*/
typedef struct {
/**
- * @defgroup struct_global Global values
- * Global values, specified in the stream header / extradata or used
- * all over.
+ * @name Global values specified in the stream header / extradata or used all over.
* @{
*/
GetBitContext gb; ///< packet bitreader. During decoder init,
/**
* @}
- * @defgroup struct_packet Packet values
- * Packet values, specified in the packet header or related to a packet.
+ *
+ * @name Packet values specified in the packet header or related to a packet.
+ *
* A packet is considered to be a single unit of data provided to this
* decoder by the demuxer.
* @{
/**
* @}
- * @defgroup struct_frame Frame and superframe values
+ *
+ * @name Frame and superframe values
* Superframe and frame data - these can change from frame to frame,
* although some of them do in that case serve as a cache / history for
* the next frame or superframe.
float synth_history[MAX_LSPS]; ///< see #excitation_history
/**
* @}
- * @defgroup post_filter Postfilter values
+ *
+ * @name Postfilter values
+ *
* Variables used for postfilter implementation, mostly history for
* smoothing and so on, and context variables for FFT/iFFT.
* @{
///< by postfilter
float denoise_filter_cache[MAX_FRAMESIZE];
int denoise_filter_cache_size; ///< samples in #denoise_filter_cache
- DECLARE_ALIGNED(16, float, tilted_lpcs_pf)[0x80];
+ DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80];
///< aligned buffer for LPC tilting
- DECLARE_ALIGNED(16, float, denoise_coeffs_pf)[0x80];
+ DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80];
///< aligned buffer for denoise coefficients
- DECLARE_ALIGNED(16, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
+ DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
///< aligned buffer for postfilter speech
///< synthesis
/**
};
int cntr[8], n, res;
- memset(vbm_tree, 0xff, sizeof(vbm_tree));
+ memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25);
memset(cntr, 0, sizeof(cntr));
for (n = 0; n < 17; n++) {
res = get_bits(gb, 3);
s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8;
s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 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;
+ }
s->pitch_nbits = av_ceil_log2(pitch_range);
s->last_pitch_val = 40;
s->last_acb_type = ACB_TYPE_NONE;
s->block_conv_table[2] = (pitch_range * 44) >> 6;
s->block_conv_table[3] = s->max_pitch_val - 1;
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;
+ }
s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
s->block_pitch_range = s->block_conv_table[2] +
s->block_conv_table[3] + 1 +
}
/**
- * @defgroup postfilter Postfilter functions
+ * @name Postfilter functions
* Postfilter functions (gain control, wiener denoise filter, DC filter,
* kalman smoothening, plus surrounding code to wrap it)
* @{
}
/**
- * @defgroup lsp_dequant LSP dequantization routines
+ * @name LSP dequantization routines
* LSP dequantization routines, for 10/16LSPs and independent/residual coding.
* @note we assume enough bits are available, caller should check.
* lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
/**
* @}
- * @defgroup aw Pitch-adaptive window coding functions
+ * @name Pitch-adaptive window coding functions
* The next few functions are for pitch-adaptive window coding.
* @{
*/
int excl_range = s->aw_pulse_range; // always 16 or 24
uint16_t *use_mask_ptr = &use_mask[idx >> 4];
int first_sh = 16 - (idx & 15);
- *use_mask_ptr++ &= 0xFFFF << first_sh;
+ *use_mask_ptr++ &= 0xFFFFu << first_sh;
excl_range -= first_sh;
if (excl_range >= 16) {
*use_mask_ptr++ = 0;
{
WMAVoiceContext *s = ctx->priv_data;
GetBitContext *gb = &s->gb, s_gb;
- int n, res, n_samples = 480;
+ int n, res, out_size, n_samples = 480;
double lsps[MAX_FRAMES][MAX_LSPS];
const double *mean_lsf = s->lsps == 16 ?
wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
s->sframe_cache_size = 0;
}
- if ((res = check_bits_for_superframe(gb, s)) == 1) return 1;
+ if ((res = check_bits_for_superframe(gb, s)) == 1) {
+ *data_size = 0;
+ return 1;
+ }
/* First bit is speech/music bit, it differentiates between WMAVoice
* speech samples (the actual codec) and WMAVoice music samples, which
stabilize_lsps(lsps[n], s->lsps);
}
+ out_size = n_samples * av_get_bytes_per_sample(ctx->sample_fmt);
+ if (*data_size < out_size) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Output buffer too small (%d given - %zu needed)\n",
+ *data_size, out_size);
+ return -1;
+ }
+
/* Parse frames, optionally preceeded by per-frame (independent) LSPs. */
for (n = 0; n < 3; n++) {
if (!s->has_residual_lsps) {
&samples[n * MAX_FRAMESIZE],
lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
&excitation[s->history_nsamples + n * MAX_FRAMESIZE],
- &synth[s->lsps + n * MAX_FRAMESIZE])))
+ &synth[s->lsps + n * MAX_FRAMESIZE]))) {
+ *data_size = 0;
return res;
+ }
}
/* Statistics? FIXME - we don't check for length, a slight overrun
}
/* Specify nr. of output samples */
- *data_size = n_samples * sizeof(float);
+ *data_size = out_size;
/* Update history */
memcpy(s->prev_lsps, lsps[2],
* @param size size of the source data, in bytes
* @param gb bit I/O context specifying the current position in the source.
* data. This function might use this to align the bit position to
- * a whole-byte boundary before calling #ff_copy_bits() on aligned
+ * a whole-byte boundary before calling #avpriv_copy_bits() on aligned
* source data
* @param nbits the amount of bits to copy from source to target
*
rmn_bits = rmn_bytes = get_bits_left(gb);
if (rmn_bits < nbits)
return;
+ if (nbits > pb->size_in_bits - put_bits_count(pb))
+ return;
rmn_bits &= 7; rmn_bytes >>= 3;
if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
- ff_copy_bits(pb, data + size - rmn_bytes,
+ avpriv_copy_bits(pb, data + size - rmn_bytes,
FFMIN(nbits - rmn_bits, rmn_bytes << 3));
}
GetBitContext *gb = &s->gb;
int size, res, pos;
- if (*data_size < 480 * sizeof(float)) {
- av_log(ctx, AV_LOG_ERROR,
- "Output buffer too small (%d given - %zu needed)\n",
- *data_size, 480 * sizeof(float));
- return -1;
- }
- *data_size = 0;
-
/* Packets are sometimes a multiple of ctx->block_align, with a packet
- * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
+ * header at each ctx->block_align bytes. However, Libav's ASF demuxer
* feeds us ASF packets, which may concatenate multiple "codec" packets
* in a single "muxer" packet, so we artificially emulate that by
* capping the packet size at ctx->block_align. */
for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
- if (!size)
+ if (!size) {
+ *data_size = 0;
return 0;
+ }
init_get_bits(&s->gb, avpkt->data, size << 3);
/* size == ctx->block_align is used to indicate whether we are dealing with
}
AVCodec ff_wmavoice_decoder = {
- "wmavoice",
- AVMEDIA_TYPE_AUDIO,
- CODEC_ID_WMAVOICE,
- sizeof(WMAVoiceContext),
- wmavoice_decode_init,
- NULL,
- wmavoice_decode_end,
- wmavoice_decode_packet,
- CODEC_CAP_SUBFRAMES,
+ .name = "wmavoice",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_WMAVOICE,
+ .priv_data_size = sizeof(WMAVoiceContext),
+ .init = wmavoice_decode_init,
+ .close = wmavoice_decode_end,
+ .decode = wmavoice_decode_packet,
+ .capabilities = CODEC_CAP_SUBFRAMES,
.flush = wmavoice_flush,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
};
projects / ffmpeg / blobdiff