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Diffstat (limited to 'libavcodec/atrac1.c')
-rw-r--r--libavcodec/atrac1.c93
1 files changed, 60 insertions, 33 deletions
diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c
index 2ad99bf473..d4d5986821 100644
--- a/libavcodec/atrac1.c
+++ b/libavcodec/atrac1.c
@@ -36,6 +36,7 @@
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
+#include "fmtconvert.h"
#include "sinewin.h"
#include "atrac.h"
@@ -78,10 +79,11 @@ typedef struct {
DECLARE_ALIGNED(32, float, mid)[256];
DECLARE_ALIGNED(32, float, high)[512];
float* bands[3];
- DECLARE_ALIGNED(32, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
+ float *out_samples[AT1_MAX_CHANNELS];
FFTContext mdct_ctx[3];
int channels;
DSPContext dsp;
+ FmtConvertContext fmt_conv;
} AT1Ctx;
/** size of the transform in samples in the long mode for each QMF band */
@@ -129,7 +131,7 @@ static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
nbits = mdct_long_nbits[band_num] - log2_block_count;
if (nbits != 5 && nbits != 7 && nbits != 8)
- return -1;
+ return AVERROR_INVALIDDATA;
} else {
block_size = 32;
nbits = 5;
@@ -173,14 +175,14 @@ static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
/* low and mid band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp & 1)
- return -1;
+ return AVERROR_INVALIDDATA;
log2_block_cnt[i] = 2 - log2_block_count_tmp;
}
/* high band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
- return -1;
+ return AVERROR_INVALIDDATA;
log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
skip_bits(gb, 2);
@@ -229,7 +231,7 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
/* check for bitstream overflow */
if (bits_used > AT1_SU_MAX_BITS)
- return -1;
+ return AVERROR_INVALIDDATA;
/* get the position of the 1st spec according to the block size mode */
pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
@@ -276,14 +278,21 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AT1Ctx *q = avctx->priv_data;
- int ch, ret, i;
+ int ch, ret, out_size;
GetBitContext gb;
float* samples = data;
if (buf_size < 212 * q->channels) {
- av_log(avctx, AV_LOG_ERROR,"Not enought data to decode!\n");
- return -1;
+ av_log(avctx,AV_LOG_ERROR,"Not enough data to decode!\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ out_size = q->channels * AT1_SU_SAMPLES *
+ 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);
}
for (ch = 0; ch < q->channels; ch++) {
@@ -303,44 +312,72 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
ret = at1_imdct_block(su, q);
if (ret < 0)
return ret;
- at1_subband_synthesis(q, su, q->out_samples[ch]);
+ at1_subband_synthesis(q, su, q->channels == 1 ? samples : q->out_samples[ch]);
}
- /* interleave; FIXME, should create/use a DSP function */
- if (q->channels == 1) {
- /* mono */
- memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4);
- } else {
- /* stereo */
- for (i = 0; i < AT1_SU_SAMPLES; i++) {
- samples[i * 2] = q->out_samples[0][i];
- samples[i * 2 + 1] = q->out_samples[1][i];
- }
+ /* interleave */
+ if (q->channels == 2) {
+ q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
+ AT1_SU_SAMPLES, 2);
}
- *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
+ *data_size = out_size;
return avctx->block_align;
}
+static av_cold int atrac1_decode_end(AVCodecContext * avctx)
+{
+ AT1Ctx *q = avctx->priv_data;
+
+ av_freep(&q->out_samples[0]);
+
+ ff_mdct_end(&q->mdct_ctx[0]);
+ ff_mdct_end(&q->mdct_ctx[1]);
+ ff_mdct_end(&q->mdct_ctx[2]);
+
+ return 0;
+}
+
+
static av_cold int atrac1_decode_init(AVCodecContext *avctx)
{
AT1Ctx *q = avctx->priv_data;
+ int ret;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
+ avctx->channels);
+ return AVERROR(EINVAL);
+ }
q->channels = avctx->channels;
+ if (avctx->channels == 2) {
+ q->out_samples[0] = av_malloc(2 * AT1_SU_SAMPLES * sizeof(*q->out_samples[0]));
+ q->out_samples[1] = q->out_samples[0] + AT1_SU_SAMPLES;
+ if (!q->out_samples[0]) {
+ av_freep(&q->out_samples[0]);
+ return AVERROR(ENOMEM);
+ }
+ }
+
/* Init the mdct transforms */
- ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
- ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
- ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
+ if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
+ (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) ||
+ (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) {
+ av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
+ atrac1_decode_end(avctx);
+ return ret;
+ }
ff_init_ff_sine_windows(5);
atrac_generate_tables();
dsputil_init(&q->dsp, avctx);
+ ff_fmt_convert_init(&q->fmt_conv, avctx);
q->bands[0] = q->low;
q->bands[1] = q->mid;
@@ -356,16 +393,6 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
}
-static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
- AT1Ctx *q = avctx->priv_data;
-
- ff_mdct_end(&q->mdct_ctx[0]);
- ff_mdct_end(&q->mdct_ctx[1]);
- ff_mdct_end(&q->mdct_ctx[2]);
- return 0;
-}
-
-
AVCodec ff_atrac1_decoder = {
.name = "atrac1",
.type = AVMEDIA_TYPE_AUDIO,
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-23 02:38:25 +0000