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@ -25,16 +25,15 @@ |
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* @author Ronald S. Bultje <rsbultje@gmail.com> |
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*/ |
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#define UNCHECKED_BITSTREAM_READER 1 |
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#include <math.h> |
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#include "libavutil/channel_layout.h" |
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#include "libavutil/float_dsp.h" |
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#include "libavutil/mem.h" |
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#include "avcodec.h" |
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#include "bitstream.h" |
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#include "internal.h" |
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#include "get_bits.h" |
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#include "put_bits.h" |
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#include "wmavoice_data.h" |
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#include "celp_filters.h" |
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@ -136,7 +135,7 @@ typedef struct WMAVoiceContext { |
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* @name Global values specified in the stream header / extradata or used all over. |
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* @{ |
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*/ |
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GetBitContext gb; ///< packet bitreader. During decoder init,
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BitstreamContext bc; ///< packet bitreader. During decoder init,
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///< it contains the extradata from the
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///< demuxer. During decoding, it contains
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///< packet data.
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@ -296,20 +295,20 @@ typedef struct WMAVoiceContext { |
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/**
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* Set up the variable bit mode (VBM) tree from container extradata. |
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* @param gb bit I/O context. |
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* The bit context (s->gb) should be loaded with byte 23-46 of the |
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* @param bc bit I/O context. |
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* The bit context (s->bc) should be loaded with byte 23-46 of the |
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* container extradata (i.e. the ones containing the VBM tree). |
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* @param vbm_tree pointer to array to which the decoded VBM tree will be |
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* written. |
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* @return 0 on success, <0 on error. |
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*/ |
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static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) |
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static av_cold int decode_vbmtree(BitstreamContext *bc, int8_t vbm_tree[25]) |
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{ |
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int cntr[8] = { 0 }, n, res; |
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memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25); |
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for (n = 0; n < 17; n++) { |
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res = get_bits(gb, 3); |
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res = bitstream_read(bc, 3); |
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if (cntr[res] > 3) // should be >= 3 + (res == 7))
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return -1; |
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vbm_tree[res * 3 + cntr[res]++] = n; |
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@ -401,8 +400,8 @@ static av_cold int wmavoice_decode_init(AVCodecContext *ctx) |
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for (n = 0; n < s->lsps; n++) |
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s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); |
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init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3); |
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if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) { |
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bitstream_init8(&s->bc, ctx->extradata + 22, ctx->extradata_size - 22); |
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if (decode_vbmtree(&s->bc, s->vbm_tree) < 0) { |
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av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n"); |
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return AVERROR_INVALIDDATA; |
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} |
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@ -857,7 +856,7 @@ static void dequant_lsps(double *lsps, int num, |
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/**
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* Parse 10 independently-coded LSPs. |
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*/ |
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static void dequant_lsp10i(GetBitContext *gb, double *lsps) |
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static void dequant_lsp10i(BitstreamContext *bc, double *lsps) |
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{ |
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static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; |
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static const double mul_lsf[4] = { |
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@ -870,10 +869,10 @@ static void dequant_lsp10i(GetBitContext *gb, double *lsps) |
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}; |
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uint16_t v[4]; |
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v[0] = get_bits(gb, 8); |
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v[1] = get_bits(gb, 6); |
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v[2] = get_bits(gb, 5); |
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v[3] = get_bits(gb, 5); |
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v[0] = bitstream_read(bc, 8); |
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v[1] = bitstream_read(bc, 6); |
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v[2] = bitstream_read(bc, 5); |
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v[3] = bitstream_read(bc, 5); |
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dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, |
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mul_lsf, base_lsf); |
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@ -883,7 +882,7 @@ static void dequant_lsp10i(GetBitContext *gb, double *lsps) |
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* Parse 10 independently-coded LSPs, and then derive the tables to |
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* generate LSPs for the other frames from them (residual coding). |
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*/ |
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static void dequant_lsp10r(GetBitContext *gb, |
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static void dequant_lsp10r(BitstreamContext *bc, |
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double *i_lsps, const double *old, |
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double *a1, double *a2, int q_mode) |
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{ |
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@ -899,12 +898,12 @@ static void dequant_lsp10r(GetBitContext *gb, |
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uint16_t interpol, v[3]; |
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int n; |
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dequant_lsp10i(gb, i_lsps); |
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dequant_lsp10i(bc, i_lsps); |
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interpol = get_bits(gb, 5); |
