From 799e2324901c2a06e9a60ee281cd283475f1c4fa Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Sat, 27 Oct 2012 00:25:04 -0400 Subject: [PATCH 1/5] flacdsp: move lpc encoding from FLAC encoder to FLACDSPContext Also, templatize the functions for 16-bit and 32-bit sample range. This will be used for 24-bit FLAC encoding. --- libavcodec/flacdsp.c | 9 +- libavcodec/flacdsp.h | 2 + libavcodec/flacdsp_lpc_template.c | 141 ++++++++++++++++++++++++++++++ libavcodec/flacenc.c | 116 ++---------------------- 4 files changed, 158 insertions(+), 110 deletions(-) create mode 100644 libavcodec/flacdsp_lpc_template.c diff --git a/libavcodec/flacdsp.c b/libavcodec/flacdsp.c index 3f4f71076c..b916869321 100644 --- a/libavcodec/flacdsp.c +++ b/libavcodec/flacdsp.c @@ -26,6 +26,7 @@ #define SAMPLE_SIZE 16 #define PLANAR 0 #include "flacdsp_template.c" +#include "flacdsp_lpc_template.c" #undef PLANAR #define PLANAR 1 @@ -36,6 +37,7 @@ #define SAMPLE_SIZE 32 #define PLANAR 0 #include "flacdsp_template.c" +#include "flacdsp_lpc_template.c" #undef PLANAR #define PLANAR 1 @@ -86,10 +88,13 @@ static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32], av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int bps) { - if (bps > 16) + if (bps > 16) { c->lpc = flac_lpc_32_c; - else + c->lpc_encode = flac_lpc_encode_c_32; + } else { c->lpc = flac_lpc_16_c; + c->lpc_encode = flac_lpc_encode_c_16; + } switch (fmt) { case AV_SAMPLE_FMT_S32: diff --git a/libavcodec/flacdsp.h b/libavcodec/flacdsp.h index 429daab346..33184b589d 100644 --- a/libavcodec/flacdsp.h +++ b/libavcodec/flacdsp.h @@ -27,6 +27,8 @@ typedef struct FLACDSPContext { int len, int shift); void (*lpc)(int32_t *samples, const int coeffs[32], int order, int qlevel, int len); + void (*lpc_encode)(int32_t *res, const int32_t *smp, int len, int order, + const int32_t *coefs, int shift); } FLACDSPContext; void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int bps); diff --git a/libavcodec/flacdsp_lpc_template.c b/libavcodec/flacdsp_lpc_template.c new file mode 100644 index 0000000000..269e64b3a8 --- /dev/null +++ b/libavcodec/flacdsp_lpc_template.c @@ -0,0 +1,141 @@ +/* + * This file is part of Libav. + * + * Libav is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * Libav is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include +#include "libavutil/avutil.h" +#include "mathops.h" + +#undef FUNC +#undef sum_type +#undef MUL +#undef CLIP +#undef FSUF + +#define FUNC(n) AV_JOIN(n ## _, SAMPLE_SIZE) + +#if SAMPLE_SIZE == 32 +# define sum_type int64_t +# define MUL(a, b) MUL64(a, b) +# define CLIP(x) av_clipl_int32(x) +#else +# define sum_type int32_t +# define MUL(a, b) ((a) * (b)) +# define CLIP(x) (x) +#endif + +#define LPC1(x) { \ + int c = coefs[(x)-1]; \ + p0 += MUL(c, s); \ + s = smp[i-(x)+1]; \ + p1 += MUL(c, s); \ +} + +static av_always_inline void FUNC(lpc_encode_unrolled)(int32_t *res, + const int32_t *smp, int len, int order, + const int32_t *coefs, int shift, int big) +{ + int i; + for (i = order; i < len; i += 2) { + int s = smp[i-order]; + sum_type p0 = 0, p1 = 0; + if (big) { + switch (order) { + case 32: LPC1(32) + case 31: LPC1(31) + case 30: LPC1(30) + case 29: LPC1(29) + case 28: LPC1(28) + case 27: LPC1(27) + case 26: LPC1(26) + case 25: LPC1(25) + case 24: LPC1(24) + case 23: LPC1(23) + case 22: LPC1(22) + case 21: LPC1(21) + case 20: LPC1(20) + case 19: LPC1(19) + case 18: LPC1(18) + case 17: LPC1(17) + case 16: LPC1(16) + case 15: LPC1(15) + case 14: LPC1(14) + case 13: LPC1(13) + case 12: LPC1(12) + case 11: LPC1(11) + case 10: LPC1(10) + case 9: LPC1( 9) + LPC1( 8) + LPC1( 7) + LPC1( 6) + LPC1( 5) + LPC1( 4) + LPC1( 3) + LPC1( 2) + LPC1( 1) + } + } else { + switch (order) { + case 8: LPC1( 8) + case 7: LPC1( 7) + case 6: LPC1( 6) + case 5: LPC1( 5) + case 4: LPC1( 4) + case 3: LPC1( 3) + case 2: LPC1( 2) + case 1: LPC1( 1) + } + } + res[i ] = smp[i ] - CLIP(p0 >> shift); + res[i+1] = smp[i+1] - CLIP(p1 >> shift); + } +} + +static void FUNC(flac_lpc_encode_c)(int32_t *res, const int32_t *smp, int len, + int order, const int32_t *coefs, int shift) +{ + int i; + for (i = 0; i < order; i++) + res[i] = smp[i]; +#if CONFIG_SMALL + for (i = order; i < len; i += 2) { + int j; + int s = smp[i]; + sum_type p0 = 0, p1 = 0; + for (j = 0; j < order; j++) { + int c = coefs[j]; + p1 += MUL(c, s); + s = smp[i-j-1]; + p0 += MUL(c, s); + } + res[i ] = smp[i ] - CLIP(p0 >> shift); + res[i+1] = smp[i+1] - CLIP(p1 >> shift); + } +#else + switch (order) { + case 1: FUNC(lpc_encode_unrolled)(res, smp, len, 1, coefs, shift, 0); break; + case 2: FUNC(lpc_encode_unrolled)(res, smp, len, 2, coefs, shift, 0); break; + case 3: FUNC(lpc_encode_unrolled)(res, smp, len, 3, coefs, shift, 0); break; + case 4: FUNC(lpc_encode_unrolled)(res, smp, len, 4, coefs, shift, 0); break; + case 5: FUNC(lpc_encode_unrolled)(res, smp, len, 5, coefs, shift, 0); break; + case 6: FUNC(lpc_encode_unrolled)(res, smp, len, 6, coefs, shift, 0); break; + case 7: FUNC(lpc_encode_unrolled)(res, smp, len, 7, coefs, shift, 0); break; + case 8: FUNC(lpc_encode_unrolled)(res, smp, len, 8, coefs, shift, 0); break; + default: FUNC(lpc_encode_unrolled)(res, smp, len, order, coefs, shift, 1); break; + } +#endif +} diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c index 115664a779..71024f272e 100644 --- a/libavcodec/flacenc.c +++ b/libavcodec/flacenc.c @@ -31,6 +31,7 @@ #include "lpc.h" #include "flac.h" #include "flacdata.h" +#include "flacdsp.h" #define FLAC_SUBFRAME_CONSTANT 0 #define FLAC_SUBFRAME_VERBATIM 1 @@ -106,6 +107,7 @@ typedef struct FlacEncodeContext { uint8_t *md5_buffer; unsigned int md5_buffer_size; DSPContext dsp; + FLACDSPContext flac_dsp; } FlacEncodeContext; @@ -385,6 +387,7 @@ static av_cold int flac_encode_init(AVCodecContext *avctx) s->options.max_prediction_order, FF_LPC_TYPE_LEVINSON); ff_dsputil_init(&s->dsp, avctx); + ff_flacdsp_init(&s->flac_dsp, avctx->sample_fmt, 16); dprint_compression_options(s); @@ -684,110 +687,6 @@ static void encode_residual_fixed(int32_t *res, const int32_t *smp, int n, } -#define LPC1(x) {\ - int c = coefs[(x)-1];\ - p0 += c * s;\ - s = smp[i-(x)+1];\ - p1 += c * s;\ -} - -static av_always_inline void