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authorScott Shawcroft <scott@adafruit.com>2020-01-08 12:50:13 -0800
committerGitHub <noreply@github.com>2020-01-08 12:50:13 -0800
commita10cd8580c3227f64cc0b0a7762f37732e31b91b (patch)
tree890b421c622e0b1668896f7397fc799c41fda648
parent28c1e4f48e5edf16b3aedc0d25e8d01d014def5f (diff)
parentdd6010a17ef8e12e1cb3f36721b205cc1cb8eee9 (diff)
Merge pull request #2445 from jepler/mp3-jeplayer-fixes5.0.0-beta.3
Fixes for JEplayer
m---------ports/atmel-samd/peripherals0
-rw-r--r--py/circuitpy_defns.mk2
-rw-r--r--shared-bindings/audiomp3/MP3Decoder.c (renamed from shared-bindings/audiomp3/MP3File.c)65
-rw-r--r--shared-bindings/audiomp3/MP3Decoder.h (renamed from shared-bindings/audiomp3/MP3File.h)4
-rw-r--r--shared-bindings/audiomp3/__init__.c6
-rw-r--r--shared-module/audiomp3/MP3Decoder.c (renamed from shared-module/audiomp3/MP3File.c)76
-rw-r--r--shared-module/audiomp3/MP3Decoder.h (renamed from shared-module/audiomp3/MP3File.h)2
7 files changed, 118 insertions, 37 deletions
diff --git a/ports/atmel-samd/peripherals b/ports/atmel-samd/peripherals
-Subproject 4c0deecf889da0074c1dbc9a5e2d24cb7c7a31c
+Subproject b89811f22a24ac350079ceaf0cdf0e62aa03f4f
diff --git a/py/circuitpy_defns.mk b/py/circuitpy_defns.mk
index 35a977946..1f2d6c73e 100644
--- 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/MP3File.c b/shared-bindings/audiomp3/MP3Decoder.c
index f518bae4b..224042212 100644
--- a/shared-bindings/audiomp3/MP3File.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/MP3File.h b/shared-bindings/audiomp3/MP3Decoder.h
index adc13ea2c..36d525e93 100644
--- a/shared-bindings/audiomp3/MP3File.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
index 06f852ff1..fb2187669 100644
--- 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/MP3File.c b/shared-module/audiomp3/MP3Decoder.c
index 0aa4f24a6..30357c616 100644
--- a/shared-module/audiomp3/MP3File.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/MP3File.h b/shared-module/audiomp3/MP3Decoder.h
index 9d99e8d1f..9ee1d0949 100644
--- a/shared-module/audiomp3/MP3File.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