#include <math.h>
+#include "libavutil/channel_layout.h"
+#include "libavutil/float_dsp.h"
#include "libavutil/mem.h"
-#include "dsputil.h"
#include "avcodec.h"
+#include "internal.h"
#include "get_bits.h"
#include "put_bits.h"
#include "wmavoice_data.h"
* @name Global values specified in the stream header / extradata or used all over.
* @{
*/
- AVFrame frame;
GetBitContext gb; ///< packet bitreader. During decoder init,
///< it contains the extradata from the
///< demuxer. During decoding, it contains
* @return 0 on success, <0 on error.
*/
static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
+{
+ int cntr[8] = { 0 }, n, res;
+
+ memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25);
+ for (n = 0; n < 17; n++) {
+ res = get_bits(gb, 3);
+ if (cntr[res] > 3) // should be >= 3 + (res == 7))
+ return -1;
+ vbm_tree[res * 3 + cntr[res]++] = n;
+ }
+ return 0;
+}
+
+static av_cold void wmavoice_init_static_data(AVCodec *codec)
{
static const uint8_t bits[] = {
2, 2, 2, 4, 4, 4,
0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10
0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
};
- int cntr[8] = { 0 }, n, res;
- memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25);
- for (n = 0; n < 17; n++) {
- res = get_bits(gb, 3);
- if (cntr[res] > 3) // should be >= 3 + (res == 7))
- return -1;
- vbm_tree[res * 3 + cntr[res]++] = n;
- }
INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
bits, 1, 1, codes, 2, 2, 132);
- return 0;
}
/**
av_log(ctx, AV_LOG_ERROR,
"Invalid extradata size %d (should be 46)\n",
ctx->extradata_size);
- return -1;
+ return AVERROR_INVALIDDATA;
}
flags = AV_RL32(ctx->extradata + 18);
s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
av_log(ctx, AV_LOG_ERROR,
"Invalid denoise filter strength %d (max=11)\n",
s->denoise_strength);
- return -1;
+ return AVERROR_INVALIDDATA;
}
s->denoise_tilt_corr = !!(flags & 0x40);
s->dc_level = (flags >> 7) & 0xF;
init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8;
pitch_range = s->max_pitch_val - s->min_pitch_val;
if (pitch_range <= 0) {
av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
s->pitch_nbits = av_ceil_log2(pitch_range);
s->last_pitch_val = 40;
"Unsupported samplerate %d (min=%d, max=%d)\n",
ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
- return -1;
+ return AVERROR(ENOSYS);
}
s->block_conv_table[0] = s->min_pitch_val;
s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
if (s->block_delta_pitch_hrange <= 0) {
av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
s->block_pitch_range = s->block_conv_table[2] +
2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range);
+ ctx->channels = 1;
+ ctx->channel_layout = AV_CH_LAYOUT_MONO;
ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
- avcodec_get_frame_defaults(&s->frame);
- ctx->coded_frame = &s->frame;
-
return 0;
}
/* find best fitting point in history */
do {
- dot = ff_scalarproduct_float_c(in, ptr, size);
+ dot = avpriv_scalarproduct_float_c(in, ptr, size);
if (dot > optimal_gain) {
optimal_gain = dot;
best_hist_ptr = ptr;
if (optimal_gain <= 0)
return -1;
- dot = ff_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
+ dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
if (dot <= 0) // would be 1.0
return -1;
{
float rh0, rh1;
- rh0 = 1.0 + ff_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
- rh1 = lpcs[0] + ff_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
+ rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
+ rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
return rh1 / rh0;
}
/* 70.57 =~ 1/log10(1.0331663) */
idx = (pwr * gain_mul - 0.0295) * 70.570526123;
- if (idx > 127) { // fallback if index falls outside table range
+ if (idx > 127) { // fall back if index falls outside table range
coeffs[n] = wmavoice_energy_table[127] *
powf(1.0331663, idx - 127);
} else
}
/* calculate the Hilbert transform of the gains, which we do (since this
- * is a sinus input) by doing a phase shift (in theory, H(sin())=cos()).
+ * is a sine input) by doing a phase shift (in theory, H(sin())=cos()).
* Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
* "moment" of the LPCs in this filter. */
s->dct.dct_calc(&s->dct, lpcs);
-1.8 * tilt_factor(coeffs, remainder - 1),
coeffs, remainder);
}
- sq = (1.0 / 64.0) * sqrtf(1 / ff_scalarproduct_float_c(coeffs, coeffs, remainder));
+ sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
+ remainder));
for (n = 0; n < remainder; n++)
coeffs[n] *= sq;
}
* @param gb bit I/O context
* @param block_idx block index in frame [0, 1]
* @param fcb structure containing fixed codebook vector info
+ * @return -1 on error, 0 otherwise
*/
-static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
- int block_idx, AMRFixed *fcb)
+static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
+ int block_idx, AMRFixed *fcb)
{
uint16_t use_mask_mem[9]; // only 5 are used, rest is padding
uint16_t *use_mask = use_mask_mem + 2;
else if (use_mask[2]) idx = 0x2F;
else if (use_mask[3]) idx = 0x3F;
else if (use_mask[4]) idx = 0x4F;
- else return;
+ else return -1;
idx -= av_log2_16bit(use_mask[idx >> 4]);
}
if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
/* set offset for next block, relative to start of that block */
n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
+ return 0;
}
/**
* (fixed) codebook pulses of the speech signal. */
if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
aw_pulse_set1(s, gb, block_idx, &fcb);
- aw_pulse_set2(s, gb, block_idx, &fcb);
+ if (aw_pulse_set2(s, gb, block_idx, &fcb)) {
+ /* Conceal the block with silence and return.
