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author | Michael Niedermayer <michaelni@gmx.at> | 2012-02-12 01:02:55 +0100 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2012-02-12 01:06:13 +0100 |
commit | cd1c12b5c5b79195140a93d59cbf990d034f61d8 (patch) | |
tree | 1a1ab570f0dddd706a1b995e255272c1c6f9f453 /libavcodec | |
parent | 289520fd97395ffd5bf933ac80487e858bc4039d (diff) | |
parent | b498867d6691b5f1f107afd81aff403f66b434aa (diff) | |
download | ffmpeg-cd1c12b5c5b79195140a93d59cbf990d034f61d8.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
FATE: update reference for seek-alac_mp4
sunrast: Return AVERROR values instead of -1.
sunrast: Add support for gray8 decoding.
swscale: enforce a minimum filtersize.
alacenc: use AVCodec.encode2()
alacenc: cosmetics: indentation
alacenc: consolidate bitstream writing into a single function.
alacenc: only encode frame size in header for a final smaller frame
alacenc: store current frame size in AlacEncodeContext.
alacenc: return AVERROR codes in alac_encode_frame()
alacenc: calculate a new max frame size for the final small frame
alacenc: pretty-printing and other cosmetics
alacenc: fix error handling and potential memleaks in alac_encode_init()
alacenc: do not set coded_frame->key_frame
alacenc: do not set bits_per_coded_sample
alacenc: remove unneeded frame_size check in alac_encode_frame()
tta: error out if samplerate is zero.
ttadec: fix invalid free when an error occurs while decoding 24-bit tta
wavpack: add needed braces for 2 statements inside an if block
Conflicts:
tests/ref/acodec/alac
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/alacenc.c | 351 | ||||
-rw-r--r-- | libavcodec/sunrast.c | 17 | ||||
-rw-r--r-- | libavcodec/tta.c | 26 | ||||
-rw-r--r-- | libavcodec/wavpack.c | 3 |
4 files changed, 220 insertions, 177 deletions
diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index fde3b53e5e..cc9560462b 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -22,6 +22,7 @@ #include "avcodec.h" #include "put_bits.h" #include "dsputil.h" +#include "internal.h" #include "lpc.h" #include "mathops.h" @@ -58,6 +59,8 @@ typedef struct AlacLPCContext { } AlacLPCContext; typedef struct AlacEncodeContext { + int frame_size; /**< current frame size */ + int verbatim; /**< current frame verbatim mode flag */ int compression_level; int min_prediction_order; int max_prediction_order; @@ -82,7 +85,7 @@ static void init_sample_buffers(AlacEncodeContext *s, for (ch = 0; ch < s->avctx->channels; ch++) { const int16_t *sptr = input_samples + ch; - for (i = 0; i < s->avctx->frame_size; i++) { + for (i = 0; i < s->frame_size; i++) { s->sample_buf[ch][i] = *sptr; sptr += s->avctx->channels; } @@ -117,14 +120,20 @@ static void encode_scalar(AlacEncodeContext *s, int x, } } -static void write_frame_header(AlacEncodeContext *s, int is_verbatim) +static void write_frame_header(AlacEncodeContext *s) { - put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1 - put_bits(&s->pbctx, 16, 0); // Seems to be zero - put_bits(&s->pbctx, 1, 1); // Sample count is in the header - put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field - put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim - put_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame + int encode_fs = 0; + + if (s->frame_size < DEFAULT_FRAME_SIZE) + encode_fs = 1; + + put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1 + put_bits(&s->pbctx, 16, 0); // Seems to be zero + put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header + put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field + put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim + if (encode_fs) + put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame } static void calc_predictor_params(AlacEncodeContext *s, int ch) @@ -144,7 +153,7 @@ static void calc_predictor_params(AlacEncodeContext *s, int ch) s->lpc[ch].lpc_coeff[5] = -25; } else { opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch], - s->avctx->frame_size, + s->frame_size, s->min_prediction_order, s->max_prediction_order, ALAC_MAX_LPC_PRECISION, coefs, shift, @@ -167,8 +176,8 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) /* calculate sum of 2nd order residual for each channel */ sum[0] = sum[1] = sum[2] = sum[3] = 0; for (i = 2; i < n; i++) { - lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; - rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; + lt = left_ch[i] - 2 * left_ch[i - 1] + left_ch[i - 2]; + rt = right_ch[i] - 2 * right_ch[i - 1] + right_ch[i - 2]; sum[2] += FFABS((lt + rt) >> 1); sum[3] += FFABS(lt - rt); sum[0] += FFABS(lt); @@ -184,9 +193,8 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) /* return mode with lowest score */ best = 0; for (i = 1; i < 4; i++) { - if (score[i] < score[best]) { + if (score[i] < score[best]) best = i; - } } return best; } @@ -194,45 +202,40 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) static void alac_stereo_decorrelation(AlacEncodeContext *s) { int32_t *left = s->sample_buf[0], *right = s->sample_buf[1]; - int i, mode, n = s->avctx->frame_size; + int i, mode, n = s->frame_size; int32_t tmp; mode = estimate_stereo_mode(left, right, n); - switch(mode) - { - case ALAC_CHMODE_LEFT_RIGHT: - s->interlacing_leftweight = 0; - s->interlacing_shift = 0; - break; - - case ALAC_CHMODE_LEFT_SIDE: - for (i = 0; i < n; i++) { - right[i] = left[i] - right[i]; - } - s->interlacing_leftweight = 1; - s->interlacing_shift = 0; - break; - - case ALAC_CHMODE_RIGHT_SIDE: - for (i = 0; i < n; i++) { - tmp = right[i]; - right[i] = left[i] - right[i]; - left[i] = tmp + (right[i] >> 31); - } - s->interlacing_leftweight = 1; - s->interlacing_shift = 31; - break; - - default: - for (i = 0; i < n; i++) { - tmp = left[i]; - left[i] = (tmp + right[i]) >> 1; - right[i] = tmp - right[i]; - } - s->interlacing_leftweight = 1; - s->interlacing_shift = 1; - break; + switch (mode) { + case ALAC_CHMODE_LEFT_RIGHT: + s->interlacing_leftweight = 0; + s->interlacing_shift = 0; + break; + case ALAC_CHMODE_LEFT_SIDE: + for (i = 0; i < n; i++) + right[i] = left[i] - right[i]; + s->interlacing_leftweight = 1; + s->interlacing_shift = 0; + break; + case ALAC_CHMODE_RIGHT_SIDE: + for (i = 0; i < n; i++) { + tmp = right[i]; + right[i] = left[i] - right[i]; + left[i] = tmp + (right[i] >> 31); + } + s->interlacing_leftweight = 1; + s->interlacing_shift = 31; + break; + default: + for (i = 0; i < n; i++) { + tmp = left[i]; + left[i] = (tmp + right[i]) >> 1; + right[i] = tmp - right[i]; + } + s->interlacing_leftweight = 1; + s->interlacing_shift = 1; + break; } } @@ -244,8 +247,10 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) if (lpc.lpc_order == 31) { s->predictor_buf[0] = s->sample_buf[ch][0]; - for (i = 1; i < s->avctx->frame_size; i++) - s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1]; + for (i = 1; i < s->frame_size; i++) { + s->predictor_buf[i] = s->sample_buf[ch][i ] - + s->sample_buf[ch][i - 1]; + } return; } @@ -262,12 +267,12 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) residual[i] = samples[i] - samples[i-1]; // perform lpc on remaining samples - for (i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) { + for (i = lpc.lpc_order + 1; i < s->frame_size; i++) { int sum = 1 << (lpc.lpc_quant - 1), res_val, j; for (j = 0; j < lpc.lpc_order; j++) { sum += (samples[lpc.lpc_order-j] - samples[0]) * - lpc.lpc_coeff[j]; + lpc.lpc_coeff[j]; } sum >>= lpc.lpc_quant; @@ -276,21 +281,20 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) s->write_sample_size); res_val = residual[i]; - if(res_val) { + if (res_val) { int index = lpc.lpc_order - 1; int neg = (res_val < 0); - while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) { - int val = samples[0] - samples[lpc.lpc_order - index]; + while (index >= 0 && (neg ? (res_val < 0) : (res_val > 0))) { + int val = samples[0] - samples[lpc.lpc_order - index]; int sign = (val ? FFSIGN(val) : 0); - if(neg) - sign*=-1; + if (neg) + sign *= -1; lpc.lpc_coeff[index] -= sign; val *= sign; - res_val -= ((val >> lpc.lpc_quant) * - (lpc.lpc_order - index)); + res_val -= (val >> lpc.lpc_quant) * (lpc.lpc_order - index); index--; } } @@ -305,95 +309,122 @@ static void alac_entropy_coder(AlacEncodeContext *s) int sign_modifier = 0, i, k; int32_t *samples = s->predictor_buf; - for (i = 0; i < s->avctx->frame_size;) { + for (i = 0; i < s->frame_size;) { int x; k = av_log2((history >> 9) + 3); - x = -2*(*samples)-1; - x ^= (x>>31); + x = -2 * (*samples) -1; + x ^= x >> 31; samples++; i++; encode_scalar(s, x - sign_modifier, k, s->write_sample_size); - history += x * s->rc.history_mult - - ((history * s->rc.history_mult) >> 9); + history += x * s->rc.history_mult - + ((history * s->rc.history_mult) >> 9); sign_modifier = 0; if (x > 0xFFFF) history = 0xFFFF; - if (history < 128 && i < s->avctx->frame_size) { + if (history < 128 && i < s->frame_size) { unsigned int block_size = 0; k = 7 - av_log2(history) + ((history + 16) >> 6); - while (*samples == 0 && i < s->avctx->frame_size) { + while (*samples == 0 && i < s->frame_size) { samples++; i++; block_size++; } encode_scalar(s, block_size, k, 16); - sign_modifier = (block_size <= 0xFFFF); - history = 0; } } } -static void write_compressed_frame(AlacEncodeContext *s) +static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, + const int16_t *samples) { int i, j; int prediction_type = 0; + PutBitContext *pb = &s->pbctx; - if (s->avctx->channels == 2) - alac_stereo_decorrelation(s); - put_bits(&s->pbctx, 8, s->interlacing_shift); - put_bits(&s->pbctx, 8, s->interlacing_leftweight); + init_put_bits(pb, avpkt->data, avpkt->size); + + if (s->verbatim) { + write_frame_header(s); + for (i = 0; i < s->frame_size * s->avctx->channels; i++) + put_sbits(pb, 16, *samples++); + } else { + init_sample_buffers(s, samples); + write_frame_header(s); - for (i = 0; i < s->avctx->channels; i++) { + if (s->avctx->channels == 2) + alac_stereo_decorrelation(s); + put_bits(pb, 8, s->interlacing_shift); + put_bits(pb, 8, s->interlacing_leftweight); - calc_predictor_params(s, i); + for (i = 0; i < s->avctx->channels; i++) { + calc_predictor_params(s, i); - put_bits(&s->pbctx, 4, prediction_type); - put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant); + put_bits(pb, 4, prediction_type); + put_bits(pb, 4, s->lpc[i].lpc_quant); - put_bits(&s->pbctx, 3, s->rc.rice_modifier); - put_bits(&s->pbctx, 5, s->lpc[i].lpc_order); - // predictor coeff. table - for (j = 0; j < s->lpc[i].lpc_order; j++) { - put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); + put_bits(pb, 3, s->rc.rice_modifier); + put_bits(pb, 5, s->lpc[i].lpc_order); + // predictor coeff. table + for (j = 0; j < s->lpc[i].lpc_order; j++) + put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]); } - } - // apply lpc and entropy coding to audio samples + // apply lpc and entropy coding to audio samples - for (i = 0; i < s->avctx->channels; i++) { - alac_linear_predictor(s, i); + for (i = 0; i < s->avctx->channels; i++) { + alac_linear_predictor(s, i); - // TODO: