diff options
Diffstat (limited to 'shared-module/audiomp3/MP3Decoder.c')
| -rw-r--r-- | shared-module/audiomp3/MP3Decoder.c | 113 |
1 files changed, 58 insertions, 55 deletions
diff --git a/shared-module/audiomp3/MP3Decoder.c b/shared-module/audiomp3/MP3Decoder.c index 5f3f42a51..2b623233e 100644 --- a/shared-module/audiomp3/MP3Decoder.c +++ b/shared-module/audiomp3/MP3Decoder.c @@ -51,7 +51,7 @@ * * Sets self->eof if any read of the file returns 0 bytes */ -STATIC bool mp3file_update_inbuf_always(audiomp3_mp3file_obj_t* self) { +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) { @@ -90,7 +90,7 @@ STATIC bool mp3file_update_inbuf_always(audiomp3_mp3file_obj_t* self) { * 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) { +STATIC void mp3file_update_inbuf_cb(void *self) { mp3file_update_inbuf_always(self); } @@ -98,9 +98,11 @@ STATIC void mp3file_update_inbuf_cb(void* self) { * * Returns the same as mp3file_update_inbuf_always. */ -STATIC bool mp3file_update_inbuf_half(audiomp3_mp3file_obj_t* self) { +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; + if (self->inbuf_offset < self->inbuf_length / 2) { + return true; + } return mp3file_update_inbuf_always(self); } @@ -111,23 +113,23 @@ STATIC bool mp3file_update_inbuf_half(audiomp3_mp3file_obj_t* self) { // http://id3.org/d3v2.3.0 // http://id3.org/id3v2.3.0 -STATIC void mp3file_skip_id3v2(audiomp3_mp3file_obj_t* self) { +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)) { + 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]); @@ -145,7 +147,7 @@ STATIC void mp3file_skip_id3v2(audiomp3_mp3file_obj_t* self) { /* 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) { +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)); @@ -159,7 +161,7 @@ STATIC bool mp3file_find_sync_word(audiomp3_mp3file_obj_t* self) { return false; } -STATIC bool mp3file_get_next_frame_info(audiomp3_mp3file_obj_t* self, MP3FrameInfo* fi) { +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)); @@ -172,10 +174,10 @@ STATIC bool mp3file_get_next_frame_info(audiomp3_mp3file_obj_t* self, MP3FrameIn 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) { +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. @@ -192,41 +194,42 @@ void common_hal_audiomp3_mp3file_construct(audiomp3_mp3file_obj_t* self, if (self->inbuf == NULL) { common_hal_audiomp3_mp3file_deinit(self); mp_raise_msg(&mp_type_MemoryError, - translate("Couldn't allocate input buffer")); + 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")); + translate("Couldn't allocate decoder")); } if ((intptr_t)buffer & 1) { - buffer += 1; buffer_size -= 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); + 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")); + 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")); + 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) { +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; @@ -248,16 +251,16 @@ void common_hal_audiomp3_mp3file_set_file(audiomp3_mp3file_obj_t* self, pyb_file background_callback_end_critical_section(); if (!result) { mp_raise_msg(&mp_type_RuntimeError, - translate("Failed to parse MP3 file")); + 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->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) { +void common_hal_audiomp3_mp3file_deinit(audiomp3_mp3file_obj_t *self) { MP3FreeDecoder(self->decoder); self->decoder = NULL; self->inbuf = NULL; @@ -266,34 +269,34 @@ void common_hal_audiomp3_mp3file_deinit(audiomp3_mp3file_obj_t* self) { self->file = NULL; } -bool common_hal_audiomp3_mp3file_deinited(audiomp3_mp3file_obj_t* self) { +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) { +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) { +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) { +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) { +uint8_t common_hal_audiomp3_mp3file_get_channel_count(audiomp3_mp3file_obj_t *self) { return self->channel_count; } -bool audiomp3_mp3file_samples_signed(audiomp3_mp3file_obj_t* self) { +bool audiomp3_mp3file_samples_signed(audiomp3_mp3file_obj_t *self) { return true; } -void audiomp3_mp3file_reset_buffer(audiomp3_mp3file_obj_t* self, - bool single_channel, - uint8_t channel) { +void audiomp3_mp3file_reset_buffer(audiomp3_mp3file_obj_t *self, + bool single_channel, + uint8_t channel) { if (single_channel && channel == 1) { return; } @@ -310,11 +313,11 @@ void audiomp3_mp3file_reset_buffer(audiomp3_mp3file_obj_t* self, background_callback_end_critical_section(); } -audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t* self, - bool single_channel, - uint8_t channel, - uint8_t** bufptr, - uint32_t* buffer_length) { +audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t *self, + bool single_channel, + uint8_t channel, + uint8_t **bufptr, + uint32_t *buffer_length) { if (!self->inbuf) { return GET_BUFFER_ERROR; } @@ -325,17 +328,17 @@ audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t* *buffer_length = self->frame_buffer_size; if (channel == self->other_channel) { - *bufptr = (uint8_t*)(self->buffers[self->other_buffer_index] + channel); + *bufptr = (uint8_t *)(self->buffers[self->other_buffer_index] + channel); self->other_channel = -1; return GET_BUFFER_MORE_DATA; } self->buffer_index = !self->buffer_index; - self->other_channel = 1-channel; + 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; + *bufptr = (uint8_t *)buffer; mp3file_skip_id3v2(self); if (!mp3file_find_sync_word(self)) { @@ -360,9 +363,9 @@ audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t* return GET_BUFFER_MORE_DATA; } -void audiomp3_mp3file_get_buffer_structure(audiomp3_mp3file_obj_t* self, bool single_channel, - bool* single_buffer, bool* samples_signed, - uint32_t* max_buffer_length, uint8_t* spacing) { +void audiomp3_mp3file_get_buffer_structure(audiomp3_mp3file_obj_t *self, bool single_channel, + 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; @@ -373,11 +376,11 @@ void audiomp3_mp3file_get_buffer_structure(audiomp3_mp3file_obj_t* self, bool si } } -float common_hal_audiomp3_mp3file_get_rms_level(audiomp3_mp3file_obj_t* self) { +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++) { + 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)); |