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v[0] = get_bits(gb, 7); |
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v[1] = get_bits(gb, 6); |
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v[2] = get_bits(gb, 6); |
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interpol = bitstream_read(bc, 5); |
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v[0] = bitstream_read(bc, 7); |
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v[1] = bitstream_read(bc, 6); |
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v[2] = bitstream_read(bc, 6); |
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for (n = 0; n < 10; n++) { |
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double delta = old[n] - i_lsps[n]; |
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@ -919,7 +918,7 @@ static void dequant_lsp10r(GetBitContext *gb, |
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/**
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* Parse 16 independently-coded LSPs. |
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*/ |
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static void dequant_lsp16i(GetBitContext *gb, double *lsps) |
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static void dequant_lsp16i(BitstreamContext *bc, double *lsps) |
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{ |
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static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; |
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static const double mul_lsf[5] = { |
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@ -934,11 +933,11 @@ static void dequant_lsp16i(GetBitContext *gb, double *lsps) |
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}; |
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uint16_t v[5]; |
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v[0] = get_bits(gb, 8); |
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v[1] = get_bits(gb, 6); |
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v[2] = get_bits(gb, 7); |
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v[3] = get_bits(gb, 6); |
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v[4] = get_bits(gb, 7); |
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v[0] = bitstream_read(bc, 8); |
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v[1] = bitstream_read(bc, 6); |
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v[2] = bitstream_read(bc, 7); |
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v[3] = bitstream_read(bc, 6); |
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v[4] = bitstream_read(bc, 7); |
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dequant_lsps( lsps, 5, v, vec_sizes, 2, |
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wmavoice_dq_lsp16i1, mul_lsf, base_lsf); |
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@ -952,7 +951,7 @@ static void dequant_lsp16i(GetBitContext *gb, double *lsps) |
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* Parse 16 independently-coded LSPs, and then derive the tables to |
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* generate LSPs for the other frames from them (residual coding). |
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*/ |
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static void dequant_lsp16r(GetBitContext *gb, |
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static void dequant_lsp16r(BitstreamContext *bc, |
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double *i_lsps, const double *old, |
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double *a1, double *a2, int q_mode) |
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{ |
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@ -968,12 +967,12 @@ static void dequant_lsp16r(GetBitContext *gb, |
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uint16_t interpol, v[3]; |
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int n; |
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dequant_lsp16i(gb, i_lsps); |
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dequant_lsp16i(bc, i_lsps); |
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interpol = get_bits(gb, 5); |
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v[0] = get_bits(gb, 7); |
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v[1] = get_bits(gb, 7); |
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v[2] = get_bits(gb, 7); |
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interpol = bitstream_read(bc, 5); |
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v[0] = bitstream_read(bc, 7); |
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v[1] = bitstream_read(bc, 7); |
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v[2] = bitstream_read(bc, 7); |
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for (n = 0; n < 16; n++) { |
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double delta = old[n] - i_lsps[n]; |
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@ -999,10 +998,10 @@ static void dequant_lsp16r(GetBitContext *gb, |
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* Parse the offset of the first pitch-adaptive window pulses, and |
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* the distribution of pulses between the two blocks in this frame. |
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* @param s WMA Voice decoding context private data |
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* @param gb bit I/O context |
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* @param bc bit I/O context |
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* @param pitch pitch for each block in this frame |
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*/ |
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static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, |
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static void aw_parse_coords(WMAVoiceContext *s, BitstreamContext *bc, |
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const int *pitch) |
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{ |
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static const int16_t start_offset[94] = { |
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@ -1019,9 +1018,9 @@ static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, |
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/* position of pulse */ |
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s->aw_idx_is_ext = 0; |
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if ((bits = get_bits(gb, 6)) >= 54) { |
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if ((bits = bitstream_read(bc, 6)) >= 54) { |
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s->aw_idx_is_ext = 1; |
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bits += (bits - 54) * 3 + get_bits(gb, 2); |
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bits += (bits - 54) * 3 + bitstream_read(bc, 2); |
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} |
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/* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
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@ -1049,12 +1048,12 @@ static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, |
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/**
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* Apply second set of pitch-adaptive window pulses. |
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* @param s WMA Voice decoding context private data |
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* @param gb bit I/O context |
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* @param bc bit I/O context |
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* @param block_idx block index in frame [0, 1] |
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* @param fcb structure containing fixed codebook vector info |
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* @return -1 on error, 0 otherwise |
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*/ |
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static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, |
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static int aw_pulse_set2(WMAVoiceContext *s, BitstreamContext *bc, |
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int block_idx, AMRFixed *fcb) |
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{ |
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uint16_t use_mask_mem[9]; // only 5 are used, rest is padding