+ * Skip the correct amount of bits to read the next
+ * block from the correct offset. */
+ int r_idx = pRNG(s->frame_cntr, block_idx, size);
+
+ for (n = 0; n < size; n++)
+ excitation[n] =
+ wmavoice_std_codebook[r_idx + n] * s->silence_gain;
+ skip_bits(gb, 7 + 1);
+ return;
+ }
} else /* FCB_TYPE_EXC_PULSES */ {
int offset_nbits = 5 - frame_desc->log_n_blocks;
/* Calculate gain for adaptive & fixed codebook signal.
* see ff_amr_set_fixed_gain(). */
idx = get_bits(gb, 7);
- fcb_gain = expf(ff_scalarproduct_float_c(s->gain_pred_err, gain_coeff, 6) -
+ fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err,
+ gain_coeff, 6) -
5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
acb_gain = wmavoice_gain_codebook_acb[idx];
pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
if (bd_idx < 0) {
av_log(ctx, AV_LOG_ERROR,
"Invalid frame type VLC code, skipping\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
* 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 -1 if unsupported, 1 on not enough bits or 0 if OK.
+ * @return < 0 on error, 1 on not enough bits or 0 if OK.
*/
static int check_bits_for_superframe(GetBitContext *orig_gb,
WMAVoiceContext *s)
if (get_bits_left(gb) < 14)
return 1;
if (!get_bits1(gb))
- return -1; // WMAPro-in-WMAVoice superframe
+ return AVERROR(ENOSYS); // WMAPro-in-WMAVoice superframe
if (get_bits1(gb)) skip_bits(gb, 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)
}
bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)];
if (bd_idx < 0)
- return -1; // invalid frame type VLC code
+ 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)
* @return 0 on success, <0 on error or 1 if there was not enough data to
* fully parse the superframe
*/
-static int synth_superframe(AVCodecContext *ctx, int *got_frame_ptr)
+static int synth_superframe(AVCodecContext *ctx, AVFrame *frame,
+ int *got_frame_ptr)
{
WMAVoiceContext *s = ctx->priv_data;
GetBitContext *gb = &s->gb, s_gb;
if ((res = check_bits_for_superframe(gb, s)) == 1) {
*got_frame_ptr = 0;
return 1;
- }
+ } else if (res < 0)
+ return res;
/* First bit is speech/music bit, it differentiates between WMAVoice
* 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)) {
- av_log_missing_feature(ctx, "WMAPro-in-WMAVoice", 1);
+ avpriv_request_sample(ctx, "WMAPro-in-WMAVoice");
return AVERROR_PATCHWELCOME;
}
av_log(ctx, AV_LOG_ERROR,
"Superframe encodes >480 samples (%d), not allowed\n",
n_samples);
- return -1;
+ return AVERROR_INVALIDDATA;
}
}
/* Parse LSPs, if global for the superframe (can also be per-frame). */
}
/* get output buffer */
- s->frame.nb_samples = 480;
- if ((res = ctx->get_buffer(ctx, &s->frame)) < 0) {
+ frame->nb_samples = 480;
+ if ((res = ff_get_buffer(ctx, frame, 0)) < 0) {
av_log(ctx, AV_LOG_ERROR, "get_buffer() failed\n");
return res;
}
- s->frame.nb_samples = n_samples;
- samples = (float *)s->frame.data[0];
+ frame->nb_samples = n_samples;
+ samples = (float *)frame->data[0];
/* Parse frames, optionally preceded by per-frame (independent) LSPs. */
for (n = 0; n < 3; n++) {
copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
flush_put_bits(&s->pb);
s->sframe_cache_size += s->spillover_nbits;
- if ((res = synth_superframe(ctx, got_frame_ptr)) == 0 &&
+ if ((res = synth_superframe(ctx, data, got_frame_ptr)) == 0 &&
*got_frame_ptr) {
cnt += s->spillover_nbits;
s->skip_bits_next = cnt & 7;
- *(AVFrame *)data = s->frame;
return cnt >> 3;
} else
skip_bits_long (gb, s->spillover_nbits - cnt +
s->sframe_cache_size = 0;
s->skip_bits_next = 0;
pos = get_bits_left(gb);
- if ((res = synth_superframe(ctx, got_frame_ptr)) < 0) {
+ if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) {
return res;
} else if (*got_frame_ptr) {
int cnt = get_bits_count(gb);
s->skip_bits_next = cnt & 7;
- *(AVFrame *)data = s->frame;
return cnt >> 3;
} else if ((s->sframe_cache_size = pos) > 0) {
/* rewind bit reader to start of last (incomplete) superframe... */
}
AVCodec ff_wmavoice_decoder = {
- .name = "wmavoice",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_WMAVOICE,
- .priv_data_size = sizeof(WMAVoiceContext),
- .init = wmavoice_decode_init,
- .close = wmavoice_decode_end,
- .decode = wmavoice_decode_packet,
- .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
- .flush = wmavoice_flush,
- .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
+ .name = "wmavoice",
+ .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_WMAVOICE,
+ .priv_data_size = sizeof(WMAVoiceContext),
+ .init = wmavoice_decode_init,
+ .init_static_data = wmavoice_init_static_data,
+ .close = wmavoice_decode_end,
+ .decode = wmavoice_decode_packet,
+ .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
+ .flush = wmavoice_flush,
};