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authorSolomon Peachy <pizza@shaftnet.org>2018-12-22 20:01:42 -0500
committerSolomon Peachy <pizza@shaftnet.org>2018-12-22 20:12:10 -0500
commit928557bb174fdc6ae44d784137e19a61b4f42693 (patch)
tree37b6a3effd923b839c8fb02b806c43ea7506be9f /lib
parent9b9b30bd547c829157f3f83c71378f0bbd43241d (diff)
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AAC bitstream format files support
Files with extension "aac" in ADTS or ADIF format are now playable. Full credit goes to Igor Poretsky. Change-Id: I413b34e15e5242fea60d3461966ae0984080f530
Diffstat (limited to 'lib')
-rw-r--r--lib/rbcodec/SOURCES1
-rw-r--r--lib/rbcodec/codecs/SOURCES1
-rw-r--r--lib/rbcodec/codecs/aac_bsf.c157
-rw-r--r--lib/rbcodec/codecs/codecs.make1
-rw-r--r--lib/rbcodec/metadata/aac.c122
-rw-r--r--lib/rbcodec/metadata/metadata.c3
-rw-r--r--lib/rbcodec/metadata/metadata.h1
-rw-r--r--lib/rbcodec/metadata/metadata_parsers.h1
8 files changed, 287 insertions, 0 deletions
diff --git a/lib/rbcodec/SOURCES b/lib/rbcodec/SOURCES
index c2a32e9c28..9180237632 100644
--- a/lib/rbcodec/SOURCES
+++ b/lib/rbcodec/SOURCES
@@ -63,4 +63,5 @@ metadata/vorbis.c
metadata/vox.c
metadata/wave.c
metadata/wavpack.c
+metadata/aac.c
#endif
diff --git a/lib/rbcodec/codecs/SOURCES b/lib/rbcodec/codecs/SOURCES
index 039772cf9a..f0787d267d 100644
--- a/lib/rbcodec/codecs/SOURCES
+++ b/lib/rbcodec/codecs/SOURCES
@@ -42,6 +42,7 @@ vgm.c
#if MEMORYSIZE > 2
kss.c
#endif
+aac_bsf.c
#ifdef HAVE_RECORDING
diff --git a/lib/rbcodec/codecs/aac_bsf.c b/lib/rbcodec/codecs/aac_bsf.c
new file mode 100644
index 0000000000..3bce283958
--- /dev/null
+++ b/lib/rbcodec/codecs/aac_bsf.c
@@ -0,0 +1,157 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Codec for aac files without container
+ *
+ * Written by Igor B. Poretsky
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "codeclib.h"
+#include "libfaad/common.h"
+#include "libfaad/structs.h"
+#include "libfaad/decoder.h"
+
+CODEC_HEADER
+
+/* The maximum buffer size handled by faad. 12 bytes are required by libfaad
+ * as headroom (see libfaad/bits.c). FAAD_BYTE_BUFFER_SIZE bytes are buffered
+ * for each frame. */
+#define FAAD_BYTE_BUFFER_SIZE (2048-12)
+
+static void update_playing_time(void)
+{
+ ci->set_elapsed((unsigned long)((ci->id3->offset - ci->id3->first_frame_offset) * 8LL / ci->id3->bitrate));
+}
+
+/* this is the codec entry point */
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+ if (reason == CODEC_LOAD) {
+ /* Generic codec initialisation */
+ ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
+ ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
+ }
+
+ return CODEC_OK;
+}
+
+/* this is called for each file to process */
+enum codec_status codec_run(void)
+{
+ size_t n;
+ int32_t bread;
+ unsigned int frame_samples;
+ uint32_t s = 0;
+ unsigned char c = 0;
+ long action = CODEC_ACTION_NULL;
+ intptr_t param;
+ unsigned char* buffer;
+ NeAACDecFrameInfo frame_info;
+ NeAACDecHandle decoder;
+ NeAACDecConfigurationPtr conf;
+
+ /* Clean and initialize decoder structures */
+ if (codec_init()) {
+ LOGF("FAAD: Codec init error\n");
+ return CODEC_ERROR;
+ }
+
+ ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
+ codec_set_replaygain(ci->id3);
+
+ ci->seek_buffer(ci->id3->first_frame_offset);
+
+ /* initialise the sound converter */
+ decoder = NeAACDecOpen();
+
+ if (!decoder) {
+ LOGF("FAAD: Decode open error\n");
+ return CODEC_ERROR;
+ }
+
+ conf = NeAACDecGetCurrentConfiguration(decoder);
+ conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
