From 9d7cee50aa349563aa5faca1cff256ffccff6551 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Fri, 10 Feb 2012 20:18:10 -0500 Subject: [PATCH 01/19] wavpack: add needed braces for 2 statements inside an if block --- libavcodec/wavpack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/libavcodec/wavpack.c b/libavcodec/wavpack.c index 31604f51fc..fb0e0b2fa7 100644 --- a/libavcodec/wavpack.c +++ b/libavcodec/wavpack.c @@ -911,8 +911,9 @@ 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); } From 6ab681a4c1ffc0d5c36ebf13a10e0ecc61c81429 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Thu, 9 Feb 2012 14:49:59 -0500 Subject: [PATCH 02/19] ttadec: fix invalid free when an error occurs while decoding 24-bit tta --- libavcodec/tta.c | 23 +++++++++++++++++------ 1 file changed, 17 insertions(+), 6 deletions(-) diff --git a/libavcodec/tta.c b/libavcodec/tta.c index 49d59538d3..853f6a2aae 100644 --- a/libavcodec/tta.c +++ b/libavcodec/tta.c @@ -339,12 +339,16 @@ 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; @@ -397,8 +401,10 @@ 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 @@ -419,6 +425,11 @@ 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) { From 7416d610362807848236ceff1bc6740dbc82842d Mon Sep 17 00:00:00 2001 From: "Ronald S. Bultje" Date: Fri, 10 Feb 2012 10:51:43 -0800 Subject: [PATCH 03/19] tta: error out if samplerate is zero. Prevents a division by zero later on. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org --- libavcodec/tta.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/libavcodec/tta.c b/libavcodec/tta.c index 853f6a2aae..28d0e9b7e9 100644 --- a/libavcodec/tta.c +++ b/libavcodec/tta.c @@ -223,6 +223,9 @@ 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) { From 43a4cb070bf7588c53fd192e8fbc71a52fa14a4c Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Wed, 1 Feb 2012 20:38:06 -0500 Subject: [PATCH 04/19] alacenc: remove unneeded frame_size check in alac_encode_frame() --- libavcodec/alacenc.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 19708eb431..8130542121 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -492,11 +492,6 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, 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; - } - if (buf_size < 2 * s->max_coded_frame_size) { av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); return -1; From 64fe3eaeb351582787cbef75a2fe160253663363 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Wed, 1 Feb 2012 20:41:40 -0500 Subject: [PATCH 05/19] alacenc: do not set bits_per_coded_sample encoded ALAC does not have a fixed number of bits per sample --- libavcodec/alacenc.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 8130542121..9725be8185 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -393,7 +393,6 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1); avctx->frame_size = DEFAULT_FRAME_SIZE; - avctx->bits_per_coded_sample = DEFAULT_SAMPLE_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 +419,18 @@ 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 = 8 + (avctx->frame_size * avctx->channels * DEFAULT_SAMPLE_SIZE >> 3); - s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes + s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1; // FIXME: consider wasted_bytes 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 From 6e63228323ef9b492156523fb2329ee8440d6b9a Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Wed, 1 Feb 2012 20:43:43 -0500 Subject: [PATCH 06/19] alacenc: do not set coded_frame->key_frame It is already set in avcodec_alloc_frame() --- libavcodec/alacenc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 9725be8185..7fb0de3721 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -475,7 +475,6 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) avctx->extradata_size = ALAC_EXTRADATA_SIZE; avctx->coded_frame = avcodec_alloc_frame(); - avctx->coded_frame->key_frame = 1; s->avctx = avctx; ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size, s->max_prediction_order, From 51c24838625ab58341bee0e45e3d168d6f4a98fe Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Wed, 1 Feb 2012 20:54:34 -0500 Subject: [PATCH 07/19] alacenc: fix error handling and potential memleaks in alac_encode_init() --- libavcodec/alacenc.c | 57 +++++++++++++++++++++++++++++--------------- 1 file changed, 38 insertions(+), 19 deletions(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 7fb0de3721..934541e39f 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -386,11 +386,21 @@ static void write_compressed_frame(AlacEncodeContext *s) } } +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; int ret; - uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1); + uint8_t *alac_extradata; avctx->frame_size = DEFAULT_FRAME_SIZE; @@ -423,6 +433,14 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1; // FIXME: consider wasted_bytes + 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); @@ -446,7 +464,8 @@ 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; @@ -458,7 +477,8 @@ 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; @@ -468,18 +488,27 @@ 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(); + 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; } @@ -529,16 +558,6 @@ verbatim: 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); - return 0; -} - AVCodec ff_alac_encoder = { .name = "alac", .type = AVMEDIA_TYPE_AUDIO, From fc9cf0b2a6a0bd3933fcef216860c594b767834e Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Wed, 1 Feb 2012 21:21:24 -0500 Subject: [PATCH 