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-rw-r--r--libavcodec/wmavoice.c22
1 files changed, 11 insertions, 11 deletions
diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c
index 4a7ba6dabc..0cf10da8c4 100644
--- a/libavcodec/wmavoice.c
+++ b/libavcodec/wmavoice.c
@@ -2,20 +2,20 @@
* Windows Media Audio Voice decoder.
* Copyright (c) 2009 Ronald S. Bultje
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg 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,
+ * FFmpeg 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
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -515,7 +515,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch,
float optimal_gain = 0, dot;
const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
*end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
- *best_hist_ptr;
+ *best_hist_ptr = NULL;
/* find best fitting point in history */
do {
@@ -774,7 +774,7 @@ static void postfilter(WMAVoiceContext *s, const float *synth,
*synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
*synth_filter_in = zero_exc_pf;
- assert(size <= MAX_FRAMESIZE / 2);
+ av_assert0(size <= MAX_FRAMESIZE / 2);
/* generate excitation from input signal */
ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
@@ -1241,7 +1241,7 @@ static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
float gain;
int n, r_idx;
- assert(size <= MAX_FRAMESIZE);
+ av_assert0(size <= MAX_FRAMESIZE);
/* Set the offset from which we start reading wmavoice_std_codebook */
if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
@@ -1277,7 +1277,7 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
int n, idx, gain_weight;
AMRFixed fcb;
- assert(size <= MAX_FRAMESIZE / 2);
+ av_assert0(size <= MAX_FRAMESIZE / 2);
memset(pulses, 0, sizeof(*pulses) * size);
fcb.pitch_lag = block_pitch_sh2 >> 2;
@@ -1654,7 +1654,7 @@ static int check_bits_for_superframe(GetBitContext *orig_gb,
/* initialize a copy */
init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
skip_bits_long(gb, get_bits_count(orig_gb));
- assert(get_bits_left(gb) == get_bits_left(orig_gb));
+ av_assert1(get_bits_left(gb) == get_bits_left(orig_gb));
/* superframe header */
if (get_bits_left(gb) < 14)
@@ -1933,7 +1933,7 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
int size, res, pos;
/* Packets are sometimes a multiple of ctx->block_align, with a packet
- * header at each ctx->block_align bytes. However, Libav's ASF demuxer
+ * header at each ctx->block_align bytes. However, FFmpeg'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. */
@@ -1990,7 +1990,7 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
/* rewind bit reader to start of last (incomplete) superframe... */
init_get_bits(gb, avpkt->data, size << 3);
skip_bits_long(gb, (size << 3) - pos);
- assert(get_bits_left(gb) == pos);
+ av_assert1(get_bits_left(gb) == pos);
/* ...and cache it for spillover in next packet */
init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);