X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fbinkaudio.c;h=71ad344bbfa680f0daf58ae41ee4e016832722b1;hb=f290e48d86e10f34b5ddc519127636bcebec7c43;hp=a90467abfb580dca61bd400f96810a12cdf4e0e1;hpb=425a84350507e18c57ba0bee32366eb5963a9da5;p=ffmpeg diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c index a90467abfb5..71ad344bbfa 100644 --- a/libavcodec/binkaudio.c +++ b/libavcodec/binkaudio.c @@ -28,26 +28,23 @@ * http://wiki.multimedia.cx/index.php?title=Bink_Audio */ +#include "libavutil/channel_layout.h" #include "avcodec.h" -#define ALT_BITSTREAM_READER_LE +#define BITSTREAM_READER_LE #include "get_bits.h" -#include "dsputil.h" #include "dct.h" #include "rdft.h" -#include "fmtconvert.h" -#include "libavutil/intfloat_readwrite.h" +#include "internal.h" +#include "wma_freqs.h" +#include "libavutil/intfloat.h" -extern const uint16_t ff_wma_critical_freqs[25]; - -static float quant_table[95]; +static float quant_table[96]; #define MAX_CHANNELS 2 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11) -typedef struct { +typedef struct BinkAudioContext { GetBitContext gb; - DSPContext dsp; - FmtConvertContext fmt_conv; int version_b; ///< Bink version 'b' int first; int channels; @@ -58,10 +55,7 @@ typedef struct { unsigned int *bands; float root; DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE]; - DECLARE_ALIGNED(16, int16_t, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block - DECLARE_ALIGNED(16, int16_t, current)[BINK_BLOCK_MAX_SIZE / 16]; - float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave - float *prev_ptr[MAX_CHANNELS]; ///< pointers to the overlap points in the coeffs array + float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block uint8_t *packet_buffer; union { RDFTContext rdft; @@ -78,9 +72,6 @@ static av_cold int decode_init(AVCodecContext *avctx) int i; int frame_len_bits; - dsputil_init(&s->dsp, avctx); - ff_fmt_convert_init(&s->fmt_conv, avctx); - /* determine frame length */ if (avctx->sample_rate < 22050) { frame_len_bits = 9; @@ -94,25 +85,32 @@ static av_cold int decode_init(AVCodecContext *avctx) av_log(avctx, AV_LOG_ERROR, "too many channels: %d\n", avctx->channels); return -1; } + avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO : + AV_CH_LAYOUT_STEREO; s->version_b = avctx->extradata && avctx->extradata[3] == 'b'; - if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) { + if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) { // audio is already interleaved for the RDFT format variant + avctx->sample_fmt = AV_SAMPLE_FMT_FLT; sample_rate *= avctx->channels; s->channels = 1; if (!s->version_b) frame_len_bits += av_log2(avctx->channels); } else { s->channels = avctx->channels; + avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; } s->frame_len = 1 << frame_len_bits; s->overlap_len = s->frame_len / 16; s->block_size = (s->frame_len - s->overlap_len) * s->channels; sample_rate_half = (sample_rate + 1) / 2; - s->root = 2.0 / sqrt(s->frame_len); - for (i = 0; i < 95; i++) { + if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) + s->root = 2.0 / (sqrt(s->frame_len) * 32768.0); + else + s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0); + for (i = 0; i < 96; i++) { /* constant is result of 0.066399999/log10(M_E) */ quant_table[i] = expf(i * 0.15289164787221953823f) * s->root; } @@ -133,14 +131,8 @@ static av_cold int decode_init(AVCodecContext *avctx) s->bands[s->num_bands] = s->frame_len; s->first = 1; - avctx->sample_fmt = AV_SAMPLE_FMT_S16; - for (i = 0; i < s->channels; i++) { - s->coeffs_ptr[i] = s->coeffs + i * s->frame_len; - s->prev_ptr[i] = s->coeffs_ptr[i] + s->frame_len - s->overlap_len; - } - - if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) + if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R); else if (CONFIG_BINKAUDIO_DCT_DECODER) ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III); @@ -163,18 +155,12 @@ static const uint8_t rle_length_tab[16] = { 2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64 }; -#define GET_BITS_SAFE(out, nbits) do { \ - if (get_bits_left(gb) < nbits) \ - return AVERROR_INVALIDDATA; \ - out = get_bits(gb, nbits); \ -} while (0) - /** * Decode Bink Audio block * @param[out] out Output buffer (must contain s->block_size elements) * @return 0 on success, negative error code on failure */ -static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct) +static int decode_block(BinkAudioContext *s, float **out, int use_dct) { int ch, i, j, k; float q, quant[25]; @@ -185,12 +171,13 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct) skip_bits(gb, 2); for (ch = 0; ch < s->channels; ch++) { - FFTSample *coeffs = s->coeffs_ptr[ch]; + FFTSample *coeffs = out[ch]; + if (s->version_b) { if (get_bits_left(gb) < 64) return AVERROR_INVALIDDATA; - coeffs[0] = av_int2flt(get_bits(gb, 32)) * s->root; - coeffs[1] = av_int2flt(get_bits(gb, 32)) * s->root; + coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root; + coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root; } else { if (get_bits_left(gb) < 