mirror of https://git.ffmpeg.org/ffmpeg.git
ac3dec: output planar float only
Signed-off-by: Mans Rullgard <mans@mansr.com>
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288bb3da16
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@ -172,14 +172,7 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
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ff_fmt_convert_init(&s->fmt_conv, avctx);
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av_lfg_init(&s->dith_state, 0);
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/* set scale value for float to int16 conversion */
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if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
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s->mul_bias = 1.0f;
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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} else {
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s->mul_bias = 32767.0f;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
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/* allow downmixing to stereo or mono */
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if (avctx->channels > 0 && avctx->request_channels > 0 &&
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@ -1206,7 +1199,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
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/* apply scaling to coefficients (headroom, dynrng) */
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for (ch = 1; ch <= s->channels; ch++) {
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float gain = s->mul_bias / 4194304.0f;
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float gain = 1.0 / 4194304.0f;
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if (s->channel_mode == AC3_CHMODE_DUALMONO) {
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gain *= s->dynamic_range[2 - ch];
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} else {
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@ -1268,8 +1261,6 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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AC3DecodeContext *s = avctx->priv_data;
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float *out_samples_flt;
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int16_t *out_samples_s16;
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int blk, ch, err, ret;
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const uint8_t *channel_map;
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const float *output[AC3_MAX_CHANNELS];
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@ -1375,8 +1366,6 @@ static int ac3_decode_frame(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_flt = (float *)s->frame.data[0];
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out_samples_s16 = (int16_t *)s->frame.data[0];
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/* decode the audio blocks */
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channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
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@ -1387,15 +1376,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
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av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
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err = 1;
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}
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if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
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s->fmt_conv.float_interleave(out_samples_flt, output, 256,
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s->out_channels);
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out_samples_flt += 256 * s->out_channels;
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} else {
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s->fmt_conv.float_to_int16_interleave(out_samples_s16, output, 256,
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s->out_channels);
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out_samples_s16 += 256 * s->out_channels;
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}
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for (ch = 0; ch < s->out_channels; ch++)
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memcpy(s->frame.data[ch] + blk * 1024, output[ch], 1024);
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}
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*got_frame_ptr = 1;
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@ -1440,8 +1422,7 @@ AVCodec ff_ac3_decoder = {
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.decode = ac3_decode_frame,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_S16,
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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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.priv_class = &ac3_decoder_class,
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};
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@ -1464,8 +1445,7 @@ AVCodec ff_eac3_decoder = {
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.decode = ac3_decode_frame,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_S16,
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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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.priv_class = &eac3_decoder_class,
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};
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@ -195,7 +195,6 @@ typedef struct AC3DecodeContext {
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DSPContext dsp; ///< for optimization
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AC3DSPContext ac3dsp;
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FmtConvertContext fmt_conv; ///< optimized conversion functions
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float mul_bias; ///< scaling for float_to_int16 conversion
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///@}
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///@name Aligned arrays
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