encode_residual_lpc_unrolled(int32_t *res, - const int32_t *smp, int n, int order, - const int32_t *coefs, int shift, int big) -{ - int i; - for (i = order; i < n; i += 2) { - int s = smp[i-order]; - int p0 = 0, p1 = 0; - if (big) { - switch (order) { - case 32: LPC1(32) - case 31: LPC1(31) - case 30: LPC1(30) - case 29: LPC1(29) - case 28: LPC1(28) - case 27: LPC1(27) - case 26: LPC1(26) - case 25: LPC1(25) - case 24: LPC1(24) - case 23: LPC1(23) - case 22: LPC1(22) - case 21: LPC1(21) - case 20: LPC1(20) - case 19: LPC1(19) - case 18: LPC1(18) - case 17: LPC1(17) - case 16: LPC1(16) - case 15: LPC1(15) - case 14: LPC1(14) - case 13: LPC1(13) - case 12: LPC1(12) - case 11: LPC1(11) - case 10: LPC1(10) - case 9: LPC1( 9) - LPC1( 8) - LPC1( 7) - LPC1( 6) - LPC1( 5) - LPC1( 4) - LPC1( 3) - LPC1( 2) - LPC1( 1) - } - } else { - switch (order) { - case 8: LPC1( 8) - case 7: LPC1( 7) - case 6: LPC1( 6) - case 5: LPC1( 5) - case 4: LPC1( 4) - case 3: LPC1( 3) - case 2: LPC1( 2) - case 1: LPC1( 1) - } - } - res[i ] = smp[i ] - (p0 >> shift); - res[i+1] = smp[i+1] - (p1 >> shift); - } -} - - -static void encode_residual_lpc(int32_t *res, const int32_t *smp, int n, - int order, const int32_t *coefs, int shift) -{ - int i; - for (i = 0; i < order; i++) - res[i] = smp[i]; -#if CONFIG_SMALL - for (i = order; i < n; i += 2) { - int j; - int s = smp[i]; - int p0 = 0, p1 = 0; - for (j = 0; j < order; j++) { - int c = coefs[j]; - p1 += c * s; - s = smp[i-j-1]; - p0 += c * s; - } - res[i ] = smp[i ] - (p0 >> shift); - res[i+1] = smp[i+1] - (p1 >> shift); - } -#else - switch (order) { - case 1: encode_residual_lpc_unrolled(res, smp, n, 1, coefs, shift, 0); break; - case 2: encode_residual_lpc_unrolled(res, smp, n, 2, coefs, shift, 0); break; - case 3: encode_residual_lpc_unrolled(res, smp, n, 3, coefs, shift, 0); break; - case 4: encode_residual_lpc_unrolled(res, smp, n, 4, coefs, shift, 0); break; - case 5: encode_residual_lpc_unrolled(res, smp, n, 5, coefs, shift, 0); break; - case 6: encode_residual_lpc_unrolled(res, smp, n, 6, coefs, shift, 0); break; - case 7: encode_residual_lpc_unrolled(res, smp, n, 7, coefs, shift, 0); break; - case 8: encode_residual_lpc_unrolled(res, smp, n, 8, coefs, shift, 0); break; - default: encode_residual_lpc_unrolled(res, smp, n, order, coefs, shift, 1); break; - } -#endif -} - - static int encode_residual_ch(FlacEncodeContext *s, int ch) { int i, n; @@ -869,7 +768,8 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) order = min_order + (((max_order-min_order+1) * (i+1)) / levels)-1; if (order < 0) order = 0; - encode_residual_lpc(res, smp, n, order+1, coefs[order], shift[order]); + s->flac_dsp.lpc_encode(res, smp, n, order+1, coefs[order], + shift[order]); bits[i] = find_subframe_rice_params(s, sub, order+1); if (bits[i] < bits[opt_index]) { opt_index = i; @@ -883,7 +783,7 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) opt_order = 0; bits[0] = UINT32_MAX; for (i = min_order-1; i < max_order; i++) { - encode_residual_lpc(res, smp, n, i+1, coefs[i], shift[i]); + s->flac_dsp.lpc_encode(res, smp, n, i+1, coefs[i], shift[i]); bits[i] = find_subframe_rice_params(s, sub, i+1); if (bits[i] < bits[opt_order]) opt_order = i; @@ -901,7 +801,7 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) for (i = last-step; i <= last+step; i += step) { if (i < min_order-1 || i >= max_order || bits[i] < UINT32_MAX) continue; - encode_residual_lpc(res, smp, n, i+1, coefs[i], shift[i]); + s->flac_dsp.lpc_encode(res, smp, n, i+1, coefs[i], shift[i]); bits[i] = find_subframe_rice_params(s, sub, i+1); if (bits[i] < bits[opt_order]) opt_order = i; @@ -916,7 +816,7 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch) for (i = 0; i < sub->order; i++) sub->coefs[i] = coefs[sub->order-1][i]; - encode_residual_lpc(res, smp, n, sub->order, sub->coefs, sub->shift); + s->flac_dsp.lpc_encode(res, smp, n, sub->order, sub->coefs, sub->shift); find_subframe_rice_params(s, sub, sub->order); From 13e1ee6c84f095b052026b18611ce68c76666474 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Sat, 27 Oct 2012 00:26:02 -0400 Subject: [PATCH 2/5] flacenc: add 24-bit encoding --- libavcodec/flacenc.c | 86 +++++++++++++++++++++++++++++++------------- 1 file changed, 62 insertions(+), 24 deletions(-) diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c index 71024f272e..b00df9580b 100644 --- a/libavcodec/flacenc.c +++ b/libavcodec/flacenc.c @@ -92,6 +92,7 @@ typedef struct FlacEncodeContext { int channels; int samplerate; int sr_code[2]; + int bps_code; int max_blocksize; int min_framesize; int max_framesize; @@ -128,7 +129,7 @@ static void write_streaminfo(FlacEncodeContext *s, uint8_t *header) put_bits(&pb, 24, s->max_framesize); put_bits(&pb, 20, s->samplerate); put_bits(&pb, 3, s->channels-1); - put_bits(&pb, 5, 15); /* bits per sample - 1 */ + put_bits(&pb, 5, s->avctx->bits_per_raw_sample - 1); /* write 36-bit sample count in 2 put_bits() calls */ put_bits(&pb, 24, (s->sample_count & 0xFFFFFF000LL) >> 12); put_bits(&pb, 12, s->sample_count & 0x000000FFFLL); @@ -228,8 +229,18 @@ static av_cold int flac_encode_init(AVCodecContext *avctx) s->avctx = avctx; - if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) - return -1; + switch (avctx->sample_fmt) { + case AV_SAMPLE_FMT_S16: + avctx->bits_per_raw_sample = 16; + s->bps_code = 4; + break; + case AV_SAMPLE_FMT_S32: + if (avctx->bits_per_raw_sample != 24) + av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n"); + avctx->bits_per_raw_sample = 24; + s->bps_code = 6; + break; + } if (channels < 1 || channels > FLAC_MAX_CHANNELS) return -1; @@ -359,7 +370,8 @@ static av_cold int flac_encode_init(AVCodecContext *avctx) /* set maximum encoded frame size in verbatim mode */ s->max_framesize = ff_flac_get_max_frame_size(s->avctx->frame_size, - s->channels, 16); + s->channels, + s->avctx->bits_per_raw_sample); /* initialize MD5 context */ s->md5ctx = av_md5_alloc(); @@ -387,7 +399,8 @@ static av_cold int flac_encode_init(AVCodecContext *avctx) s->options.max_prediction_order, FF_LPC_TYPE_LEVINSON); ff_dsputil_init(&s->dsp, avctx); - ff_flacdsp_init(&s->flac_dsp, avctx->sample_fmt, 