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@ -1108,7 +1107,7 @@ static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, |
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} |
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/* find the 'aidx'th offset that is not excluded */ |
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aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); |
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aidx = bitstream_read(bc, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); |
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for (n = 0; n <= aidx; pulse_start++) { |
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for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; |
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if (idx >= MAX_FRAMESIZE / 2) { // find from zero
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@ -1128,7 +1127,7 @@ static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, |
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} |
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fcb->x[fcb->n] = start_off; |
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fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; |
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fcb->y[fcb->n] = bitstream_read_bit(bc) ? -1.0 : 1.0; |
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fcb->n++; |
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/* set offset for next block, relative to start of that block */ |
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@ -1140,14 +1139,14 @@ static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, |
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/**
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* Apply first set of pitch-adaptive window pulses. |
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* @param s WMA Voice decoding context private data |
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* @param gb bit I/O context |
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* @param bc bit I/O context |
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* @param block_idx block index in frame [0, 1] |
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* @param fcb storage location for fixed codebook pulse info |
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*/ |
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static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, |
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static void aw_pulse_set1(WMAVoiceContext *s, BitstreamContext *bc, |
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int block_idx, AMRFixed *fcb) |
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{ |
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int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); |
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int val = bitstream_read(bc, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); |
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float v; |
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if (s->aw_n_pulses[block_idx] > 0) { |
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@ -1241,7 +1240,7 @@ static int pRNG(int frame_cntr, int block_num, int block_size) |
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* Parse hardcoded signal for a single block. |
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* @note see #synth_block(). |
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*/ |
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static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, |
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static void synth_block_hardcoded(WMAVoiceContext *s, BitstreamContext *bc, |
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int block_idx, int size, |
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const struct frame_type_desc *frame_desc, |
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float *excitation) |
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@ -1256,8 +1255,8 @@ static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, |
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r_idx = pRNG(s->frame_cntr, block_idx, size); |
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gain = s->silence_gain; |
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} else /* FCB_TYPE_HARDCODED */ { |
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r_idx = get_bits(gb, 8); |
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gain = wmavoice_gain_universal[get_bits(gb, 6)]; |
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r_idx = bitstream_read(bc, 8); |
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gain = wmavoice_gain_universal[bitstream_read(bc, 6)]; |
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} |
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/* Clear gain prediction parameters */ |
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@ -1272,7 +1271,7 @@ static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, |
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* Parse FCB/ACB signal for a single block. |
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* @note see #synth_block(). |
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*/ |
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static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, |
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static void synth_block_fcb_acb(WMAVoiceContext *s, BitstreamContext *bc, |
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int block_idx, int size, |
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int block_pitch_sh2, |
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const struct frame_type_desc *frame_desc, |
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@ -1296,8 +1295,8 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, |
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/* For the other frame types, this is where we apply the innovation
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* (fixed) codebook pulses of the speech signal. */ |
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if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { |
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aw_pulse_set1(s, gb, block_idx, &fcb); |
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if (aw_pulse_set2(s, gb, block_idx, &fcb)) { |
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aw_pulse_set1(s, bc, block_idx, &fcb); |
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if (aw_pulse_set2(s, bc, block_idx, &fcb)) { |
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/* Conceal the block with silence and return.
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* Skip the correct amount of bits to read the next |
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* block from the correct offset. */ |
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@ -1306,7 +1305,7 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, |
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for (n = 0; n < size; n++) |
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excitation[n] = |
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wmavoice_std_codebook[r_idx + n] * s->silence_gain; |
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skip_bits(gb, 7 + 1); |
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|
|
|
bitstream_skip(bc, 7 + 1); |
|
|
|
|
return; |
|
|
|
|
} |
|
|
|
|
} else /* FCB_TYPE_EXC_PULSES */ { |
|
|
|
|
@ -1319,12 +1318,12 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, |
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|
|
|
float sign; |
|
|
|
|
int pos1, pos2; |
|
|
|
|
|
|
|
|
|
sign = get_bits1(gb) ? 1.0 : -1.0; |
|
|
|
|
pos1 = get_bits(gb, offset_nbits); |
|
|
|
|
sign = bitstream_read_bit(bc) ? 1.0 : -1.0; |
|
|
|
|
pos1 = bitstream_read(bc, offset_nbits); |
|
|
|
|
fcb.x[fcb.n] = n + 5 * pos1; |
|
|
|
|
fcb.y[fcb.n++] = sign; |
|
|
|
|
if (n < frame_desc->dbl_pulses) { |
|
|
|
|
pos2 = get_bits(gb, offset_nbits); |
|
|
|
|
pos2 = bitstream_read(bc, offset_nbits); |
|
|
|
|
fcb.x[fcb.n] = n + 5 * pos2; |
|
|
|
|
fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; |
|
|
|
|
} |
|
|
|
|
@ -1334,7 +1333,7 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, |
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|
|
|
|
|
|
|
/* Calculate gain for adaptive & fixed codebook signal.
