From 9aa8193a234ccb6a79cba5cc550531f62ffb0a17 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Fri, 22 Apr 2011 21:30:19 -0400 Subject: Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders. Based on patches by clsid2 in ffdshow-tryout. --- 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 5f9dd834a0..f2d50f4aba 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 7a217abfda..81b0b4c001 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 dca83305ad..153070d3f2 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 015ebaebec..2966c33b25 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 f1cd64e9ed..dbadebacff 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 7443e98a64..f6ec74f4a1 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 + }, }; -- cgit v1.2.3