X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fatrac1.c;h=d0b58704f6d4204d06fd504f61a65b141a4c2b04;hb=ebd4c3add1ecad2c65ac80f0787ca5c1e78b600e;hp=1ba580cbacaea05735f7b56b501717ffa133c9ca;hpb=bff5b2c1ca1290ea30587ff2f76171f9e3854872;p=ffmpeg diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c index 1ba580cbaca..d0b58704f6d 100644 --- a/libavcodec/atrac1.c +++ b/libavcodec/atrac1.c @@ -36,6 +36,7 @@ #include "get_bits.h" #include "dsputil.h" #include "fft.h" +#include "fmtconvert.h" #include "sinewin.h" #include "atrac.h" @@ -71,6 +72,7 @@ typedef struct { * The atrac1 context, holds all needed parameters for decoding */ typedef struct { + AVFrame frame; AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer @@ -78,10 +80,11 @@ typedef struct { DECLARE_ALIGNED(32, float, mid)[256]; DECLARE_ALIGNED(32, float, high)[512]; float* bands[3]; - DECLARE_ALIGNED(32, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]; + float *out_samples[AT1_MAX_CHANNELS]; FFTContext mdct_ctx[3]; int channels; DSPContext dsp; + FmtConvertContext fmt_conv; } AT1Ctx; /** size of the transform in samples in the long mode for each QMF band */ @@ -129,7 +132,7 @@ static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) nbits = mdct_long_nbits[band_num] - log2_block_count; if (nbits != 5 && nbits != 7 && nbits != 8) - return -1; + return AVERROR_INVALIDDATA; } else { block_size = 32; nbits = 5; @@ -173,14 +176,14 @@ static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS]) /* low and mid band */ log2_block_count_tmp = get_bits(gb, 2); if (log2_block_count_tmp & 1) - return -1; + return AVERROR_INVALIDDATA; log2_block_cnt[i] = 2 - log2_block_count_tmp; } /* high band */ log2_block_count_tmp = get_bits(gb, 2); if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3) - return -1; + return AVERROR_INVALIDDATA; log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; skip_bits(gb, 2); @@ -229,7 +232,7 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, /* check for bitstream overflow */ if (bits_used > AT1_SU_MAX_BITS) - return -1; + return AVERROR_INVALIDDATA; /* get the position of the 1st spec according to the block size mode */ pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num]; @@ -259,39 +262,40 @@ static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) float iqmf_temp[512 + 46]; /* combine low and middle bands */ - atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp); + ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp); /* delay the signal of the high band by 23 samples */ memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23); memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256); /* combine (low + middle) and high bands */ - atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); + ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); } static int atrac1_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; AT1Ctx *q = avctx->priv_data; - int ch, ret, i, out_size; + int ch, ret; GetBitContext gb; - float* samples = data; + float *samples; if (buf_size < 212 * q->channels) { - av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n"); - return -1; + av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n"); + return AVERROR_INVALIDDATA; } - out_size = q->channels * AT1_SU_SAMPLES * - 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 */ + q->frame.nb_samples = AT1_SU_SAMPLES; + if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (float *)q->frame.data[0]; for (ch = 0; ch < q->channels; ch++) { AT1SUCtx* su = &q->SUs[ch]; @@ -313,22 +317,37 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data, at1_subband_synthesis(q, su, q->channels == 1 ? samples : q->out_samples[ch]); } - /* interleave; FIXME, should create/use a DSP function */ + /* interleave */ if (q->channels == 2) { - for (i = 0; i < AT1_SU_SAMPLES; i++) { - samples[i * 2] = q->out_samples[0][i]; - samples[i * 2 + 1] = q->out_samples[1][i]; - } + q->fmt_conv.float_interleave(samples, (const float **)q->out_samples, + AT1_SU_SAMPLES, 2); } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = q->frame; + return avctx->block_align; } +static av_cold int atrac1_decode_end(AVCodecContext * avctx) +{ + AT1Ctx *q = avctx->priv_data; + + av_freep(&q->out_samples[0]); + + ff_mdct_end(&q->mdct_ctx[0]); + ff_mdct_end(&q->mdct_ctx[1]); + ff_mdct_end(&q->mdct_ctx[2]); + + return 0; +} + + static av_cold int atrac1_decode_init(AVCodecContext *avctx) { AT1Ctx *q = avctx->priv_data; + int ret; avctx->sample_fmt = AV_SAMPLE_FMT_FLT; @@ -339,16 +358,30 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx) } q->channels = avctx->channels; + if (avctx->channels == 2) { + q->out_samples[0] = av_malloc(2 * AT1_SU_SAMPLES * sizeof(*q->out_samples[0])); + q->out_samples[1] = q->out_samples[0] + AT1_SU_SAMPLES; + if (!q->out_samples[0]) { + av_freep(&q->out_samples[0]); + return AVERROR(ENOMEM); + } + } + /* Init the mdct transforms */ - ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15)); - ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15)); - ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)); + if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) || + (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) || + (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) { + av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n"); + atrac1_decode_end(avctx); + return ret; + } ff_init_ff_sine_windows(5); - atrac_generate_tables(); + ff_atrac_generate_tables(); - dsputil_init(&q->dsp, avctx); + ff_dsputil_init(&q->dsp, avctx); + ff_fmt_convert_init(&q->fmt_conv, avctx); q->bands[0] = q->low; q->bands[1] = q->mid; @@ -360,27 +393,21 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx) q->SUs[1].spectrum[0] = q->SUs[1].spec1; q->SUs[1].spectrum[1] = q->SUs[1].spec2; - return 0; -} + avcodec_get_frame_defaults(&q->frame); + avctx->coded_frame = &q->frame; - -static av_cold int atrac1_decode_end(AVCodecContext * avctx) { - AT1Ctx *q = avctx->priv_data; - - ff_mdct_end(&q->mdct_ctx[0]); - ff_mdct_end(&q->mdct_ctx[1]); - ff_mdct_end(&q->mdct_ctx[2]); return 0; } AVCodec ff_atrac1_decoder = { - .name = "atrac1", - .type = AVMEDIA_TYPE_AUDIO, - .id = CODEC_ID_ATRAC1, + .name = "atrac1", + .type = AVMEDIA_TYPE_AUDIO, + .id = CODEC_ID_ATRAC1, .priv_data_size = sizeof(AT1Ctx), - .init = atrac1_decode_init, - .close = atrac1_decode_end, - .decode = atrac1_decode_frame, - .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"), + .init = atrac1_decode_init, + .close = atrac1_decode_end, + .decode = atrac1_decode_frame, + .capabilities = CODEC_CAP_DR1, + .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"), };