|
|
|
|
|
* see ff_amr_set_fixed_gain(). */ |
|
|
|
|
idx = get_bits(gb, 7); |
|
|
|
|
idx = bitstream_read(bc, 7); |
|
|
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|
fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err, |
|
|
|
|
gain_coeff, 6) - |
|
|
|
|
5.2409161640 + wmavoice_gain_codebook_fcb[idx]); |
|
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|
|
@ -1396,7 +1395,7 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, |
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|
|
|
* @note we assume enough bits are available, caller should check. |
|
|
|
|
* |
|
|
|
|
* @param s WMA Voice decoding context private data |
|
|
|
|
* @param gb bit I/O context |
|
|
|
|
* @param bc bit I/O context |
|
|
|
|
* @param block_idx index of the to-be-read block |
|
|
|
|
* @param size amount of samples to be read in this block |
|
|
|
|
* @param block_pitch_sh2 pitch for this block << 2 |
|
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|
@ -1407,7 +1406,7 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, |
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|
|
|
* @param synth target memory for the speech synthesis filter output |
|
|
|
|
* @return 0 on success, <0 on error. |
|
|
|
|
*/ |
|
|
|
|
static void synth_block(WMAVoiceContext *s, GetBitContext *gb, |
|
|
|
|
static void synth_block(WMAVoiceContext *s, BitstreamContext *bc, |
|
|
|
|
int block_idx, int size, |
|
|
|
|
int block_pitch_sh2, |
|
|
|
|
const double *lsps, const double *prev_lsps, |
|
|
|
|
@ -1420,9 +1419,9 @@ static void synth_block(WMAVoiceContext *s, GetBitContext *gb, |
|
|
|
|
int n; |
|
|
|
|
|
|
|
|
|
if (frame_desc->acb_type == ACB_TYPE_NONE) |
|
|
|
|
synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); |
|
|
|
|
synth_block_hardcoded(s, bc, block_idx, size, frame_desc, excitation); |
|
|
|
|
else |
|
|
|
|
synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, |
|
|
|
|
synth_block_fcb_acb(s, bc, block_idx, size, block_pitch_sh2, |
|
|
|
|
frame_desc, excitation); |
|
|
|
|
|
|
|
|
|
/* convert interpolated LSPs to LPCs */ |
|
|
|
|
@ -1440,7 +1439,7 @@ static void synth_block(WMAVoiceContext *s, GetBitContext *gb, |
|
|
|
|
* @note we assume enough bits are available, caller should check. |
|
|
|
|
* |
|
|
|
|
* @param ctx WMA Voice decoder context |
|
|
|
|
* @param gb bit I/O context (s->gb or one for cross-packet superframes) |
|
|
|
|
* @param bc bit I/O context (s->bc or one for cross-packet superframes) |
|
|
|
|
* @param frame_idx Frame number within superframe [0-2] |
|
|
|
|
* @param samples pointer to output sample buffer, has space for at least 160 |
|
|
|
|
* samples |
|
|
|
|
@ -1450,8 +1449,8 @@ static void synth_block(WMAVoiceContext *s, GetBitContext *gb, |
|
|
|
|
* @param synth target buffer for synthesized speech data |
|
|
|
|
* @return 0 on success, <0 on error. |
|
|
|
|
*/ |
|
|
|
|
static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, |
|
|
|
|
float *samples, |
|
|
|
|
static int synth_frame(AVCodecContext *ctx, BitstreamContext *bc, |
|
|
|
|
int frame_idx, float *samples, |
|
|
|
|
const double *lsps, const double *prev_lsps, |
|
|
|
|
float *excitation, float *synth) |
|
|
|
|
{ |
|
|
|
|
@ -1460,7 +1459,7 @@ static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, |
|
|
|
|
int pitch[MAX_BLOCKS], last_block_pitch; |
|
|
|
|
|
|
|
|
|
/* Parse frame type ("frame header"), see frame_descs */ |
|
|
|
|
int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples; |
|
|
|
|
int bd_idx = s->vbm_tree[bitstream_read_vlc(bc, frame_type_vlc.table, 6, 3)], block_nsamples; |
|
|
|
|
|
|
|
|
|
if (bd_idx < 0) { |
|
|
|
|
av_log(ctx, AV_LOG_ERROR, |
|
|
|
|
@ -1478,7 +1477,7 @@ static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, |
|
|
|
|
* incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ |
|
|
|
|
n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; |
|
|
|
|
log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; |
|
|
|
|
cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); |
|
|
|
|
cur_pitch_val = s->min_pitch_val + bitstream_read(bc, s->pitch_nbits); |
|
|
|
|
cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); |
|
|
|
|
if (s->last_acb_type == ACB_TYPE_NONE || |
|
|
|
|
20 * abs(cur_pitch_val - s->last_pitch_val) > |
|
|
|
|
@ -1502,10 +1501,10 @@ static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, |
|
|
|
|
/* Global gain (if silence) and pitch-adaptive window coordinates */ |
|
|
|
|
switch (frame_descs[bd_idx].fcb_type) { |
|
|
|
|
case FCB_TYPE_SILENCE: |
|
|
|
|
s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)]; |
|
|
|
|
s->silence_gain = wmavoice_gain_silence[bitstream_read(bc, 8)]; |
|
|
|
|
break; |
|
|
|
|
case FCB_TYPE_AW_PULSES: |
|
|
|
|
aw_parse_coords(s, gb, pitch); |
|
|
|
|
aw_parse_coords(s, bc, pitch); |
|
|
|
|
break; |
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
@ -1526,10 +1525,10 @@ static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, |
|
|
|
|
t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1; |
|
|
|
|
|
|
|
|
|
if (n == 0) { |
|
|
|
|
block_pitch = get_bits(gb, s->block_pitch_nbits); |
|
|
|
|
block_pitch = bitstream_read(bc, s->block_pitch_nbits); |
|
|
|
|
} else |
|
|
|
|
block_pitch = last_block_pitch - s->block_delta_pitch_hrange + |
|
|
|
|
get_bits(gb, s->block_delta_pitch_nbits); |
|
|
|
|
bitstream_read(bc, s->block_delta_pitch_nbits); |
|
|
|
|
/* Convert last_ so that any next delta is within _range */ |
|
|
|
|