16); + ff_flacdsp_init(&s->flac_dsp, avctx->sample_fmt, + avctx->bits_per_raw_sample); dprint_compression_options(s); @@ -423,7 +436,7 @@ static void init_frame(FlacEncodeContext *s, int nb_samples) for (ch = 0; ch < s->channels; ch++) { frame->subframes[ch].wasted = 0; - frame->subframes[ch].obits = 16; + frame->subframes[ch].obits = s->avctx->bits_per_raw_sample; } frame->verbatim_only = 0; @@ -433,15 +446,25 @@ static void init_frame(FlacEncodeContext *s, int nb_samples) /** * Copy channel-interleaved input samples into separate subframes. */ -static void copy_samples(FlacEncodeContext *s, const int16_t *samples) +static void copy_samples(FlacEncodeContext *s, const void *samples) { int i, j, ch; FlacFrame *frame; + int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 - + s->avctx->bits_per_raw_sample; - frame = &s->frame; - for (i = 0, j = 0; i < frame->blocksize; i++) - for (ch = 0; ch < s->channels; ch++, j++) - frame->subframes[ch].samples[i] = samples[j]; +#define COPY_SAMPLES(bits) do { \ + const int ## bits ## _t *samples0 = samples; \ + frame = &s->frame; \ + for (i = 0, j = 0; i < frame->blocksize; i++) \ + for (ch = 0; ch < s->channels; ch++, j++) \ + frame->subframes[ch].samples[i] = samples0[j] >> shift; \ +} while (0) + + if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S16) + COPY_SAMPLES(16); + else + COPY_SAMPLES(32); } @@ -1017,7 +1040,7 @@ static void write_frame_header(FlacEncodeContext *s) else put_bits(&s->pb, 4, frame->ch_mode + FLAC_MAX_CHANNELS - 1); - put_bits(&s->pb, 3, 4); /* bits-per-sample code */ + put_bits(&s->pb, 3, s->bps_code); put_bits(&s->pb, 1, 0); write_utf8(&s->pb, s->frame_count); @@ -1119,23 +1142,38 @@ static int write_frame(FlacEncodeContext *s, AVPacket *avpkt) } -static int update_md5_sum(FlacEncodeContext *s, const int16_t *samples) +static int update_md5_sum(FlacEncodeContext *s, const void *samples) { const uint8_t *buf; - int buf_size = s->frame.blocksize * s->channels * 2; + int buf_size = s->frame.blocksize * s->channels * + ((s->avctx->bits_per_raw_sample + 7) / 8); - if (HAVE_BIGENDIAN) { + if (s->avctx->bits_per_raw_sample > 16 || HAVE_BIGENDIAN) { av_fast_malloc(&s->md5_buffer, &s->md5_buffer_size, buf_size); if (!s->md5_buffer) return AVERROR(ENOMEM); } - buf = (const uint8_t *)samples; + if (s->avctx->bits_per_raw_sample <= 16) { + buf = (const uint8_t *)samples; #if HAVE_BIGENDIAN - s->dsp.bswap16_buf((uint16_t *)s->md5_buffer, - (const uint16_t *)samples, buf_size / 2); - buf = s->md5_buffer; + s->dsp.bswap16_buf((uint16_t *)s->md5_buffer, + (const uint16_t *)samples, buf_size / 2); + buf = s->md5_buffer; #endif + } else { + int i; + const int32_t *samples0 = samples; + uint8_t *tmp = s->md5_buffer; + + for (i = 0; i < s->frame.blocksize * s->channels; i++) { + int32_t v = samples0[i] >> 8; + *tmp++ = (v ) & 0xFF; + *tmp++ = (v >> 8) & 0xFF; + *tmp++ = (v >> 16) & 0xFF; + } + buf = s->md5_buffer; + } av_md5_update(s->md5ctx, buf, buf_size); return 0; @@ -1146,7 +1184,6 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { FlacEncodeContext *s; - const int16_t *samples; int frame_bytes, out_bytes, ret; s = avctx->priv_data; @@ -1158,17 +1195,17 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, write_streaminfo(s, avctx->extradata); return 0; } - samples = (const int16_t *)frame->data[0]; /* change max_framesize for small final frame */ if (frame->nb_samples < s->frame.blocksize) { s->max_framesize = ff_flac_get_max_frame_size(frame->nb_samples, - s->channels, 16); + s->channels, + avctx->bits_per_raw_sample); } init_frame(s, frame->nb_samples); - copy_samples(s, samples); + copy_samples(s, frame->data[0]); channel_decorrelation(s); @@ -1196,7 +1233,7 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, s->frame_count++; s->sample_count += frame->nb_samples; - if ((ret = update_md5_sum(s, samples)) < 0) { + if ((ret = update_md5_sum(s, frame->data[0])) < 0) { av_log(avctx, AV_LOG_ERROR, "Error updating MD5 checksum\n"); return ret; } @@ -1273,6 +1310,7 @@ AVCodec ff_flac_encoder = { .close = flac_encode_close, .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_NONE }, .long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"), .priv_class = &flac_encoder_class, From 2e76f34387762405b79d2f7f26659826089c7275 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Sat, 27 Oct 2012 00:46:43 -0400 Subject: [PATCH 3/5] flacenc: use RICE2 entropy coding mode for 24-bit --- libavcodec/flacenc.c | 55 +++++++++++++++++++++++++++++++------------- 1 file changed, 39 insertions(+), 16 deletions(-) diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c index b00df9580b..93d4646604 100644 --- a/libavcodec/flacenc.c +++ b/libavcodec/flacenc.c @@ -43,7 +43,11 @@ #define MAX_PARTITIONS (1 << MAX_PARTITION_ORDER) #define MAX_LPC_PRECISION 15 #define MAX_LPC_SHIFT 15 -#define MAX_RICE_PARAM 14 + +enum CodingMode { + CODING_MODE_RICE = 4, + CODING_MODE_RICE2 = 5, +}; typedef struct CompressionOptions { int compression_level; @@ -60,6 +64,7 @@ typedef struct CompressionOptions { } CompressionOptions; typedef struct RiceContext { + enum CodingMode coding_mode; int porder; int params[MAX_PARTITIONS]; } RiceContext; @@ -435,8 +440,15 @@ static void init_frame(FlacEncodeContext *s, int nb_samples) } for (ch = 0; ch < s->channels; ch++) { - frame->subframes[ch].wasted = 0; - frame->subframes[ch].obits = s->avctx->bits_per_raw_sample; + FlacSubframe *sub = &frame->subframes[ch]; + + sub->wasted = 0; + sub->obits = s->avctx->bits_per_raw_sample; + + if (sub->obits > 16) + sub->rc.coding_mode = CODING_MODE_RICE2; + else + sub->rc.coding_mode = CODING_MODE_RICE; } frame->verbatim_only = 0; @@ -518,7 +530,7 @@ static uint64_t subframe_count_exact(FlacEncodeContext *s, FlacSubframe *sub, part_end = psize; for (p = 0; p < 1 << porder; p++) { int k = sub->rc.params[p]; - count += 4; + count += sub->rc.coding_mode; count += rice_count_exact(&sub->residual[i], part_end - i, k); i = part_end; part_end = FFMIN(s->frame.blocksize, part_end + psize); @@ -534,7 +546,7 @@ static uint64_t subframe_count_exact(FlacEncodeContext *s, FlacSubframe *sub, /** * Solve for d/dk(rice_encode_count) = n-((sum-(n>>1))>>(k+1)) = 0. */ -static int find_optimal_param(uint64_t sum, int n) +static