08/19] alacenc: pretty-printing and other cosmetics --- libavcodec/alacenc.c | 135 ++++++++++++++++++++----------------------- 1 file changed, 64 insertions(+), 71 deletions(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 934541e39f..9e632aeca1 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -119,12 +119,12 @@ static void encode_scalar(AlacEncodeContext *s, int x, static void write_frame_header(AlacEncodeContext *s, int is_verbatim) { - 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 + 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 } static void calc_predictor_params(AlacEncodeContext *s, int ch) @@ -167,8 +167,8 @@ 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 +184,8 @@ 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; } @@ -199,40 +198,35 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s) 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 +238,10 @@ 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->avctx->frame_size; i++) { + s->predictor_buf[i] = s->sample_buf[ch][i ] - + s->sample_buf[ch][i - 1]; + } return; } @@ -267,7 +263,7 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) 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 +272,20 @@ 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--; } } @@ -310,16 +305,16 @@ static void alac_entropy_coder(AlacEncodeContext *s) 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) @@ -336,9 +331,7 @@ static void alac_entropy_coder(AlacEncodeContext *s) block_size++; } encode_scalar(s, block_size, k, 16); - sign_modifier = (block_size <= 0xFFFF); - history = 0; } @@ -356,7 +349,6 @@ static void write_compressed_frame(AlacEncodeContext *s) put_bits(&s->pbctx, 8, s->interlacing_leftweight); for (i = 0; i < s->avctx->channels; i++) { - calc_predictor_params(s, i); put_bits(&s->pbctx, 4, prediction_type); @@ -365,9 +357,8 @@ static void write_compressed_frame(AlacEncodeContext *s) 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++) { + for (j = 0; j < s->lpc[i].lpc_order; j++) put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); - } } // apply lpc and entropy coding to audio samples @@ -398,11 +389,11 @@ static av_cold int alac_encode_close(AVCodecContext *avctx) 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; - avctx->frame_size = DEFAULT_FRAME_SIZE; + avctx->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"); @@ -429,9 +420,11 @@ 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 * DEFAULT_SAMPLE_SIZE >> 3); + s->max_coded_frame_size = 8 + (avctx->frame_size * avctx->channels * + DEFAULT_SAMPLE_SIZE >> 3); - s->write_sample_size = DEFAULT_SAMPLE_SIZE + 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) { @@ -566,8 +559,8 @@ AVCodec ff_alac_encoder = { .init = alac_encode_init, .encode = 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)"), }; From 302daf580027c7151668395ff3ec706c92d2ccc6 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Thu, 2 Feb 2012 16:03:41 -0500 Subject: [PATCH 09/19] alacenc: calculate a new max frame size for the final small frame Gives a better estimate of buffer requirements and a better decision of whether or not to use verbatim mode. --- libavcodec/alacenc.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 9e632aeca1..bd8ab35e37 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -377,6 +377,11 @@ static void write_compressed_frame(AlacEncodeContext *s) } } +static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps) +{ + return FFALIGN(55 + bps * ch * frame_size + 3, 8) / 8; +} + static av_cold int alac_encode_close(AVCodecContext *avctx) { AlacEncodeContext *s = avctx->priv_data; @@ -420,8 +425,9 @@ 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 * - DEFAULT_SAMPLE_SIZE >> 3); + s->max_coded_frame_size = get_max_frame_size(avctx->frame_size, + avctx->channels, + DEFAULT_SAMPLE_SIZE); // FIXME: consider wasted_bytes s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1; @@ -511,8 +517,15 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, AlacEncodeContext *s = avctx->priv_data; PutBitContext *pb = &s->pbctx; int i, out_bytes, verbatim_flag = 0; + int max_frame_size; - if (buf_size < 2 * s->max_coded_frame_size) { + if (avctx->frame_size < DEFAULT_FRAME_SIZE) + max_frame_size = get_max_frame_size(avctx->frame_size, avctx->channels, + DEFAULT_SAMPLE_SIZE); + else + max_frame_size = s->max_coded_frame_size; + + if (buf_size < 2 * max_frame_size) { av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); return -1; } @@ -537,7 +550,7 @@ verbatim: flush_put_bits(pb); out_bytes = put_bits_count(pb) >> 3; - 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. */ From 65d15aec77254ef46c8972c50ce4b4a12e0c4de9 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Thu, 2 Feb 2012 16:11:17 -0500 Subject: [PATCH 10/19] alacenc: return AVERROR codes in alac_encode_frame() --- libavcodec/alacenc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index bd8ab35e37..7bc5a19491 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -527,7 +527,7 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, if (buf_size < 2 * max_frame_size) { av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); - return -1; + return AVERROR(EINVAL); } verbatim: @@ -555,7 +555,7 @@ verbatim: 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; + return AVERROR_BUG; } verbatim_flag = 1; goto verbatim; From ba821b098b5748e46db0fea875679365b33110e3 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Thu, 2 Feb 2012 18:06:28 -0500 Subject: [PATCH 11/19] alacenc: store current frame size in