+ NeAACDecSetConfiguration(decoder, conf);
+
+ buffer=ci->request_buffer(&n, FAAD_BYTE_BUFFER_SIZE);
+ bread = NeAACDecInit(decoder, buffer, n, &s, &c);
+ if (bread < 0) {
+ LOGF("FAAD: DecInit: %ld, %d\n", bread, decoder->object_type);
+ return CODEC_ERROR;
+ }
+ ci->advance_buffer(bread);
+
+ if (ci->id3->offset > ci->id3->first_frame_offset) {
+ /* Resume the desired (byte) position. */
+ ci->seek_buffer(ci->id3->offset);
+ NeAACDecPostSeekReset(decoder, 0);
+ update_playing_time();
+ } else if (ci->id3->elapsed) {
+ action = CODEC_ACTION_SEEK_TIME;
+ param = ci->id3->elapsed;
+ } else {
+ ci->set_elapsed(0);
+ ci->set_offset(ci->id3->first_frame_offset);
+ }
+
+ /* The main decoding loop */
+ while (1) {
+ if (action == CODEC_ACTION_NULL)
+ action = ci->get_command(&param);
+
+ if (action == CODEC_ACTION_HALT)
+ break;
+
+ /* Deal with any pending seek requests */
+ if (action == CODEC_ACTION_SEEK_TIME) {
+ /* Seek to the desired time position. */
+ ci->seek_buffer(ci->id3->first_frame_offset + (uint32_t)((uint64_t)param * ci->id3->bitrate / 8));
+ ci->set_elapsed((unsigned long)param);
+ NeAACDecPostSeekReset(decoder, 0);
+ ci->seek_complete();
+ }
+
+ action = CODEC_ACTION_NULL;
+
+ /* Request the required number of bytes from the input buffer */
+ buffer=ci->request_buffer(&n, FAAD_BYTE_BUFFER_SIZE);
+
+ if (n == 0) /* End of Stream */
+ break;
+
+ /* Decode one block - returned samples will be host-endian */
+ if (NeAACDecDecode(decoder, &frame_info, buffer, n) == NULL || frame_info.error > 0) {
+ LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
+ return CODEC_ERROR;
+ }
+
+ /* Advance codec buffer (no need to call set_offset because of this) */
+ ci->advance_buffer(frame_info.bytesconsumed);
+
+ /* Output the audio */
+ ci->yield();
+ frame_samples = frame_info.samples >> 1;
+ ci->pcmbuf_insert(&decoder->time_out[0][0], &decoder->time_out[1][0], frame_samples);
+
+ /* Update the elapsed-time indicator */
+ update_playing_time();
+ }
+
+ LOGF("AAC: Decoding complete\n");
+ return CODEC_OK;
+}
diff --git a/lib/rbcodec/codecs/codecs.make b/lib/rbcodec/codecs/codecs.make
index 8934272fcf..afb8396938 100644
--- a/lib/rbcodec/codecs/codecs.make
+++ b/lib/rbcodec/codecs/codecs.make
@@ -181,6 +181,7 @@ $(CODECDIR)/sgc.codec : $(CODECDIR)/libsgc.a $(CODECDIR)/libemu2413.a
$(CODECDIR)/vgm.codec : $(CODECDIR)/libvgm.a $(CODECDIR)/libemu2413.a
$(CODECDIR)/kss.codec : $(CODECDIR)/libkss.a $(CODECDIR)/libemu2413.a
$(CODECDIR)/opus.codec : $(CODECDIR)/libopus.a $(TLSFLIB)
+$(CODECDIR)/aac_bsf.codec : $(CODECDIR)/libfaad.a
$(CODECS): $(CODEC_LIBS) # this must be last in codec dependency list
diff --git a/lib/rbcodec/metadata/aac.c b/lib/rbcodec/metadata/aac.c
new file mode 100644
index 0000000000..82adeacbde
--- /dev/null
+++ b/lib/rbcodec/metadata/aac.c
@@ -0,0 +1,122 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Parsing ADTS and ADIF headers
+ *
+ * Written by Igor B. Poretsky
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include <stdlib.h>
+#include <stdbool.h>
+#include <string.h>
+
+#include "platform.h"
+
+#include "metadata.h"
+#include "metadata_common.h"
+#include "metadata_parsers.h"
+
+static const int sample_rates[] =
+{
+ 96000, 88200, 64000, 48000,
+ 44100, 32000, 24000, 22050,
+ 16000, 12000, 11025, 8000,
+ 7350, 0, 0, 0
+};
+
+static bool check_adts_syncword(int fd)
+{
+ uint16_t syncword;
+
+ read_uint16be(fd, &syncword);
+ return (syncword & 0xFFF6) == 0xFFF0;
+}
+
+bool get_aac_metadata(int fd, struct mp3entry *entry)
+{
+ unsigned char buf[5];