last_block_pitch = av_clip(block_pitch, |
|
|
|
|
s->block_delta_pitch_hrange, |
|
|
|
|
@ -1567,7 +1566,7 @@ static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, |
|
|
|
|
break; |
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, |
|
|
|
|
synth_block(s, bc, n, block_nsamples, bl_pitch_sh2, |
|
|
|
|
lsps, prev_lsps, &frame_descs[bd_idx], |
|
|
|
|
&excitation[n * block_nsamples], |
|
|
|
|
&synth[n * block_nsamples]); |
|
|
|
|
@ -1658,32 +1657,32 @@ static void stabilize_lsps(double *lsps, int num) |
|
|
|
|
/**
|
|
|
|
|
* Test if there's enough bits to read 1 superframe. |
|
|
|
|
* |
|
|
|
|
* @param orig_gb bit I/O context used for reading. This function |
|
|
|
|
* @param orig_bc bit I/O context used for reading. This function |
|
|
|
|
* does not modify the state of the bitreader; it |
|
|
|
|
* only uses it to copy the current stream position |
|
|
|
|
* @param s WMA Voice decoding context private data |
|
|
|
|
* @return < 0 on error, 1 on not enough bits or 0 if OK. |
|
|
|
|
*/ |
|
|
|
|
static int check_bits_for_superframe(GetBitContext *orig_gb, |
|
|
|
|
static int check_bits_for_superframe(BitstreamContext *orig_bc, |
|
|
|
|
WMAVoiceContext *s) |
|
|
|
|
{ |
|
|
|
|
GetBitContext s_gb, *gb = &s_gb; |
|
|
|
|
BitstreamContext s_bc, *bc = &s_bc; |
|
|
|
|
int n, need_bits, bd_idx; |
|
|
|
|
const struct frame_type_desc *frame_desc; |
|
|
|
|
|
|
|
|
|
/* initialize a copy */ |
|
|
|
|
*gb = *orig_gb; |
|
|
|
|
*bc = *orig_bc; |
|
|
|
|
|
|
|
|
|
/* superframe header */ |
|
|
|
|
if (get_bits_left(gb) < 14) |
|
|
|
|
if (bitstream_bits_left(bc) < 14) |
|
|
|
|
return 1; |
|
|
|
|
if (!get_bits1(gb)) |
|
|
|
|
if (!bitstream_read_bit(bc)) |
|
|
|
|
return AVERROR(ENOSYS); // WMAPro-in-WMAVoice superframe
|
|
|
|
|
if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe
|
|
|
|
|
if (bitstream_read_bit(bc)) bitstream_skip(bc, 12); // number of samples in superframe
|
|
|
|
|
if (s->has_residual_lsps) { // residual LSPs (for all frames)
|
|
|
|
|
if (get_bits_left(gb) < s->sframe_lsp_bitsize) |
|
|
|
|
if (bitstream_bits_left(bc) < s->sframe_lsp_bitsize) |
|
|
|
|
return 1; |
|
|
|
|
skip_bits_long(gb, s->sframe_lsp_bitsize); |
|
|
|
|
bitstream_skip(bc, s->sframe_lsp_bitsize); |
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
/* frames */ |
|
|
|
|
@ -1691,24 +1690,25 @@ static int check_bits_for_superframe(GetBitContext *orig_gb, |
|
|
|
|
int aw_idx_is_ext = 0; |
|
|
|
|
|
|
|
|
|
if (!s->has_residual_lsps) { // independent LSPs (per-frame)
|
|
|
|
|
if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1; |
|
|
|
|
skip_bits_long(gb, s->frame_lsp_bitsize); |
|
|
|
|
if (bitstream_bits_left(bc) < s->frame_lsp_bitsize) |
|
|
|
|
return 1; |
|
|
|
|
bitstream_skip(bc, s->frame_lsp_bitsize); |
|
|
|
|
} |
|
|
|
|
bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)]; |
|
|
|
|
bd_idx = s->vbm_tree[bitstream_read_vlc(bc, frame_type_vlc.table, 6, 3)]; |
|
|
|
|
if (bd_idx < 0) |
|
|
|
|
return AVERROR_INVALIDDATA; // invalid frame type VLC code
|
|
|
|
|
frame_desc = &frame_descs[bd_idx]; |
|
|
|
|
if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { |
|
|
|
|
if (get_bits_left(gb) < s->pitch_nbits) |
|
|
|
|
if (bitstream_bits_left(bc) < s->pitch_nbits) |
|
|
|
|
return 1; |
|
|
|
|
skip_bits_long(gb, s->pitch_nbits); |
|
|
|
|
bitstream_skip(bc, s->pitch_nbits); |
|
|
|
|
} |
|
|
|
|
if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { |
|
|
|
|
skip_bits(gb, 8); |
|
|
|
|
bitstream_skip(bc, 8); |
|
|
|
|
} else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { |
|
|
|
|
int tmp = get_bits(gb, 6); |
|
|
|
|
int tmp = bitstream_read(bc, 6); |
|
|
|
|
if (tmp >= 0x36) { |
|
|
|
|
skip_bits(gb, 2); |
|
|
|
|
bitstream_skip(bc, 2); |
|
|
|
|
aw_idx_is_ext = 1; |
|
|
|
|
} |
|
|
|
|
} |
|
|
|
|
@ -1722,9 +1722,9 @@ static int check_bits_for_superframe(GetBitContext *orig_gb, |
|
|
|
|
} else |
|
|
|
|
need_bits = 0; |
|
|
|
|
need_bits += frame_desc->frame_size; |
|
|
|
|
if (get_bits_left(gb) < need_bits) |
|
|
|
|
if (bitstream_bits_left(bc) < need_bits) |
|
|
|
|
return 1; |
|
|
|
|
skip_bits_long(gb, need_bits); |
|
|
|
|
bitstream_skip(bc, need_bits); |
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
return 0; |
|
|
|
|
@ -1733,7 +1733,7 @@ static int check_bits_for_superframe(GetBitContext *orig_gb, |
|
|
|
|
/**
|
|
|
|
|
* Synthesize output samples for a single superframe. If we have any data |
|
|
|
|
* cached in s->sframe_cache, that will be used instead of whatever is loaded |
|
|
|
|
* in s->gb. |
|
|
|
|
* in s->bc. |
|
|
|
|
* |
|
|
|
|
* WMA Voice superframes contain 3 frames, each containing 160 audio samples, |
|
|
|
|
* to give a total of 480 samples per frame. See #synth_frame() for frame |
|
|
|
|
@ -1751,7 +1751,7 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, |
|
|
|
|
int *got_frame_ptr) |
|
|
|
|
{ |
|
|
|
|
WMAVoiceContext *s = ctx->priv_data; |
|
|
|
|
GetBitContext *gb = &s->gb, s_gb; |
|
|
|
|
BitstreamContext *bc = &s->bc, s_bc; |
|
|
|
|
int n, res, n_samples = 480; |
|
|
|
|
double lsps[MAX_FRAMES][MAX_LSPS]; |
|
|
|
|
const double *mean_lsf = s->lsps == 16 ? |
|
|
|
|
@ -1766,12 +1766,12 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, |
|
|
|
|
s->history_nsamples * sizeof(*excitation)); |
|
|
|
|
|
|
|
|
|
if (s->sframe_cache_size > 0) { |
|
|
|
|
gb = &s_gb; |
|
|
|
|
init_get_bits(gb, s->sframe_cache, s->sframe_cache_size); |
|
|
|
|
bc = &s_bc; |
|
|
|
|
bitstream_init(bc, s->sframe_cache, s->sframe_cache_size); |
