Commit b8f3ab8e authored by Mans Rullgard's avatar Mans Rullgard

ac3dec: output planar float only

Signed-off-by: 's avatarMans Rullgard <mans@mansr.com>
parent 288bb3da
......@@ -172,14 +172,7 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
ff_fmt_convert_init(&s->fmt_conv, avctx);
av_lfg_init(&s->dith_state, 0);
/* set scale value for float to int16 conversion */
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;
}
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
/* allow downmixing to stereo or mono */
if (avctx->channels > 0 && avctx->request_channels > 0 &&
......@@ -1206,7 +1199,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
/* apply scaling to coefficients (headroom, dynrng) */
for (ch = 1; ch <= s->channels; ch++) {
float gain = s->mul_bias / 4194304.0f;
float gain = 1.0 / 4194304.0f;
if (s->channel_mode == AC3_CHMODE_DUALMONO) {
gain *= s->dynamic_range[2 - ch];
} else {
......@@ -1268,8 +1261,6 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
float *out_samples_flt;
int16_t *out_samples_s16;
int blk, ch, err, ret;
const uint8_t *channel_map;
const float *output[AC3_MAX_CHANNELS];
......@@ -1375,8 +1366,6 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
out_samples_flt = (float *)s->frame.data[0];
out_samples_s16 = (int16_t *)s->frame.data[0];
/* decode the audio blocks */
channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
......@@ -1387,15 +1376,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
err = 1;
}
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;
}
for (ch = 0; ch < s->out_channels; ch++)
memcpy(s->frame.data[ch] + blk * 1024, output[ch], 1024);
}
*got_frame_ptr = 1;
......@@ -1440,8 +1422,7 @@ AVCodec ff_ac3_decoder = {
.decode = ac3_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.priv_class = &ac3_decoder_class,
};
......@@ -1464,8 +1445,7 @@ AVCodec ff_eac3_decoder = {
.decode = ac3_decode_frame,
.capabilities = CODEC_CAP_DR1,
.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,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.priv_class = &eac3_decoder_class,
};
......
......@@ -195,7 +195,6 @@ typedef struct AC3DecodeContext {
DSPContext dsp; ///< for optimization
AC3DSPContext ac3dsp;
FmtConvertContext fmt_conv; ///< optimized conversion functions
float mul_bias; ///< scaling for float_to_int16 conversion
///@}
///@name Aligned arrays
......
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