---
libavcodec/mpegaudiodec.c | 67 +++++++++++++-------------------------
libavcodec/mpegaudiodec_float.c | 10 ++++++
2 files changed, 33 insertions(+), 44 deletions(-)
diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c
index 03094f6..2fb0e9d 100644
--- a/libavcodec/mpegaudiodec.c
+++ b/libavcodec/mpegaudiodec.c
@@ -93,7 +93,7 @@ typedef struct MPADecodeContext {
# define MULH3(x, y, s) ((s)*(y)*(x))
# define MULLx(x, y, s) ((y)*(x))
# define RENAME(a) a ## _float
-# define OUT_FMT AV_SAMPLE_FMT_FLT
+# define OUT_FMT AV_SAMPLE_FMT_FLTP
#else
# define SHR(a,b) ((a)>>(b))
/* WARNING: only correct for positive numbers */
@@ -103,7 +103,7 @@ typedef struct MPADecodeContext {
# define MULH3(x, y, s) MULH((s)*(x), y)
# define MULLx(x, y, s) MULL(x,y,s)
# define RENAME(a) a ## _fixed
-# define OUT_FMT AV_SAMPLE_FMT_S16
+# define OUT_FMT AV_SAMPLE_FMT_S16P
#endif
/****************/
@@ -1550,11 +1550,10 @@ static int mp_decode_layer3(MPADecodeContext *s)
return nb_granules * 18;
}
-static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
+static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,
const uint8_t *buf, int buf_size)
{
int i, nb_frames, ch, ret;
- OUT_INT *samples_ptr;
init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
@@ -1610,20 +1609,17 @@ static int mp_decode_frame(MPADecodeContext *s, OUT_INT
*samples,
av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- samples = (OUT_INT *)s->frame.data[0];
+ samples = (OUT_INT **)s->frame.extended_data;
}
/* apply the synthesis filter */
for (ch = 0; ch < s->nb_channels; ch++) {
- samples_ptr = samples + ch;
for (i = 0; i < nb_frames; i++) {
RENAME(ff_mpa_synth_filter)(
&s->mpadsp,
s->synth_buf[ch], &(s->synth_buf_offset[ch]),
RENAME(ff_mpa_synth_window), &s->dither_state,
- samples_ptr, s->nb_channels,
- s->sb_samples[ch][i]);
- samples_ptr += 32 * s->nb_channels;
+ samples[ch] + 32 * i, 1, s->sb_samples[ch][i]);
}
}
@@ -1756,7 +1752,6 @@ typedef struct MP3On4DecodeContext {
int syncword; ///< syncword patch
const uint8_t *coff; ///< channel offsets in output buffer
MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder
instance
- OUT_INT *decoded_buf; ///< output buffer for decoded samples
} MP3On4DecodeContext;
#include "mpeg4audio.h"
@@ -1798,8 +1793,6 @@ static av_cold int decode_close_mp3on4(AVCodecContext *
avctx)
for (i = 0; i < s->frames; i++)
av_free(s->mp3decctx[i]);
- av_freep(&s->decoded_buf);
-
return 0;
}
@@ -1860,14 +1853,6 @@ static int decode_init_mp3on4(AVCodecContext * avctx)
s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
}
- /* Allocate buffer for multi-channel output if needed */
- if (s->frames > 1) {
- s->decoded_buf = av_malloc(MPA_FRAME_SIZE * MPA_MAX_CHANNELS *
- sizeof(*s->decoded_buf));
- if (!s->decoded_buf)
- goto alloc_fail;
- }
-
return 0;
alloc_fail:
decode_close_mp3on4(avctx);
@@ -1897,9 +1882,9 @@ static int decode_frame_mp3on4(AVCodecContext *avctx,
void *data,
MPADecodeContext *m;
int fsize, len = buf_size, out_size = 0;
uint32_t header;
- OUT_INT *out_samples;
- OUT_INT *outptr, *bp;
- int fr, j, n, ch, ret;
+ OUT_INT **out_samples;
+ OUT_INT *outptr[2];
+ int fr, ch, ret;
/* get output buffer */
s->frame->nb_samples = MPA_FRAME_SIZE;
@@ -1907,15 +1892,12 @@ static int decode_frame_mp3on4(AVCodecContext *avctx,
void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- out_samples = (OUT_INT *)s->frame->data[0];
+ out_samples = (OUT_INT **)s->frame->extended_data;
// Discard too short frames
if (buf_size < HEADER_SIZE)
return AVERROR_INVALIDDATA;
- // If only one decoder interleave is not needed
- outptr = s->frames == 1 ? out_samples : s->decoded_buf;
-
avctx->bit_rate = 0;
ch = 0;
@@ -1943,27 +1925,14 @@ static int decode_frame_mp3on4(AVCodecContext *avctx,
void *data,
}
ch += m->nb_channels;
+ outptr[0] = out_samples[s->coff[fr] ];
+ if (m->nb_channels > 1)
+ outptr[1] = out_samples[s->coff[fr] + 1];
+
out_size += mp_decode_frame(m, outptr, buf, fsize);
buf += fsize;
len -= fsize;
- if (s->frames > 1) {
- n = m->avctx->frame_size*m->nb_channels;
- /* interleave output data */
- bp = out_samples + s->coff[fr];
- if (m->nb_channels == 1) {
- for (j = 0; j < n; j++) {
- *bp = s->decoded_buf[j];
- bp += avctx->channels;
- }
- } else {
- for (j = 0; j < n; j++) {
- bp[0] = s->decoded_buf[j++];
- bp[1] = s->decoded_buf[j];
- bp += avctx->channels;
- }
- }
- }
avctx->bit_rate += m->bit_rate;
}
@@ -1990,6 +1959,8 @@ AVCodec ff_mp1_decoder = {
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP2_DECODER
@@ -2003,6 +1974,8 @@ AVCodec ff_mp2_decoder = {
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3_DECODER
@@ -2016,6 +1989,8 @@ AVCodec ff_mp3_decoder = {
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3ADU_DECODER
@@ -2029,6 +2004,8 @@ AVCodec ff_mp3adu_decoder = {
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3
(MPEG audio layer 3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3ON4_DECODER
@@ -2043,6 +2020,8 @@ AVCodec ff_mp3on4_decoder = {
.capabilities = CODEC_CAP_DR1,
.flush = flush_mp3on4,
.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#endif
diff --git a/libavcodec/mpegaudiodec_float.c b/libavcodec/mpegaudiodec_float.c
index 93468f5..73eefbb 100644
--- a/libavcodec/mpegaudiodec_float.c
+++ b/libavcodec/mpegaudiodec_float.c
@@ -33,6 +33,8 @@ AVCodec ff_mp1float_decoder = {
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP2FLOAT_DECODER
@@ -46,6 +48,8 @@ AVCodec ff_mp2float_decoder = {
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3FLOAT_DECODER
@@ -59,6 +63,8 @@ AVCodec ff_mp3float_decoder = {
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3ADUFLOAT_DECODER
@@ -72,6 +78,8 @@ AVCodec ff_mp3adufloat_decoder = {
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3
(MPEG audio layer 3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3ON4FLOAT_DECODER
@@ -86,5 +94,7 @@ AVCodec ff_mp3on4float_decoder = {
.capabilities = CODEC_CAP_DR1,
.flush = flush_mp3on4,
.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
#endif
--
1.7.1
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