From 7bfd1766d1c18f07b0a2dd042418a874d49ea60d Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Sun, 26 Aug 2012 20:41:45 -0400 Subject: binkaudio: use float sample format Use planar for DCT codec, interleaved for RDFT codec. --- libavcodec/binkaudio.c | 56 +++++++++++++++++++------------------------------- 1 file changed, 21 insertions(+), 35 deletions(-) (limited to 'libavcodec/binkaudio.c') diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c index 915e7aa171..957af79c03 100644 --- a/libavcodec/binkaudio.c +++ b/libavcodec/binkaudio.c @@ -47,7 +47,6 @@ static float quant_table[96]; typedef struct { AVFrame frame; GetBitContext gb; - FmtConvertContext fmt_conv; int version_b; ///< Bink version 'b' int first; int channels; @@ -58,10 +57,7 @@ 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; @@ -78,8 +74,6 @@ static av_cold int decode_init(AVCodecContext *avctx) int i; int frame_len_bits; - ff_fmt_convert_init(&s->fmt_conv, avctx); - /* determine frame length */ if (avctx->sample_rate < 22050) { frame_len_bits = 9; @@ -98,12 +92,14 @@ static av_cold int decode_init(AVCodecContext *avctx) 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; @@ -111,9 +107,9 @@ static av_cold int decode_init(AVCodecContext *avctx) s->block_size = (s->frame_len - s->overlap_len) * s->channels; sample_rate_half = (sample_rate + 1) / 2; if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) - s->root = 2.0 / sqrt(s->frame_len); + s->root = 2.0 / (sqrt(s->frame_len) * 32768.0); else - s->root = s->frame_len / sqrt(s->frame_len); + 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; @@ -135,12 +131,6 @@ 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 == AV_CODEC_ID_BINKAUDIO_RDFT) ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R); @@ -179,7 +169,7 @@ static const uint8_t rle_length_tab[16] = { * @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]; @@ -190,7 +180,8 @@ 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; @@ -265,24 +256,19 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct) 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; @@ -311,7 +297,6 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { BinkAudioContext *s = avctx->priv_data; - int16_t *samples; GetBitContext *gb = &s->gb; int ret, consumed = 0; @@ -339,19 +324,20 @@ static int decode_frame(AVCodecContext *avctx, void *data, } /* get output buffer */ - s->frame.nb_samples = s->block_size / avctx->channels; + s->frame.nb_samples = s->frame_len; if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); return ret; } - samples = (int16_t *)s->frame.data[0]; - if (decode_block(s, samples, avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) { + if (decode_block(s, (float **)s->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); + s->frame.nb_samples = s->block_size / avctx->channels; *got_frame_ptr = 1; *(AVFrame *)data = s->frame; -- cgit v1.2.3