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-rw-r--r--lib/rbcodec/codecs/aac.c297
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diff --git a/lib/rbcodec/codecs/aac.c b/lib/rbcodec/codecs/aac.c
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+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Dave Chapman
+ *
+ * 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 "libm4a/m4a.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)
+
+/* 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)
+{
+ /* Note that when dealing with QuickTime/MPEG4 files, terminology is
+ * a bit confusing. Files with sound are split up in chunks, where
+ * each chunk contains one or more samples. Each sample in turn
+ * contains a number of "sound samples" (the kind you refer to with
+ * the sampling frequency).
+ */
+ size_t n;
+ demux_res_t demux_res;
+ stream_t input_stream;
+ uint32_t sound_samples_done;
+ uint32_t elapsed_time;
+ int file_offset;
+ int framelength;
+ int lead_trim = 0;
+ unsigned int frame_samples;
+ unsigned int i;
+ unsigned char* buffer;
+ NeAACDecFrameInfo frame_info;
+ NeAACDecHandle decoder;
+ int err;
+ uint32_t seek_idx = 0;
+ uint32_t s = 0;
+ uint32_t sbr_fac = 1;
+ unsigned char c = 0;
+ void *ret;
+ intptr_t param;
+ bool empty_first_frame = false;
+
+ /* Clean and initialize decoder structures */
+ memset(&demux_res , 0, sizeof(demux_res));
+ if (codec_init()) {
+ LOGF("FAAD: Codec init error\n");
+ return CODEC_ERROR;
+ }
+
+ file_offset = ci->id3->offset;
+
+ ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
+ codec_set_replaygain(ci->id3);
+
+ stream_create(&input_stream,ci);
+
+ ci->seek_buffer(ci->id3->first_frame_offset);
+
+ /* if qtmovie_read returns successfully, the stream is up to
+ * the movie data, which can be used directly by the decoder */
+ if (!qtmovie_read(&input_stream, &demux_res)) {
+ LOGF("FAAD: File init error\n");
+ return CODEC_ERROR;
+ }
+
+ /* initialise the sound converter */
+ decoder = NeAACDecOpen();
+
+ if (!decoder) {
+ LOGF("FAAD: Decode open error\n");
+ return CODEC_ERROR;
+ }
+
+ NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
+ conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
+ NeAACDecSetConfiguration(decoder, conf);
+
+ err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
+ if (err) {
+ LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
+ return CODEC_ERROR;
+ }
+
+#ifdef SBR_DEC
+ /* Check for need of special handling for seek/resume and elapsed time. */
+ if (ci->id3->needs_upsampling_correction) {
+ sbr_fac = 2;
+ } else {
+ sbr_fac = 1;
+ }
+#endif
+
+ i = 0;
+
+ if (file_offset > 0) {
+ /* Resume the desired (byte) position. Important: When resuming SBR
+ * upsampling files the resulting sound_samples_done must be expanded
+ * by a factor of 2. This is done via using sbr_fac. */
+ if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
+ &sound_samples_done, (int*) &i)) {
+ sound_samples_done *= sbr_fac;
+ } else {
+ sound_samples_done = 0;
+ }
+ NeAACDecPostSeekReset(decoder, i);
+ } else {
+ sound_samples_done = 0;
+ }
+
+ elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
+ ci->set_elapsed(elapsed_time);
+
+ if (i == 0)
+ {
+ lead_trim = ci->id3->lead_trim;
+ }
+
+ /* The main decoding loop */
+ while (i < demux_res.num_sample_byte_sizes) {
+ enum codec_command_action 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. Important: When seeking in SBR
+ * upsampling files the seek_time must be divided by 2 when calling
+ * m4a_seek and the resulting sound_samples_done must be expanded
+ * by a factor 2. This is done via using sbr_fac. */
+ if (m4a_seek(&demux_res, &input_stream,
+ (param/10/sbr_fac)*(ci->id3->frequency/100),
+ &sound_samples_done, (int*) &i)) {
+ sound_samples_done *= sbr_fac;
+ elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
+ ci->set_elapsed(elapsed_time);
+ seek_idx = 0;
+
+ if (i == 0)
+ {
+ lead_trim = ci->id3->lead_trim;
+ }
+ }
+ NeAACDecPostSeekReset(decoder, i);
+ ci->seek_complete();
+ }
+
+ /* There can be gaps between chunks, so skip ahead if needed. It
+ * doesn't seem to happen much, but it probably means that a
+ * "proper" file can have chunks out of order. Why one would want
+ * that an good question (but files with gaps do exist, so who
+ * knows?), so we don't support that - for now, at least.
+ */
+ file_offset = m4a_check_sample_offset(&demux_res, i, &seek_idx);
+
+ if (file_offset > ci->curpos)
+ {
+ ci->advance_buffer(file_offset - ci->curpos);
+ }
+ else if (file_offset == 0)
+ {
+ LOGF("AAC: get_sample_offset error\n");
+ return CODEC_ERROR;
+ }
+
+ /* Request the required number of bytes from the input buffer */
+ buffer=ci->request_buffer(&n, FAAD_BYTE_BUFFER_SIZE);
+
+ /* Decode one block - returned samples will be host-endian */
+ ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
+
+ /* NeAACDecDecode may sometimes return NULL without setting error. */
+ if (ret == 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;
+
+ if (empty_first_frame)
+ {
+ /* Remove the first frame from lead_trim, under the assumption
+ * that it had the same size as this frame
+ */
+ empty_first_frame = false;
+ lead_trim -= frame_samples;
+
+ if (lead_trim < 0)
+ {
+ lead_trim = 0;
+ }
+ }
+
+ /* Gather number of samples for the decoded frame. */
+ framelength = frame_samples - lead_trim;
+
+ if (i == demux_res.num_sample_byte_sizes - 1)
+ {
+ // Size of the last frame
+ const uint32_t sample_duration = (demux_res.num_time_to_samples > 0) ?
+ demux_res.time_to_sample[demux_res.num_time_to_samples - 1].sample_duration :
+ frame_samples;
+
+ /* Currently limited to at most one frame of tail_trim.
+ * Seems to be enough.
+ */
+ if (ci->id3->tail_trim == 0 && sample_duration < frame_samples)
+ {
+ /* Subtract lead_trim just in case we decode a file with only
+ * one audio frame with actual data (lead_trim is usually zero
+ * here).
+ */
+ framelength = sample_duration - lead_trim;
+ }
+ else
+ {
+ framelength -= ci->id3->tail_trim;
+ }
+ }
+
+ if (framelength > 0)
+ {
+ ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
+ &decoder->time_out[1][lead_trim],
+ framelength);
+ sound_samples_done += framelength;
+ /* Update the elapsed-time indicator */
+ elapsed_time = ((uint64_t) sound_samples_done * 1000) /
+ ci->id3->frequency;
+ ci->set_elapsed(elapsed_time);
+ }
+
+ if (lead_trim > 0)
+ {
+ /* frame_info.samples can be 0 for frame 0. We still want to
+ * remove it from lead_trim, so do that during frame 1.
+ */
+ if (0 == i && 0 == frame_info.samples)
+ {
+ empty_first_frame = true;
+ }
+
+ lead_trim -= frame_samples;
+
+ if (lead_trim < 0)
+ {
+ lead_trim = 0;
+ }
+ }
+
+ ++i;
+ }
+
+ LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
+ return CODEC_OK;
+}