[FFmpeg-cvslog] Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis
Justin Ruggles
git at videolan.org
Thu May 19 06:02:21 CEST 2011
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Fri Apr 22 21:30:19 2011 -0400| [9aa8193a234ccb6a79cba5cc550531f62ffb0a17] | committer: Justin Ruggles
Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis
decoders.
Based on patches by clsid2 in ffdshow-tryout.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=9aa8193a234ccb6a79cba5cc550531f62ffb0a17
---
libavcodec/aacdec.c | 34 ++++++++++++++++++++++++----------
libavcodec/aacsbr.c | 11 ++++++++---
libavcodec/aacsbr.h | 2 +-
libavcodec/ac3dec.c | 32 ++++++++++++++++++++++++++------
libavcodec/dca.c | 34 +++++++++++++++++++++++++++-------
libavcodec/vorbisdec.c | 23 ++++++++++++++++++-----
6 files changed, 104 insertions(+), 32 deletions(-)
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 5f9dd83..f2d50f4 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -186,7 +186,7 @@ static av_cold int che_configure(AACContext *ac,
if (che_pos[type][id]) {
if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
return AVERROR(ENOMEM);
- ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
+ ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
if (type != TYPE_CCE) {
ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
if (type == TYPE_CPE ||
@@ -546,6 +546,7 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
static av_cold int aac_decode_init(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
+ float output_scale_factor;
ac->avctx = avctx;
ac->m4ac.sample_rate = avctx->sample_rate;
@@ -557,7 +558,13 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
return -1;
}
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ output_scale_factor = 1.0 / 32768.0;
+ } else {
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ output_scale_factor = 1.0;
+ }
AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270);
@@ -585,9 +592,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
352);
- ff_mdct_init(&ac->mdct, 11, 1, 1.0/1024.0);
- ff_mdct_init(&ac->mdct_small, 8, 1, 1.0/128.0);
- ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0);
+ ff_mdct_init(&ac->mdct, 11, 1, output_scale_factor/1024.0);
+ ff_mdct_init(&ac->mdct_small, 8, 1, output_scale_factor/128.0);
+ ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0/output_scale_factor);
// window initialization
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
@@ -2169,7 +2176,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
avctx->frame_size = samples;
}
- data_size_tmp = samples * avctx->channels * sizeof(int16_t);
+ data_size_tmp = samples * avctx->channels *
+ (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
if (*data_size < data_size_tmp) {
av_log(avctx, AV_LOG_ERROR,
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
@@ -2178,8 +2186,14 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
}
*data_size = data_size_tmp;
- if (samples)
- ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
+ if (samples) {
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
+ ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
+ samples, avctx->channels);
+ else
+ ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
+ samples, avctx->channels);
+ }
if (ac->output_configured)
ac->output_configured = OC_LOCKED;
@@ -2497,7 +2511,7 @@ AVCodec ff_aac_decoder = {
aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.sample_fmts = (const enum AVSampleFormat[]) {
- AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.channel_layouts = aac_channel_layout,
};
@@ -2517,7 +2531,7 @@ AVCodec ff_aac_latm_decoder = {
.decode = latm_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
.sample_fmts = (const enum AVSampleFormat[]) {
- AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.channel_layouts = aac_channel_layout,
};
diff --git a/libavcodec/aacsbr.c b/libavcodec/aacsbr.c
index 7a217ab..81b0b4c 100644
--- a/libavcodec/aacsbr.c
+++ b/libavcodec/aacsbr.c
@@ -126,14 +126,19 @@ av_cold void ff_aac_sbr_init(void)
ff_ps_init();
}
-av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr)
+av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
{
+ float mdct_scale;
sbr->kx[0] = sbr->kx[1] = 32; //Typo in spec, kx' inits to 32
sbr->data[0].e_a[1] = sbr->data[1].e_a[1] = -1;
sbr->data[0].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
sbr->data[1].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
- ff_mdct_init(&sbr->mdct, 7, 1, 1.0/64);
- ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0);
+ /* SBR requires samples to be scaled to +/-32768.0 to work correctly.
