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https://github.com/FFmpeg/FFmpeg.git
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mpegaudiodec: use planar sample format for output unless packed is requested
This commit is contained in:
Justin Ruggles
parent
2d3993ce8c
commit
3ffed68c2a
@ -93,7 +93,8 @@ typedef struct MPADecodeContext {
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# define MULH3(x, y, s) ((s)*(y)*(x))
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# define MULLx(x, y, s) ((y)*(x))
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# define RENAME(a) a ## _float
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# define OUT_FMT AV_SAMPLE_FMT_FLT
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# define OUT_FMT AV_SAMPLE_FMT_FLT
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# define OUT_FMT_P AV_SAMPLE_FMT_FLTP
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#else
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# define SHR(a,b) ((a)>>(b))
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/* WARNING: only correct for positive numbers */
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@ -103,7 +104,8 @@ typedef struct MPADecodeContext {
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# define MULH3(x, y, s) MULH((s)*(x), y)
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# define MULLx(x, y, s) MULL(x,y,s)
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# define RENAME(a) a ## _fixed
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# define OUT_FMT AV_SAMPLE_FMT_S16
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# define OUT_FMT AV_SAMPLE_FMT_S16
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# define OUT_FMT_P AV_SAMPLE_FMT_S16P
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#endif
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/****************/
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@ -434,7 +436,11 @@ static av_cold int decode_init(AVCodecContext * avctx)
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ff_mpadsp_init(&s->mpadsp);
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ff_dsputil_init(&s->dsp, avctx);
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avctx->sample_fmt= OUT_FMT;
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if (avctx->request_sample_fmt == OUT_FMT &&
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avctx->codec_id != AV_CODEC_ID_MP3ON4)
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avctx->sample_fmt = OUT_FMT;
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else
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avctx->sample_fmt = OUT_FMT_P;
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s->err_recognition = avctx->err_recognition;
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if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
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@ -1546,7 +1552,7 @@ static int mp_decode_layer3(MPADecodeContext *s)
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return nb_granules * 18;
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}
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static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
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static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,
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const uint8_t *buf, int buf_size)
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{
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int i, nb_frames, ch, ret;
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@ -1609,20 +1615,26 @@ static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
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av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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samples = (OUT_INT *)s->frame.data[0];
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samples = (OUT_INT **)s->frame.extended_data;
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}
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/* apply the synthesis filter */
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for (ch = 0; ch < s->nb_channels; ch++) {
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samples_ptr = samples + ch;
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int sample_stride;
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if (s->avctx->sample_fmt == OUT_FMT_P) {
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samples_ptr = samples[ch];
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sample_stride = 1;
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} else {
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samples_ptr = samples[0] + ch;
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sample_stride = s->nb_channels;
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}
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for (i = 0; i < nb_frames; i++) {
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RENAME(ff_mpa_synth_filter)(
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&s->mpadsp,
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s->synth_buf[ch], &(s->synth_buf_offset[ch]),
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RENAME(ff_mpa_synth_window), &s->dither_state,
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samples_ptr, s->nb_channels,
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s->sb_samples[ch][i]);
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samples_ptr += 32 * s->nb_channels;
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RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
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&(s->synth_buf_offset[ch]),
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RENAME(ff_mpa_synth_window),
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&s->dither_state, samples_ptr,
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sample_stride, s->sb_samples[ch][i]);
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samples_ptr += 32 * sample_stride;
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}
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}
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@ -1760,7 +1772,6 @@ typedef struct MP3On4DecodeContext {
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int syncword; ///< syncword patch
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const uint8_t *coff; ///< channel offsets in output buffer
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MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
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OUT_INT *decoded_buf; ///< output buffer for decoded samples
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} MP3On4DecodeContext;
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#include "mpeg4audio.h"
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@ -1802,8 +1813,6 @@ static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
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for (i = 0; i < s->frames; i++)
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av_free(s->mp3decctx[i]);
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av_freep(&s->decoded_buf);
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return 0;
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}
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@ -1864,14 +1873,6 @@ static int decode_init_mp3on4(AVCodecContext * avctx)
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s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
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}
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/* Allocate buffer for multi-channel output if needed */
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if (s->frames > 1) {
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s->decoded_buf = av_malloc(MPA_FRAME_SIZE * MPA_MAX_CHANNELS *
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sizeof(*s->decoded_buf));
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if (!s->decoded_buf)
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goto alloc_fail;
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}
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return 0;
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alloc_fail:
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decode_close_mp3on4(avctx);
