diff options
Diffstat (limited to 'libavcodec/ac3dec.c')
| -rw-r--r-- | libavcodec/ac3dec.c | 32 |
1 files changed, 6 insertions, 26 deletions
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c index 74dd8c1e98..c5507cc3e9 100644 --- a/libavcodec/ac3dec.c +++ b/libavcodec/ac3dec.c @@ -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, }; |