+ * mdct scale factors are adjusted to scale up from +/-1.0 at analysis
+ * and scale back down at synthesis. */
+ mdct_scale = ac->avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? 32768.0f : 1.0f;
+ ff_mdct_init(&sbr->mdct, 7, 1, 1.0 / (64 * mdct_scale));
+ ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * mdct_scale);
ff_ps_ctx_init(&sbr->ps);
}
diff --git a/libavcodec/aacsbr.h b/libavcodec/aacsbr.h
index dca8330..153070d 100644
--- a/libavcodec/aacsbr.h
+++ b/libavcodec/aacsbr.h
@@ -36,7 +36,7 @@
/** Initialize SBR. */
av_cold void ff_aac_sbr_init(void);
/** Initialize one SBR context. */
-av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr);
+av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr);
/** Close one SBR context. */
av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr);
/** Decode one SBR element. */
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index 015ebae..2966c33 100644
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -189,7 +189,13 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
av_lfg_init(&s->dith_state, 0);
/* set scale value for float to int16 conversion */
- s->mul_bias = 32767.0f;
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+ s->mul_bias = 1.0f;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ } else {
+ s->mul_bias = 32767.0f;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ }
/* allow downmixing to stereo or mono */
if (avctx->channels > 0 && avctx->request_channels > 0 &&
@@ -204,7 +210,6 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
if (!s->input_buffer)
return AVERROR(ENOMEM);
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}
@@ -1299,7 +1304,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
- int16_t *out_samples = (int16_t *)data;
+ float *out_samples_flt = data;
+ int16_t *out_samples_s16 = data;
int blk, ch, err;
const uint8_t *channel_map;
const float *output[AC3_MAX_CHANNELS];
@@ -1405,10 +1411,18 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
err = 1;
}
- s->fmt_conv.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
- out_samples += 256 * s->out_channels;
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+ s->fmt_conv.float_interleave(out_samples_flt, output, 256,
+ s->out_channels);
+ out_samples_flt += 256 * s->out_channels;
+ } else {
+ s->fmt_conv.float_to_int16_interleave(out_samples_s16, output, 256,
+ s->out_channels);
+ out_samples_s16 += 256 * s->out_channels;
+ }
}
- *data_size = s->num_blocks * 256 * avctx->channels * sizeof (int16_t);
+ *data_size = s->num_blocks * 256 * avctx->channels *
+ (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
return FFMIN(buf_size, s->frame_size);
}
@@ -1435,6 +1449,9 @@ AVCodec ff_ac3_decoder = {
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+ },
};
#if CONFIG_EAC3_DECODER
@@ -1447,5 +1464,8 @@ AVCodec ff_eac3_decoder = {
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+ },
};
#endif
diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index f1cd64e..dbadeba 100644
--- a/libavcodec/dca.c
+++ b/libavcodec/dca.c
@@ -1626,7 +1626,9 @@ static int dca_decode_frame(AVCodecContext * avctx,
int lfe_samples;
int num_core_channels = 0;
int i;
- int16_t *samples = data;
+ float *samples_flt = data;
+ int16_t *samples_s16 = data;
+ int out_size;
DCAContext *s = avctx->priv_data;
int channels;
int core_ss_end;
@@ -1812,9 +1814,11 @@ static int dca_decode_frame(AVCodecContext * avctx,
return -1;
}
- if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
+ out_size = 256 / 8 * s->sample_blocks * channels *
+ (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
+ if (*data_size < out_size)
return -1;
- *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
+ *data_size = out_size;
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
@@ -1833,8 +1837,16 @@ static int dca_decode_frame(AVCodecContext * avctx,
}
}
- s->fmt_conv.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
- samples += 256 * channels;
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+ s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,
+ channels);
+ samples_flt += 256 * channels;
+ } else {
+ s->fmt_conv.float_to_int16_interleave(samples_s16,
+ s->samples_chanptr, 256,
+ channels);
+ samples_s16 += 256 * channels;
+ }
}
/* update lfe history */
@@ -1870,9 +1882,14 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- s->scale_bias = 1.0;
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ s->scale_bias = 1.0 / 32768.0;
+ } else {
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ s->scale_bias = 1.0;
+ }
/* allow downmixing to stereo */
if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
@@ -1909,5 +1926,8 @@ AVCodec ff_dca_decoder = {
.close = dca_decode_end,
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.capabilities = CODEC_CAP_CHANNEL_CONF,
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+ },
.profiles = NULL_IF_CONFIG_SMALL(profiles),
};
diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c
index 7443e98..f6ec74f 100644
--- a/libavcodec/vorbisdec.c
+++ b/libavcodec/vorbisdec.c
@@ -979,7 +979,13 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
dsputil_init(&vc->dsp, avccontext);
ff_fmt_convert_init(&vc->fmt_conv, avccontext);
- vc->scale_bias = 32768.0f;
+ if (avccontext->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+ avccontext->sample_fmt = AV_SAMPLE_FMT_FLT;
+ vc->scale_bias = 1.0f;
+ } else {
+ avccontext->sample_fmt = AV_SAMPLE_FMT_S16;
+ vc->scale_bias = 32768.0f;
+ }
if (!headers_len) {
av_log(avccontext, AV_LOG_ERROR, "Extradata missing.\n");
@@ -1024,7 +1030,6 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
avccontext->channels = vc->audio_channels;
avccontext->sample_rate = vc->audio_samplerate;
avccontext->frame_size = FFMIN(vc->blocksize[0], vc->blocksize[1]) >> 2;
- avccontext->sample_fmt = AV_SAMPLE_FMT_S16;
return 0 ;
}
@@ -1634,9 +1639,14 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
len * ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i];
}
- vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len,
- vc->audio_channels);
- *data_size = len * 2 * vc->audio_channels;
+ if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT)
+ vc->fmt_conv.float_interleave(data, channel_ptrs, len, vc->audio_channels);
+ else
+ vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len,
+ vc->audio_channels);
+
+ *data_size = len * vc->audio_channels *
+ (av_get_bits_per_sample_fmt(avccontext->sample_fmt) / 8);
return buf_size ;
}
@@ -1663,5 +1673,8 @@ AVCodec ff_vorbis_decoder = {
vorbis_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
.channel_layouts = ff_vorbis_channel_layouts,
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+ },
};
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