3file_get_rms_level(audiomp3_mp3file_obj_t* self);
 
 #endif // MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOIO_MP3FILE_H
diff --git a/shared-bindings/audiomp3/__init__.c b/shared-bindings/audiomp3/__init__.c
@@ -29,7 +29,7 @@
 #include "py/obj.h"
 #include "py/runtime.h"
 
-#include "shared-bindings/audiomp3/MP3File.h"
+#include "shared-bindings/audiomp3/MP3Decoder.h"
 
 //| :mod:`audiomp3` --- Support for MP3-compressed audio files
 //| ==========================================================
@@ -44,12 +44,12 @@
 //| .. toctree::
 //|     :maxdepth: 3
 //|
-//|     MP3File
+//|     MP3Decoder
 //|
 
 STATIC const mp_rom_map_elem_t audiomp3_module_globals_table[] = {
     { MP_ROM_QSTR(MP_QSTR___name__), MP_ROM_QSTR(MP_QSTR_audiomp3) },
-    { MP_ROM_QSTR(MP_QSTR_MP3File), MP_ROM_PTR(&audiomp3_mp3file_type) },
+    { MP_ROM_QSTR(MP_QSTR_MP3Decoder), MP_ROM_PTR(&audiomp3_mp3file_type) },
 };
 
 STATIC MP_DEFINE_CONST_DICT(audiomp3_module_globals, audiomp3_module_globals_table);
diff --git a/shared-module/audiomp3/MP3Decoder.c b/shared-module/audiomp3/MP3Decoder.c
@@ -25,18 +25,21 @@
  * THE SOFTWARE.
  */
 
-#include "shared-bindings/audiomp3/MP3File.h"
+#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/MP3File.h"
+#include "shared-module/audiomp3/MP3Decoder.h"
 #include "supervisor/shared/translate.h"
 #include "lib/mp3/src/mp3common.h"
 
+#define MAX_BUFFER_LEN (MAX_NSAMP * MAX_NGRAN * MAX_NCHAN * sizeof(int16_t))
+
 /** Fill the input buffer if it is less than half full.
  *
  * Returns true if the input buffer contains any useful data,
@@ -143,7 +146,7 @@ STATIC bool mp3file_get_next_frame_info(audiomp3_mp3file_obj_t* self, MP3FrameIn
     do {
         err = MP3GetNextFrameInfo(self->decoder, fi, READ_PTR(self));
         if (err == ERR_MP3_NONE) {
-           break;
+            break;
         }
         CONSUME(self, 1);
         mp3file_find_sync_word(self);
@@ -165,7 +168,6 @@ void common_hal_audiomp3_mp3file_construct(audiomp3_mp3file_obj_t* self,
     // than the two 4kB output buffers, except that the alignment allows to
     // never allocate that extra frame buffer.
 
-    self->file = file;
     self->inbuf_length = 2048;
     self->inbuf_offset = self->inbuf_length;
     self->inbuf = m_malloc(self->inbuf_length, false);
@@ -181,40 +183,56 @@ void common_hal_audiomp3_mp3file_construct(audiomp3_mp3file_obj_t* self,
                      translate("Couldn't allocate decoder"));
     }
 
-    mp3file_find_sync_word(self);
-    MP3FrameInfo fi;
-    if(!mp3file_get_next_frame_info(self, &fi)) {
-        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);
-
     if ((intptr_t)buffer & 1) {
         buffer += 1; buffer_size -= 1;
     }
-    if (buffer_size >= 2 * self->frame_buffer_size) {
-        self->len = buffer_size / 2 / self->frame_buffer_size * self->frame_buffer_size;
+    if (buffer_size >= 2 * MAX_BUFFER_LEN) {
         self->buffers[0] = (int16_t*)(void*)buffer;
-        self->buffers[1] = (int16_t*)(void*)buffer + self->len;
+        self->buffers[1] = (int16_t*)(void*)(buffer + MAX_BUFFER_LEN);
     } else {
-        self->len = 2 * self->frame_buffer_size;
-        self->buffers[0] = m_malloc(self->len, false);
+        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(self->len, false);
+        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) {
+    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(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;
+    if(!mp3file_get_next_frame_info(self, &fi)) {
+        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;
 }
 
 void common_hal_audiomp3_mp3file_deinit(audiomp3_mp3file_obj_t* self) {
@@ -280,7 +298,6 @@ audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t*
         channel = 0;
     }
 
-    *bufptr = (uint8_t*)(self->buffers[self->buffer_index] + channel);
     *buffer_length = self->frame_buffer_size;
 
     if (channel == self->other_channel) {
@@ -289,11 +306,12 @@ audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t*
         return GET_BUFFER_MORE_DATA;
     }
 
-    self->other_channel = 1-channel;
-    self->other_buffer_index = self->buffer_index;
 
     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)) {
@@ -322,3 +340,13 @@ void audiomp3_mp3file_get_buffer_structure(audiomp3_mp3file_obj_t* self, bool si
         *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));
+}
diff --git a/shared-module/audiomp3/MP3Decoder.h b/shared-module/audiomp3/MP3Decoder.h
@@ -67,4 +67,6 @@ void audiomp3_mp3file_get_buffer_structure(audiomp3_mp3file_obj_t* self, bool si
                                            bool* single_buffer, bool* samples_signed,
                                            uint32_t* max_buffer_length, uint8_t* spacing);
 
+float audiomp3_mp3file_get_rms_level(audiomp3_mp3file_obj_t* self);
+
 #endif // MICROPY_INCLUDED_SHARED_MODULE_AUDIOIO_MP3FILE_H