int find_optimal_param(uint64_t sum, int n, int max_param) { int k; uint64_t sum2; @@ -543,7 +555,7 @@ static int find_optimal_param(uint64_t sum, int n) return 0; sum2 = sum - (n >> 1); k = av_log2(av_clipl_int32(sum2 / n)); - return FFMIN(k, MAX_RICE_PARAM); + return FFMIN(k, max_param); } @@ -551,15 +563,17 @@ static uint64_t calc_optimal_rice_params(RiceContext *rc, int porder, uint64_t *sums, int n, int pred_order) { int i; - int k, cnt, part; + int k, cnt, part, max_param; uint64_t all_bits; + max_param = (1 << rc->coding_mode) - 2; + part = (1 << porder); all_bits = 4 * part; cnt = (n >> porder) - pred_order; for (i = 0; i < part; i++) { - k = find_optimal_param(sums[i], cnt); + k = find_optimal_param(sums[i], cnt, max_param); rc->params[i] = k; all_bits += rice_encode_count(sums[i], cnt, k); cnt = n >> porder; @@ -612,6 +626,8 @@ static uint64_t calc_rice_params(RiceContext *rc, int pmin, int pmax, assert(pmax >= 0 && pmax <= MAX_PARTITION_ORDER); assert(pmin <= pmax); + tmp_rc.coding_mode = rc->coding_mode; + udata = av_malloc(n * sizeof(uint32_t)); for (i = 0; i < n; i++) udata[i] = (2*data[i]) ^ (data[i]>>31); @@ -650,7 +666,7 @@ static uint64_t find_subframe_rice_params(FlacEncodeContext *s, int pmax = get_max_p_order(s->options.max_partition_order, s->frame.blocksize, pred_order); - uint64_t bits = 8 + pred_order * sub->obits + 2 + 4; + uint64_t bits = 8 + pred_order * sub->obits + 2 + sub->rc.coding_mode; if (sub->type == FLAC_SUBFRAME_LPC) bits += 4 + 5 + pred_order * s->options.lpc_coeff_precision; bits += calc_rice_params(&sub->rc, pmin, pmax, sub->residual, @@ -925,12 +941,18 @@ static void remove_wasted_bits(FlacEncodeContext *s) sub->wasted = v; sub->obits -= v; + + /* for 24-bit, check if removing wasted bits makes the range better + suited for using RICE instead of RICE2 for entropy coding */ + if (sub->obits <= 17) + sub->rc.coding_mode = CODING_MODE_RICE; } } } -static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) +static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n, + int max_rice_param) { int i, best; int32_t lt, rt; @@ -950,7 +972,7 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) } /* estimate bit counts */ for (i = 0; i < 4; i++) { - k = find_optimal_param(2 * sum[i], n); + k = find_optimal_param(2 * sum[i], n, max_rice_param); sum[i] = rice_encode_count( 2 * sum[i], n, k); } @@ -989,9 +1011,10 @@ static void channel_decorrelation(FlacEncodeContext *s) return; } - if (s->options.ch_mode < 0) - frame->ch_mode = estimate_stereo_mode(left, right, n); - else + if (s->options.ch_mode < 0) { + int max_rice_param = (1 << frame->subframes[0].rc.coding_mode) - 2; + frame->ch_mode = estimate_stereo_mode(left, right, n, max_rice_param); + } else frame->ch_mode = s->options.ch_mode; /* perform decorrelation and adjust bits-per-sample */ @@ -1100,7 +1123,7 @@ static void write_subframes(FlacEncodeContext *s) } /* rice-encoded block */ - put_bits(&s->pb, 2, 0); + put_bits(&s->pb, 2, sub->rc.coding_mode - 4); /* partition order */ porder = sub->rc.porder; @@ -1111,7 +1134,7 @@ static void