determine when this will actually help. for now it's not used. - if (prediction_type == 15) { - // 2nd pass 1st order filter - for (j = s->avctx->frame_size - 1; j > 0; j--) - s->predictor_buf[j] -= s->predictor_buf[j - 1]; - } + // TODO: determine when this will actually help. for now it's not used. + if (prediction_type == 15) { + // 2nd pass 1st order filter + for (j = s->frame_size - 1; j > 0; j--) + s->predictor_buf[j] -= s->predictor_buf[j - 1]; + } - alac_entropy_coder(s); + alac_entropy_coder(s); + } } + put_bits(pb, 3, 7); + flush_put_bits(pb); + return put_bits_count(pb) >> 3; +} + +static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps) +{ + int header_bits = 23 + 32 * (frame_size < DEFAULT_FRAME_SIZE); + return FFALIGN(header_bits + bps * ch * frame_size + 3, 8) / 8; +} + +static av_cold int alac_encode_close(AVCodecContext *avctx) +{ + AlacEncodeContext *s = avctx->priv_data; + ff_lpc_end(&s->lpc_ctx); + av_freep(&avctx->extradata); + avctx->extradata_size = 0; + av_freep(&avctx->coded_frame); + return 0; } static av_cold int alac_encode_init(AVCodecContext *avctx) { - AlacEncodeContext *s = avctx->priv_data; + AlacEncodeContext *s = avctx->priv_data; int ret; - uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1); + uint8_t *alac_extradata; - avctx->frame_size = DEFAULT_FRAME_SIZE; - avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE; + avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE; if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) { av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n"); @@ -420,18 +451,29 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) s->rc.k_modifier = 14; s->rc.rice_modifier = 4; - s->max_coded_frame_size = 8 + (avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample>>3); + s->max_coded_frame_size = get_max_frame_size(avctx->frame_size, + avctx->channels, + DEFAULT_SAMPLE_SIZE); - s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes + // FIXME: consider wasted_bytes + s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1; + avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE); + if (!avctx->extradata) { + ret = AVERROR(ENOMEM); + goto error; + } + avctx->extradata_size = ALAC_EXTRADATA_SIZE; + + alac_extradata = avctx->extradata; AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE); AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c')); AV_WB32(alac_extradata+12, avctx->frame_size); - AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample); + AV_WB8 (alac_extradata+17, DEFAULT_SAMPLE_SIZE); AV_WB8 (alac_extradata+21, avctx->channels); AV_WB32(alac_extradata+24, s->max_coded_frame_size); AV_WB32(alac_extradata+28, - avctx->sample_rate * avctx->channels * avctx->bits_per_coded_sample); // average bitrate + avctx->sample_rate * avctx->channels * DEFAULT_SAMPLE_SIZE); // average bitrate AV_WB32(alac_extradata+32, avctx->sample_rate); // Set relevant extradata fields @@ -447,7 +489,8 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) { av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order); - return -1; + ret = AVERROR(EINVAL); + goto error; } s->min_prediction_order = avctx->min_prediction_order; @@ -459,7 +502,8 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) { av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order); - return -1; + ret = AVERROR(EINVAL); + goto error; } s->max_prediction_order = avctx->max_prediction_order; @@ -469,80 +513,63 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n", s->min_prediction_order, s->max_prediction_order); - return -1; + ret = AVERROR(EINVAL); + goto error; } - avctx->extradata = alac_extradata; - avctx->extradata_size = ALAC_EXTRADATA_SIZE; - avctx->coded_frame = avcodec_alloc_frame(); - avctx->coded_frame->key_frame = 1; + if (!avctx->coded_frame) { + ret = AVERROR(ENOMEM); + goto error; + } s->avctx = avctx; - ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size, s->max_prediction_order, - FF_LPC_TYPE_LEVINSON); + if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size, + s->max_prediction_order, + FF_LPC_TYPE_LEVINSON)) < 0) { + goto error; + } + + return 0; +error: + alac_encode_close(avctx); return ret; } -static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, - int buf_size, void *data) +static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) { AlacEncodeContext *s = avctx->priv_data; - PutBitContext *pb = &s->pbctx; - int i, out_bytes, verbatim_flag = 0; - - if (avctx->frame_size > DEFAULT_FRAME_SIZE) { - av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n"); - return -1; - } + int out_bytes, max_frame_size, ret; + const int16_t *samples = (const int16_t *)frame->data[0]; - if (buf_size < 2 * s->max_coded_frame_size) { - av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); - return -1; - } + s->frame_size = frame->nb_samples; -verbatim: - init_put_bits(pb, frame, buf_size); + if (avctx->frame_size < DEFAULT_FRAME_SIZE) + max_frame_size = get_max_frame_size(s->frame_size, avctx->channels, + DEFAULT_SAMPLE_SIZE); + else + max_frame_size = s->max_coded_frame_size; - if (s->compression_level == 0 || verbatim_flag) { - // Verbatim mode - const int16_t *samples = data; - write_frame_header(s, 1); - for (i = 0; i < avctx->frame_size * avctx->channels; i++) { - put_sbits(pb, 16, *samples++); - } - } else { - init_sample_buffers(s, data); - write_frame_header(s, 0); - write_compressed_frame(s); + if ((ret = ff_alloc_packet(avpkt, 2 * max_frame_size))) { + av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + return ret; } - put_bits(pb, 3, 7); - flush_put_bits(pb); - out_bytes = put_bits_count(pb) >> 3; + /* use verbatim mode for compression_level 0 */ + s->verbatim = !s->compression_level; + + out_bytes = write_frame(s, avpkt, samples); - if (out_bytes > s->max_coded_frame_size) { + if (out_bytes > max_frame_size) { /* frame too large. use verbatim mode */ - if (verbatim_flag || s->compression_level == 0) { - /* still too large. must be an error. */ - av_log(avctx, AV_LOG_ERROR, "error encoding frame\n"); - return -1; - } - verbatim_flag = 1; - goto verbatim; + s->verbatim = 1; + out_bytes = write_frame(s, avpkt, samples); } - return out_bytes; -} - -static av_cold int alac_encode_close(AVCodecContext *avctx) -{ - AlacEncodeContext *s = avctx->priv_data; - ff_lpc_end(&s->lpc_ctx); - av_freep(&avctx->extradata); - avctx->extradata_size = 0; - av_freep(&avctx->coded_frame); + avpkt->size = out_bytes; + *got_packet_ptr = 1; return 0; } @@ -552,10 +579,10 @@ AVCodec ff_alac_encoder = { .id = CODEC_ID_ALAC, .priv_data_size = sizeof(AlacEncodeContext), .init = alac_encode_init, - .encode = alac_encode_frame, + .encode2 = alac_encode_frame, .close = alac_encode_close, - .capabilities = CODEC_CAP_SMALL_LAST_FRAME, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, - AV_SAMPLE_FMT_NONE }, - .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), + .capabilities = CODEC_CAP_SMALL_LAST_FRAME, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, + .