58) return AVERROR_INVALIDDATA; @@ -214,10 +201,9 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct) if (s->version_b) { j = i + 16; } else { - int v; - GET_BITS_SAFE(v, 1); + int v = get_bits1(gb); if (v) { - GET_BITS_SAFE(v, 4); + v = get_bits(gb, 4); j = i + rle_length_tab[v] * 8; } else { j = i + 8; @@ -226,7 +212,7 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct) j = FFMIN(j, s->frame_len); - GET_BITS_SAFE(width, 4); + width = get_bits(gb, 4); if (width == 0) { memset(coeffs + i, 0, (j - i) * sizeof(*coeffs)); i = j; @@ -236,10 +222,10 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct) while (i < j) { if (s->bands[k] == i) q = quant[k++]; - GET_BITS_SAFE(coeff, width); + coeff = get_bits(gb, width); if (coeff) { int v; - GET_BITS_SAFE(v, 1); + v = get_bits1(gb); if (v) coeffs[i] = -q * coeff; else @@ -255,30 +241,24 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct) if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) { coeffs[0] /= 0.5; s->trans.dct.dct_calc(&s->trans.dct, coeffs); - s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len); } else if (CONFIG_BINKAUDIO_RDFT_DECODER) s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs); } - s->fmt_conv.float_to_int16_interleave(s->current, - (const float **)s->prev_ptr, - s->overlap_len, s->channels); - s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, - s->frame_len - s->overlap_len, - s->channels); - - if (!s->first) { + for (ch = 0; ch < s->channels; ch++) { + int j; int count = s->overlap_len * s->channels; - int shift = av_log2(count); - for (i = 0; i < count; i++) { - out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift; + if (!s->first) { + j = ch; + for (i = 0; i < s->overlap_len; i++, j += s->channels) + out[ch][i] = (s->previous[ch][i] * (count - j) + + out[ch][i] * j) / count; } + memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len], + s->overlap_len * sizeof(*s->previous[ch])); } - memcpy(s->previous, s->current, - s->overlap_len * s->channels * sizeof(*s->previous)); - s->first = 0; return 0; @@ -289,10 +269,11 @@ static av_cold int decode_end(AVCodecContext *avctx) BinkAudioContext * s = avctx->priv_data; av_freep(&s->bands); av_freep(&s->packet_buffer); - if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) + if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) ff_rdft_end(&s->trans.rdft); else if (CONFIG_BINKAUDIO_DCT_DECODER) ff_dct_end(&s->trans.dct); + return 0; } @@ -302,27 +283,26 @@ static void get_bits_align32(GetBitContext *s) if (n) skip_bits(s, n); } -static int decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { BinkAudioContext *s = avctx->priv_data; - int16_t *samples = data; + AVFrame *frame = data; GetBitContext *gb = &s->gb; - int out_size, consumed = 0; + int ret, consumed = 0; if (!get_bits_left(gb)) { uint8_t *buf; /* handle end-of-stream */ if (!avpkt->size) { - *data_size = 0; + *got_frame_ptr = 0; return 0; } if (avpkt->size < 4) { av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); return AVERROR_INVALIDDATA; } - buf = av_realloc(s->packet_buffer, avpkt->size + FF_INPUT_BUFFER_PADDING_SIZE); + buf = av_realloc(s->packet_buffer, avpkt->size + AV_INPUT_BUFFER_PADDING_SIZE); if (!buf) return AVERROR(ENOMEM); s->packet_buffer = buf; @@ -334,42 +314,46 @@ static int decode_frame(AVCodecContext *avctx, skip_bits_long(gb, 32); } - out_size = s->block_size * av_get_bytes_per_sample(avctx->sample_fmt); - if (*data_size < out_size) { - av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n"); - return AVERROR(EINVAL); + /* get output buffer */ + frame->nb_samples = s->frame_len; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } - if (decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT)) { + if (decode_block(s, (float **)frame->extended_data, + avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) { av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n"); return AVERROR_INVALIDDATA; } get_bits_align32(gb); - *data_size = out_size; + frame->nb_samples = s->block_size / avctx->channels; + *got_frame_ptr = 1; + return consumed; } AVCodec ff_binkaudio_rdft_decoder = { .name = "binkaudio_rdft", + .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"), .type = AVMEDIA_TYPE_AUDIO, - .id = CODEC_ID_BINKAUDIO_RDFT, + .id = AV_CODEC_ID_BINKAUDIO_RDFT, .priv_data_size = sizeof(BinkAudioContext), .init = decode_init, .close = decode_end, .decode = decode_frame, - .capabilities = CODEC_CAP_DELAY, - .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)") + .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1, }; AVCodec ff_binkaudio_dct_decoder = { .name = "binkaudio_dct", + .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"), .type = AVMEDIA_TYPE_AUDIO, - .id = CODEC_ID_BINKAUDIO_DCT, + .id = AV_CODEC_ID_BINKAUDIO_DCT, .priv_data_size = sizeof(BinkAudioContext), .init = decode_init, .close = decode_end, .decode = decode_frame, - .capabilities = CODEC_CAP_DELAY, - .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)") + .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1, };