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@ -1898,9 +1899,9 @@ static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
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MPADecodeContext *m;
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int fsize, len = buf_size, out_size = 0;
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uint32_t header;
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OUT_INT *out_samples;
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OUT_INT *outptr, *bp;
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int fr, j, n, ch, ret;
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OUT_INT **out_samples;
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OUT_INT *outptr[2];
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int fr, ch, ret;
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/* get output buffer */
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s->frame->nb_samples = MPA_FRAME_SIZE;
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@ -1908,15 +1909,12 @@ static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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out_samples = (OUT_INT *)s->frame->data[0];
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out_samples = (OUT_INT **)s->frame->extended_data;
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// Discard too short frames
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if (buf_size < HEADER_SIZE)
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return AVERROR_INVALIDDATA;
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// If only one decoder interleave is not needed
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outptr = s->frames == 1 ? out_samples : s->decoded_buf;
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avctx->bit_rate = 0;
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ch = 0;
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@ -1944,6 +1942,10 @@ static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
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}
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ch += m->nb_channels;
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outptr[0] = out_samples[s->coff[fr]];
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if (m->nb_channels > 1)
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outptr[1] = out_samples[s->coff[fr] + 1];
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if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
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return ret;
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@ -1951,23 +1953,6 @@ static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
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buf += fsize;
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len -= fsize;
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if (s->frames > 1) {
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n = m->avctx->frame_size*m->nb_channels;
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/* interleave output data */
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bp = out_samples + s->coff[fr];
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if (m->nb_channels == 1) {
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for (j = 0; j < n; j++) {
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*bp = s->decoded_buf[j];
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bp += avctx->channels;
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}
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} else {
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for (j = 0; j < n; j++) {
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bp[0] = s->decoded_buf[j++];
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bp[1] = s->decoded_buf[j];
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bp += avctx->channels;
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}
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}
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}
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avctx->bit_rate += m->bit_rate;
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}
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@ -1994,6 +1979,9 @@ AVCodec ff_mp1_decoder = {
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.capabilities = CODEC_CAP_DR1,
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.flush = flush,
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.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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#if CONFIG_MP2_DECODER
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@ -2007,6 +1995,9 @@ AVCodec ff_mp2_decoder = {
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.capabilities = CODEC_CAP_DR1,
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.flush = flush,
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.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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#if CONFIG_MP3_DECODER
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@ -2020,6 +2011,9 @@ AVCodec ff_mp3_decoder = {
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.capabilities = CODEC_CAP_DR1,
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.flush = flush,
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.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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#if CONFIG_MP3ADU_DECODER
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@ -2033,6 +2027,9 @@ AVCodec ff_mp3adu_decoder = {
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.capabilities = CODEC_CAP_DR1,
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.flush = flush,
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.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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#if CONFIG_MP3ON4_DECODER
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@ -2047,6 +2044,8 @@ AVCodec ff_mp3on4_decoder = {
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.capabilities = CODEC_CAP_DR1,
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.flush = flush_mp3on4,
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.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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#endif
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@ -33,6 +33,9 @@ AVCodec ff_mp1float_decoder = {
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.capabilities = CODEC_CAP_DR1,
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.flush = flush,
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.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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#if CONFIG_MP2FLOAT_DECODER
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@ -46,6 +49,9 @@ AVCodec ff_mp2float_decoder = {
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.capabilities = CODEC_CAP_DR1,
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.flush = flush,
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.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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#if CONFIG_MP3FLOAT_DECODER
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@ -59,6 +65,9 @@ AVCodec ff_mp3float_decoder = {
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.capabilities = CODEC_CAP_DR1,
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.flush = flush,
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.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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#if CONFIG_MP3ADUFLOAT_DECODER
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@ -72,6 +81,9 @@ AVCodec ff_mp3adufloat_decoder = {
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.capabilities = CODEC_CAP_DR1,
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.flush = flush,
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.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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#if CONFIG_MP3ON4FLOAT_DECODER
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@ -86,5 +98,7 @@ AVCodec ff_mp3on4float_decoder = {
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.capabilities = CODEC_CAP_DR1,
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.flush = flush_mp3on4,
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.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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