write_subframes(FlacEncodeContext *s) part_end = &sub->residual[psize]; for (p = 0; p < 1 << porder; p++) { int k = sub->rc.params[p]; - put_bits(&s->pb, 4, k); + put_bits(&s->pb, sub->rc.coding_mode, k); while (res < part_end) set_sr_golomb_flac(&s->pb, *res++, k, INT32_MAX, 0); part_end = FFMIN(frame_end, part_end + psize); From 31c3b9c95e2b14189f2bc9f2bd886834114da81d Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Sat, 27 Oct 2012 01:22:07 -0400 Subject: [PATCH 4/5] FATE: rename FLAC tests from flac-* to flac-16-* This will distinguish FLAC 16-bit tests from FLAC tests with other bit depths. --- tests/fate/flac.mak | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/tests/fate/flac.mak b/tests/fate/flac.mak index dc4992827d..f7ac342e24 100644 --- a/tests/fate/flac.mak +++ b/tests/fate/flac.mak @@ -1,17 +1,18 @@ -FATE_FLAC += fate-flac-chmode-indep \ - fate-flac-chmode-left_side \ - fate-flac-chmode-mid_side \ - fate-flac-chmode-right_side \ - fate-flac-fixed \ - fate-flac-lpc-cholesky \ - fate-flac-lpc-levinson \ +FATE_FLAC += fate-flac-16-chmode-indep \ + fate-flac-16-chmode-left_side \ + fate-flac-16-chmode-mid_side \ + fate-flac-16-chmode-right_side \ + fate-flac-16-fixed \ + fate-flac-16-lpc-cholesky \ + fate-flac-16-lpc-levinson \ -fate-flac-chmode-%: OPTS = -ch_mode $(@:fate-flac-chmode-%=%) -fate-flac-fixed: OPTS = -lpc_type fixed -fate-flac-lpc-%: OPTS = -lpc_type $(@:fate-flac-lpc-%=%) +fate-flac-16-chmode-%: OPTS = -ch_mode $(@:fate-flac-16-chmode-%=%) +fate-flac-16-fixed: OPTS = -lpc_type fixed +fate-flac-16-lpc-%: OPTS = -lpc_type $(@:fate-flac-16-lpc-%=%) + +fate-flac-16-%: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav +fate-flac-16-%: CMD = enc_dec_pcm flac wav s16le $(REF) -c flac $(OPTS) -fate-flac-%: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -fate-flac-%: CMD = enc_dec_pcm flac wav s16le $(REF) -c flac $(OPTS) fate-flac-%: CMP = oneoff fate-flac-%: FUZZ = 0 From 8a58894fc63c9d367c4cd6a17e277d1a8608c2c0 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Mon, 5 Nov 2012 12:49:36 -0500 Subject: [PATCH 5/5] FATE: add a 24-bit FLAC encoding test --- tests/fate/flac.mak | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/tests/fate/flac.mak b/tests/fate/flac.mak index f7ac342e24..92eb7430db 100644 --- a/tests/fate/flac.mak +++ b/tests/fate/flac.mak @@ -5,6 +5,7 @@ FATE_FLAC += fate-flac-16-chmode-indep \ fate-flac-16-fixed \ fate-flac-16-lpc-cholesky \ fate-flac-16-lpc-levinson \ + fate-flac-24-comp-8 \ fate-flac-16-chmode-%: OPTS = -ch_mode $(@:fate-flac-16-chmode-%=%) fate-flac-16-fixed: OPTS = -lpc_type fixed @@ -13,6 +14,11 @@ fate-flac-16-lpc-%: OPTS = -lpc_type $(@:fate-flac-16-lpc-%=%) fate-flac-16-%: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav fate-flac-16-%: CMD = enc_dec_pcm flac wav s16le $(REF) -c flac $(OPTS) +fate-flac-24-comp-%: OPTS = -compression_level $(@:fate-flac-24-comp-%=%) + +fate-flac-24-%: REF = $(SAMPLES)/audio-reference/divertimenti_2ch_96kHz_s24.wav +fate-flac-24-%: CMD = enc_dec_pcm flac wav s24le $(REF) -c flac $(OPTS) + fate-flac-%: CMP = oneoff fate-flac-%: FUZZ = 0