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), }; diff --git a/libavcodec/sunrast.c b/libavcodec/sunrast.c index 8562e11ac4..aab6435cdd 100644 --- a/libavcodec/sunrast.c +++ b/libavcodec/sunrast.c @@ -72,13 +72,14 @@ static int sunrast_decode_frame(AVCodecContext *avctx, void *data, unsigned int w, h, depth, type, maptype, maplength, stride, x, y, len, alen; uint8_t *ptr, *ptr2 = NULL; const uint8_t *bufstart = buf; + int ret; if (avpkt->size < 32) return AVERROR_INVALIDDATA; if (AV_RB32(buf) != RAS_MAGIC) { av_log(avctx, AV_LOG_ERROR, "this is not sunras encoded data\n"); - return -1; + return AVERROR_INVALIDDATA; } w = AV_RB32(buf + 4); @@ -95,15 +96,15 @@ static int sunrast_decode_frame(AVCodecContext *avctx, void *data, } if (type > RT_FORMAT_IFF) { av_log(avctx, AV_LOG_ERROR, "invalid (compression) type\n"); - return -1; + return AVERROR_INVALIDDATA; } if (av_image_check_size(w, h, 0, avctx)) { av_log(avctx, AV_LOG_ERROR, "invalid image size\n"); - return -1; + return AVERROR_INVALIDDATA; } if (maptype & ~1) { av_log(avctx, AV_LOG_ERROR, "invalid colormap type\n"); - return -1; + return AVERROR_INVALIDDATA; } if (type == RT_FORMAT_TIFF || type == RT_FORMAT_IFF) { @@ -129,7 +130,7 @@ static int sunrast_decode_frame(AVCodecContext *avctx, void *data, break; default: av_log(avctx, AV_LOG_ERROR, "invalid depth\n"); - return -1; + return AVERROR_INVALIDDATA; } if (p->data[0]) @@ -137,9 +138,9 @@ static int sunrast_decode_frame(AVCodecContext *avctx, void *data, if (w != avctx->width || h != avctx->height) avcodec_set_dimensions(avctx, w, h); - if (avctx->get_buffer(avctx, p) < 0) { + if ((ret = avctx->get_buffer(avctx, p)) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); - return -1; + return ret; } p->pict_type = AV_PICTURE_TYPE_I; @@ -155,7 +156,7 @@ static int sunrast_decode_frame(AVCodecContext *avctx, void *data, if (maplength % 3 || maplength > 768) { av_log(avctx, AV_LOG_WARNING, "invalid colormap length\n"); - return -1; + return AVERROR_INVALIDDATA; } ptr = p->data[1]; diff --git a/libavcodec/tta.c b/libavcodec/tta.c index 8a628a53a3..3f5ab23b1e 100644 --- a/libavcodec/tta.c +++ b/libavcodec/tta.c @@ -235,6 +235,9 @@ static av_cold int tta_decode_init(AVCodecContext * avctx) if (s->channels == 0) { av_log(s->avctx, AV_LOG_ERROR, "Invalid number of channels\n"); return AVERROR_INVALIDDATA; + } else if (avctx->sample_rate == 0) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid samplerate\n"); + return AVERROR_INVALIDDATA; } switch(s->bps) { @@ -354,12 +357,16 @@ static int tta_decode_frame(AVCodecContext *avctx, void *data, unary--; } - if (get_bits_left(&s->gb) < k) - return -1; + if (get_bits_left(&s->gb) < k) { + ret = AVERROR_INVALIDDATA; + goto error; + } if (k) { - if (k > MIN_CACHE_BITS) - return -1; + if (k > MIN_CACHE_BITS) { + ret = AVERROR_INVALIDDATA; + goto error; + } value = (unary << k) + get_bits(&s->gb, k); } else value = unary; @@ -412,8 +419,10 @@ static int tta_decode_frame(AVCodecContext *avctx, void *data, } } - if (get_bits_left(&s->gb) < 32) - return -1; + if (get_bits_left(&s->gb) < 32) { + ret = AVERROR_INVALIDDATA; + goto error; + } skip_bits_long(&s->gb, 32); // frame crc // convert to output buffer @@ -445,6 +454,11 @@ static int tta_decode_frame(AVCodecContext *avctx, void *data, *(AVFrame *)data = s->frame; return buf_size; +error: + // reset decode buffer + if (s->bps == 3) + s->decode_buffer = NULL; + return ret; } static av_cold int tta_decode_close(AVCodecContext *avctx) { diff --git a/libavcodec/wavpack.c b/libavcodec/wavpack.c index c24de459f4..e6dbab23e9 100644 --- a/libavcodec/wavpack.c +++ b/libavcodec/wavpack.c @@ -909,8 +909,9 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no, } else { for (j = 0; j < s->decorr[i].value; j++) { s->decorr[i].samplesA[j] = wp_exp2(AV_RL16(buf)); buf += 2; - if (s->stereo_in) + if (s->stereo_in) { s->decorr[i].samplesB[j] = wp_exp2(AV_RL16(buf)); buf += 2; + } } t += s->decorr[i].value * 2 * (s->stereo_in + 1); } |