+
+ entry->title = NULL;
+ entry->tracknum = 0;
+ entry->discnum = 0;
+ entry->id3v1len = 0;
+ entry->id3v2len = getid3v2len(fd);
+ entry->first_frame_offset = entry->id3v2len;
+ entry->filesize = filesize(fd) - entry->first_frame_offset;
+ entry->needs_upsampling_correction = false;
+
+ if (entry->id3v2len)
+ setid3v2title(fd, entry);
+
+ if (-1 == lseek(fd, entry->first_frame_offset, SEEK_SET))
+ return false;
+
+ if (check_adts_syncword(fd))
+ {
+ int frames;
+ int stat_length;
+ uint64_t total;
+ if (read(fd, buf, 5) != 5)
+ return false;
+ entry->frequency = sample_rates[(buf[0] >> 2) & 0x0F];
+ entry->vbr = ((buf[3] & 0x1F) == 0x1F)
+ && ((buf[4] & 0xFC) == 0xFC);
+ stat_length = entry->frequency >> ((entry->vbr) ? 5 : 7);
+ for (frames = 1, total = 0; frames < stat_length; frames++)
+ {
+ unsigned int frame_length = (((unsigned int)buf[1] & 0x3) << 11)
+ | ((unsigned int)buf[2] << 3)
+ | ((unsigned int)buf[3] >> 5);
+ total += frame_length;
+ if (frame_length < 7)
+ break;
+ if (-1 == lseek(fd, frame_length - 7, SEEK_CUR))
+ break;
+ if (!check_adts_syncword(fd))
+ break;
+ if (read(fd, buf, 5) != 5)
+ break;
+ }
+ entry->bitrate = (unsigned int)((total * entry->frequency / frames + 64000) / 128000);
+ if (entry->frequency <= 24000)
+ {
+ entry->frequency <<= 1;
+ entry->needs_upsampling_correction = true;
+ }
+ }
+ else
+ {
+ uint32_t bitrate;
+ if (-1 == lseek(fd, entry->first_frame_offset, SEEK_SET))
+ return false;
+ if (read(fd, buf, 5) != 5)
+ return false;
+ if (memcmp(buf, "ADIF", 4))
+ return false;
+ if (-1 == lseek(fd, (buf[4] & 0x80) ? (entry->first_frame_offset + 9) : entry->first_frame_offset, SEEK_SET))
+ return false;
+ read_uint32be(fd, &bitrate);
+ entry->vbr = (bitrate & 0x10000000) != 0;
+ entry->bitrate = ((bitrate & 0xFFFFFE0) + 16000) / 32000;
+ read_uint32be(fd, (uint32_t*)(&(entry->frequency)));
+ entry->frequency = sample_rates[(entry->frequency >> (entry->vbr ? 23 : 3)) & 0x0F];
+ }
+ entry->length = (unsigned long)((entry->filesize * 8LL + (entry->bitrate >> 1)) / entry->bitrate);
+
+ return true;
+}
diff --git a/lib/rbcodec/metadata/metadata.c b/lib/rbcodec/metadata/metadata.c
index 7ca4b1afd2..c24a27df2b 100644
--- a/lib/rbcodec/metadata/metadata.c
+++ b/lib/rbcodec/metadata/metadata.c
@@ -235,6 +235,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
/* Opus */
[AFMT_OPUS] =
AFMT_ENTRY("Opus", "opus", NULL, get_ogg_metadata, "opus\0"),
+ /* AAC bitstream format */
+ [AFMT_AAC_BSF] =
+ AFMT_ENTRY("AAC", "aac_bsf", NULL, get_aac_metadata, "aac\0"),
#endif
};
diff --git a/lib/rbcodec/metadata/metadata.h b/lib/rbcodec/metadata/metadata.h
index 18cfce7523..a7ebcf16e5 100644
--- a/lib/rbcodec/metadata/metadata.h
+++ b/lib/rbcodec/metadata/metadata.h
@@ -90,6 +90,7 @@ enum
AFMT_VGM, /* VGM (Video Game Music Format) */
AFMT_KSS, /* KSS (MSX computer KSS Music File) */
AFMT_OPUS, /* Opus (see http://www.opus-codec.org ) */
+ AFMT_AAC_BSF,
#endif
/* add new formats at any index above this line to have a sensible order -
diff --git a/lib/rbcodec/metadata/metadata_parsers.h b/lib/rbcodec/metadata/metadata_parsers.h
index 304e393538..9f03c79bb5 100644
--- a/lib/rbcodec/metadata/metadata_parsers.h
+++ b/lib/rbcodec/metadata/metadata_parsers.h
@@ -56,4 +56,5 @@ bool get_hes_metadata(int fd, struct mp3entry* id3);
bool get_sgc_metadata(int fd, struct mp3entry* id3);
bool get_vgm_metadata(int fd, struct mp3entry* id3);
bool get_kss_metadata(int fd, struct mp3entry* id3);
+bool get_aac_metadata(int fd, struct mp3entry* id3);
#endif /* CONFIG_CODEC == SWCODEC */