|
|
|
|
s->sframe_cache_size = 0; |
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
if ((res = check_bits_for_superframe(gb, s)) == 1) { |
|
|
|
|
if ((res = check_bits_for_superframe(bc, s)) == 1) { |
|
|
|
|
*got_frame_ptr = 0; |
|
|
|
|
return 1; |
|
|
|
|
} else if (res < 0) |
|
|
|
|
@ -1781,14 +1781,14 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, |
|
|
|
|
* speech samples (the actual codec) and WMAVoice music samples, which |
|
|
|
|
* are really WMAPro-in-WMAVoice-superframes. I've never seen those in |
|
|
|
|
* the wild yet. */ |
|
|
|
|
if (!get_bits1(gb)) { |
|
|
|
|
if (!bitstream_read_bit(bc)) { |
|
|
|
|
avpriv_request_sample(ctx, "WMAPro-in-WMAVoice"); |
|
|
|
|
return AVERROR_PATCHWELCOME; |
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
/* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ |
|
|
|
|
if (get_bits1(gb)) { |
|
|
|
|
if ((n_samples = get_bits(gb, 12)) > 480) { |
|
|
|
|
if (bitstream_read_bit(bc)) { |
|
|
|
|
if ((n_samples = bitstream_read(bc, 12)) > 480) { |
|
|
|
|
av_log(ctx, AV_LOG_ERROR, |
|
|
|
|
"Superframe encodes >480 samples (%d), not allowed\n", |
|
|
|
|
n_samples); |
|
|
|
|
@ -1803,9 +1803,9 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, |
|
|
|
|
prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; |
|
|
|
|
|
|
|
|
|
if (s->lsps == 10) { |
|
|
|
|
dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); |
|
|
|
|
dequant_lsp10r(bc, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); |
|
|
|
|
} else /* s->lsps == 16 */ |
|
|
|
|
dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); |
|
|
|
|
dequant_lsp16r(bc, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); |
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for (n = 0; n < s->lsps; n++) { |
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lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); |
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@ -1831,16 +1831,16 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, |
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int m; |
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if (s->lsps == 10) { |
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dequant_lsp10i(gb, lsps[n]); |
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dequant_lsp10i(bc, lsps[n]); |
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} else /* s->lsps == 16 */ |
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dequant_lsp16i(gb, lsps[n]); |
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dequant_lsp16i(bc, lsps[n]); |
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for (m = 0; m < s->lsps; m++) |
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lsps[n][m] += mean_lsf[m]; |
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stabilize_lsps(lsps[n], s->lsps); |
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} |
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if ((res = synth_frame(ctx, gb, n, |
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if ((res = synth_frame(ctx, bc, n, |
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&samples[n * MAX_FRAMESIZE], |
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lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], |
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&excitation[s->history_nsamples + n * MAX_FRAMESIZE], |
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@ -1853,9 +1853,9 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, |
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/* Statistics? FIXME - we don't check for length, a slight overrun
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* will be caught by internal buffer padding, and anything else |
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* will be skipped, not read. */ |
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if (get_bits1(gb)) { |
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res = get_bits(gb, 4); |
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skip_bits(gb, 10 * (res + 1)); |
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if (bitstream_read_bit(bc)) { |
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res = bitstream_read(bc, 4); |
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bitstream_skip(bc, 10 * (res + 1)); |
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} |
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*got_frame_ptr = 1; |
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@ -1883,31 +1883,31 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, |
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*/ |
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static int parse_packet_header(WMAVoiceContext *s) |
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{ |
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GetBitContext *gb = &s->gb; |
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BitstreamContext *bc = &s->bc; |
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unsigned int res; |
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if (get_bits_left(gb) < 11) |
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if (bitstream_bits_left(bc) < 11) |
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return 1; |
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skip_bits(gb, 4); // packet sequence number
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s->has_residual_lsps = get_bits1(gb); |
