* Copyright (c) 2006-2008 Maxim Poliakovski
* Copyright (c) 2006-2008 Benjamin Larsson
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
} channel_unit;
typedef struct {
+ AVFrame frame;
GetBitContext gb;
//@{
/** stream data */
}
-static av_cold int init_atrac3_transforms(ATRAC3Context *q) {
+static av_cold int init_atrac3_transforms(ATRAC3Context *q, int is_float) {
float enc_window[256];
int i;
}
/* Initialize the MDCT transform. */
- return ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768);
+ return ff_mdct_init(&q->mdct_ctx, 9, 1, is_float ? 1.0 / 32768 : 1.0);
}
/**
for (k=0; k<coded_components; k++) {
sfIndx = get_bits(gb,6);
+ if(component_count>=64)
+ return AVERROR_INVALIDDATA;
pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
max_coded_values = SAMPLES_PER_FRAME - pComponent[component_count].pos;
coded_values = coded_values_per_component + 1;
memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
/* gain compensation and overlapping */
- gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),
- &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),
- &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));
+ gainCompensateAndOverlap(pSnd->IMDCT_buf, &pSnd->prevFrame[band * 256],
+ &pOut[band * 256],
+ &pSnd->gainBlock[1 - pSnd->gcBlkSwitch].gBlock[band],
+ &pSnd->gainBlock[ pSnd->gcBlkSwitch].gBlock[band]);
}
/* Swap the gain control buffers for the next frame. */
result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, out_samples[0], 0, JOINT_STEREO);
if (result != 0)
- return (result);
+ return result;
/* Framedata of the su2 in the joint-stereo mode is encoded in
* reverse byte order so we need to swap it first. */
/* Decode Sound Unit 2. */
result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], out_samples[1], 1, JOINT_STEREO);
if (result != 0)
- return (result);
+ return result;
/* Reconstruct the channel coefficients. */
reverseMatrixing(out_samples[0], out_samples[1], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
for (i=0 ; i<q->channels ; i++) {
/* Set the bitstream reader at the start of a channel sound unit. */
- init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);
+ init_get_bits(&q->gb,
+ databuf + i * q->bytes_per_frame / q->channels,
+ q->bits_per_frame / q->channels);
result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], out_samples[i], i, q->codingMode);
if (result != 0)
- return (result);
+ return result;
}
}
* @param avctx pointer to the AVCodecContext
*/
-static int atrac3_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt) {
+static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
ATRAC3Context *q = avctx->priv_data;
- int result = 0, out_size;
+ int result;
const uint8_t* databuf;
- float *samples = data;
+ float *samples_flt;
+ int16_t *samples_s16;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR,
return AVERROR_INVALIDDATA;
}
- out_size = SAMPLES_PER_FRAME * q->channels *
- av_get_bytes_per_sample(avctx->sample_fmt);
- if (*data_size < out_size) {
- av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
- return AVERROR(EINVAL);
+ /* get output buffer */
+ q->frame.nb_samples = SAMPLES_PER_FRAME;
+ if ((result = avctx->get_buffer(avctx, &q->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return result;
}
+ samples_flt = (float *)q->frame.data[0];
+ samples_s16 = (int16_t *)q->frame.data[0];
/* Check if we need to descramble and what buffer to pass on. */
if (q->scrambled_stream) {
databuf = buf;
}
- result = decodeFrame(q, databuf, q->channels == 2 ? q->outSamples : &samples);
+ if (q->channels == 1 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
+ result = decodeFrame(q, databuf, &samples_flt);
+ else
+ result = decodeFrame(q, databuf, q->outSamples);
if (result != 0) {
av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
}
/* interleave */
- if (q->channels == 2) {
- q->fmt_conv.float_interleave(samples, (const float **)q->outSamples,
+ if (q->channels == 2 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+ q->fmt_conv.float_interleave(samples_flt,
+ (const float **)q->outSamples,
SAMPLES_PER_FRAME, 2);
+ } else if (avctx->sample_fmt == AV_SAMPLE_FMT_S16) {
+ q->fmt_conv.float_to_int16_interleave(samples_s16,
+ (const float **)q->outSamples,
+ SAMPLES_PER_FRAME, q->channels);
}
- *data_size = out_size;
+
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = q->frame;
return avctx->block_align;
}
vlcs_initialized = 1;
}
- if ((ret = init_atrac3_transforms(q))) {
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT)
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ else
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+
+ if ((ret = init_atrac3_transforms(q, avctx->sample_fmt == AV_SAMPLE_FMT_FLT))) {
av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
av_freep(&q->decoded_bytes_buffer);
return ret;
return AVERROR(ENOMEM);
}
- if (avctx->channels > 1) {
- q->outSamples[0] = av_mallocz(SAMPLES_PER_FRAME * 2 * sizeof(*q->outSamples[0]));
+ if (avctx->channels > 1 || avctx->sample_fmt == AV_SAMPLE_FMT_S16) {
+ q->outSamples[0] = av_mallocz(SAMPLES_PER_FRAME * avctx->channels * sizeof(*q->outSamples[0]));
q->outSamples[1] = q->outSamples[0] + SAMPLES_PER_FRAME;
if (!q->outSamples[0]) {
atrac3_decode_close(avctx);
}
}
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ avcodec_get_frame_defaults(&q->frame);
+ avctx->coded_frame = &q->frame;
+
return 0;
}
.init = atrac3_decode_init,
.close = atrac3_decode_close,
.decode = atrac3_decode_frame,
+ .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
};
projects / ffmpeg / blobdiff