#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
+#include "fmtconvert.h"
#include "sinewin.h"
#include "atrac.h"
* The atrac1 context, holds all needed parameters for decoding
*/
typedef struct {
+ AVFrame frame;
AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
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 */
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;
/* 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);
/* 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];
float iqmf_temp[512 + 46];
/* combine low and middle bands */
- atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
+ ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
/* delay the signal of the high band by 23 samples */
memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23);
memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256);
/* combine (low + middle) and high bands */
- atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
+ ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
}
static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
- int *data_size, AVPacket *avpkt)
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AT1Ctx *q = avctx->priv_data;
- int ch, ret, i, out_size;
+ int ch, ret;
GetBitContext gb;
- float* samples = data;
+ float *samples;
if (buf_size < 212 * q->channels) {
- av_log(q,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);
+ /* get output buffer */
+ q->frame.nb_samples = AT1_SU_SAMPLES;
+ if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
}
+ samples = (float *)q->frame.data[0];
for (ch = 0; ch < q->channels; ch++) {
AT1SUCtx* su = &q->SUs[ch];
at1_subband_synthesis(q, su, q->channels == 1 ? samples : q->out_samples[ch]);
}
- /* interleave; FIXME, should create/use a DSP function */
+ /* interleave */
if (q->channels == 2) {
- 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];
- }
+ q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
+ AT1_SU_SAMPLES, 2);
}
- *data_size = out_size;
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = q->frame;
+
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;
}
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();
+ ff_atrac_generate_tables();
- dsputil_init(&q->dsp, avctx);
+ ff_dsputil_init(&q->dsp, avctx);
+ ff_fmt_convert_init(&q->fmt_conv, avctx);
q->bands[0] = q->low;
q->bands[1] = q->mid;
q->SUs[1].spectrum[0] = q->SUs[1].spec1;
q->SUs[1].spectrum[1] = q->SUs[1].spec2;
- return 0;
-}
+ avcodec_get_frame_defaults(&q->frame);
+ avctx->coded_frame = &q->frame;
-
-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;
}
.init = atrac1_decode_init,
.close = atrac1_decode_end,
.decode = atrac1_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
};
projects / ffmpeg / blobdiff