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bitstream_skip(bc, 4); // packet sequence number
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s->has_residual_lsps = bitstream_read_bit(bc); |
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do { |
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res = get_bits(gb, 6); // number of superframes per packet
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// (minus first one if there is spillover)
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if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize) |
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res = bitstream_read(bc, 6); // number of superframes per packet
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// (minus first one if there is spillover)
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if (bitstream_bits_left(bc) < 6 * (res == 0x3F) + s->spillover_bitsize) |
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return 1; |
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} while (res == 0x3F); |
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s->spillover_nbits = get_bits(gb, s->spillover_bitsize); |
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s->spillover_nbits = bitstream_read(bc, s->spillover_bitsize); |
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return 0; |
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} |
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/**
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* Copy (unaligned) bits from gb/data/size to pb. |
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* Copy (unaligned) bits from bc/data/size to pb. |
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* |
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* @param pb target buffer to copy bits into |
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* @param data source buffer to copy bits from |
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* @param size size of the source data, in bytes |
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* @param gb bit I/O context specifying the current position in the source. |
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* @param bc bit I/O context specifying the current position in the source. |
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* data. This function might use this to align the bit position to |
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* a whole-byte boundary before calling #avpriv_copy_bits() on aligned |
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* source data |
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@ -1918,18 +1918,18 @@ static int parse_packet_header(WMAVoiceContext *s) |
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*/ |
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static void copy_bits(PutBitContext *pb, |
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const uint8_t *data, int size, |
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GetBitContext *gb, int nbits) |
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BitstreamContext *bc, int nbits) |
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{ |
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int rmn_bytes, rmn_bits; |
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rmn_bits = rmn_bytes = get_bits_left(gb); |
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rmn_bits = rmn_bytes = bitstream_bits_left(bc); |
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if (rmn_bits < nbits) |
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return; |
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if (nbits > pb->size_in_bits - put_bits_count(pb)) |
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return; |
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rmn_bits &= 7; rmn_bytes >>= 3; |
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if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) |
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put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); |
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put_bits(pb, rmn_bits, bitstream_read(bc, rmn_bits)); |
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avpriv_copy_bits(pb, data + size - rmn_bytes, |
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FFMIN(nbits - rmn_bits, rmn_bytes << 3)); |
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} |
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@ -1949,7 +1949,7 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, |
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int *got_frame_ptr, AVPacket *avpkt) |
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{ |
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WMAVoiceContext *s = ctx->priv_data; |
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GetBitContext *gb = &s->gb; |
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BitstreamContext *bc = &s->bc; |
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int size, res, pos; |
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/* Packets are sometimes a multiple of ctx->block_align, with a packet
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@ -1962,7 +1962,7 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, |
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*got_frame_ptr = 0; |
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return 0; |
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} |
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init_get_bits(&s->gb, avpkt->data, size << 3); |
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bitstream_init8(&s->bc, avpkt->data, size); |
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/* size == ctx->block_align is used to indicate whether we are dealing with
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* a new packet or a packet of which we already read the packet header |
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@ -1976,8 +1976,8 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, |
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* continuing to parse new superframes in the current packet. */ |
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if (s->spillover_nbits > 0) { |
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if (s->sframe_cache_size > 0) { |
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int cnt = get_bits_count(gb); |
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copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); |
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int cnt = bitstream_tell(bc); |
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copy_bits(&s->pb, avpkt->data, size, bc, s->spillover_nbits); |
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flush_put_bits(&s->pb); |
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s->sframe_cache_size += s->spillover_nbits; |
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if ((res = synth_superframe(ctx, data, got_frame_ptr)) == 0 && |
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@ -1986,33 +1986,33 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, |
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s->skip_bits_next = cnt & 7; |
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|
return cnt >> 3; |
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|
} else |
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skip_bits_long (gb, s->spillover_nbits - cnt + |
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|
get_bits_count(gb)); // resync
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|
bitstream_skip (bc, s->spillover_nbits - cnt + |
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|
bitstream_tell(bc)); // resync
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|
} else |
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|
skip_bits_long(gb, s->spillover_nbits); // resync
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|
bitstream_skip(bc, s->spillover_nbits); // resync
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|
} |
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|
} else if (s->skip_bits_next) |
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|
skip_bits(gb, s->skip_bits_next); |
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|
bitstream_skip(bc, s->skip_bits_next); |
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|
|
/* Try parsing superframes in current packet */ |
|
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|
|
s->sframe_cache_size = 0; |
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|
|
s->skip_bits_next = 0; |
|
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|
|
pos = get_bits_left(gb); |
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|
|
pos = bitstream_bits_left(bc); |
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|
|
if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) { |
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|
|
return res; |
|
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|
|
} else if (*got_frame_ptr) { |
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|
|
int cnt = get_bits_count(gb); |
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|
|
int cnt = bitstream_tell(bc); |
|
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|
|
s->skip_bits_next = cnt & 7; |
|
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|
|
return cnt >> 3; |
|
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|
|
} else if ((s->sframe_cache_size = pos) > 0) { |
|
|
|
|
/* rewind bit reader to start of last (incomplete) superframe... */ |
|
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|
|
init_get_bits(gb, avpkt->data, size << 3); |
|
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|
|
skip_bits_long(gb, (size << 3) - pos); |
|
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|
|
assert(get_bits_left(gb) == pos); |
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|
|
bitstream_init8(bc, avpkt->data, size); |
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|
|
bitstream_skip(bc, (size << 3) - pos); |
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|
|
assert(bitstream_bits_left(bc) == pos); |
|
|
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|
|
|
|
|
|
/* ...and cache it for spillover in next packet */ |
|
|
|
|
init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); |
|
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|
|
copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); |
|
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|
|
copy_bits(&s->pb, avpkt->data, size, bc, s->sframe_cache_size); |
|
|
|
|
// FIXME bad - just copy bytes as whole and add use the
|
|
|
|
|
// skip_bits_next field
|
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|
|
} |
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Alexandra Hájková
9 years ago
committed by
Diego Biurrun