AlacEncodeContext. This avoids an indirection and will simplify implementation of encode2() --- libavcodec/alacenc.c | 29 ++++++++++++++++------------- 1 file changed, 16 insertions(+), 13 deletions(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 7bc5a19491..88b2f82669 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -58,6 +58,7 @@ typedef struct AlacLPCContext { } AlacLPCContext; typedef struct AlacEncodeContext { + int frame_size; /**< current frame size */ int compression_level; int min_prediction_order; int max_prediction_order; @@ -82,7 +83,7 @@ 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; } @@ -124,7 +125,7 @@ static void write_frame_header(AlacEncodeContext *s, int is_verbatim) 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 + put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame } static void calc_predictor_params(AlacEncodeContext *s, int ch) @@ -144,7 +145,7 @@ 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, @@ -193,7 +194,7 @@ 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); @@ -238,7 +239,7 @@ 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++) { + for (i = 1; i < s->frame_size; i++) { s->predictor_buf[i] = s->sample_buf[ch][i ] - s->sample_buf[ch][i - 1]; } @@ -258,7 +259,7 @@ 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++) { @@ -300,7 +301,7 @@ 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); @@ -320,12 +321,12 @@ static void alac_entropy_coder(AlacEncodeContext *s) 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++; @@ -369,7 +370,7 @@ static void write_compressed_frame(AlacEncodeContext *s) // 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--) + for (j = s->frame_size - 1; j > 0; j--) s->predictor_buf[j] -= s->predictor_buf[j - 1]; } @@ -398,7 +399,7 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) int ret; uint8_t *alac_extradata; - avctx->frame_size = DEFAULT_FRAME_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"); @@ -519,8 +520,10 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, int i, out_bytes, verbatim_flag = 0; int max_frame_size; + s->frame_size = avctx->frame_size; + if (avctx->frame_size < DEFAULT_FRAME_SIZE) - max_frame_size = get_max_frame_size(avctx->frame_size, avctx->channels, + max_frame_size = get_max_frame_size(s->frame_size, avctx->channels, DEFAULT_SAMPLE_SIZE); else max_frame_size = s->max_coded_frame_size; @@ -537,7 +540,7 @@ verbatim: // Verbatim mode const int16_t *samples = data; write_frame_header(s, 1); - for (i = 0; i < avctx->frame_size * avctx->channels; i++) { + for (i = 0; i < s->frame_size * avctx->channels; i++) { put_sbits(pb, 16, *samples++); } } else { From b590f3a7bf9103ac7a7a61c48568676201d6824b Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Fri, 3 Feb 2012 17:27:31 -0500 Subject: [PATCH 12/19] alacenc: only encode frame size in header for a final smaller frame Otherwise it is not needed because it matches the frame size as encoded in the extradata. --- libavcodec/alacenc.c | 13 ++++++++++--- tests/ref/acodec/alac | 4 ++-- 2 files changed, 12 insertions(+), 5 deletions(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 88b2f82669..d2a24b145c 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -120,12 +120,18 @@ static void encode_scalar(AlacEncodeContext *s, int x, static void write_frame_header(AlacEncodeContext *s, int is_verbatim) { + 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, 1); // Sample count is in the header + 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, is_verbatim); // Audio block is verbatim - put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame + 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) @@ -380,7 +386,8 @@ static void write_compressed_frame(AlacEncodeContext *s) static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps) { - return FFALIGN(55 + bps * ch * frame_size + 3, 8) / 8; + 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) diff --git a/tests/ref/acodec/alac b/tests/ref/acodec/alac index cef12d0744..02752cf31d 100644 --- a/tests/ref/acodec/alac +++ b/tests/ref/acodec/alac @@ -1,4 +1,4 @@ -db1806d9ffd85c168c2c71a28e6d9229 *./tests/data/acodec/alac.m4a -389410 ./tests/data/acodec/alac.m4a +b9e78aa8b8774a63d187380a47201a37 *./tests/data/acodec/alac.m4a +389154 ./tests/data/acodec/alac.m4a 64151e4bcc2b717aa5a8454d424d6a1f *./tests/data/alac.acodec.out.wav stddev: 0.00 PSNR:999.99 MAXDIFF: 0 bytes: 1058400/ 1058400 From b6e8ff72ea055f40ee272a97bde3ff21b3ea6c27 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Fri, 3 Feb 2012 17:53:41 -0500 Subject: [PATCH 13/19] alacenc: consolidate bitstream writing into a single function. Simplifies use of verbatim mode. --- libavcodec/alacenc.c | 71 ++++++++++++++++++++------------------------ 1 file changed, 33 insertions(+), 38 deletions(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index d2a24b145c..332356db0b 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -59,6 +59,7 @@ typedef struct 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; @@ -118,7 +119,7 @@ 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) { int encode_fs = 0; @@ -129,7 +130,7 @@ static void write_frame_header(AlacEncodeContext *s, int is_verbatim) 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, is_verbatim); // Audio block is verbatim + 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 } @@ -345,27 +346,39 @@ static void alac_entropy_coder(AlacEncodeContext *s) } } -static void write_compressed_frame(AlacEncodeContext *s) +static int write_frame(AlacEncodeContext *s, uint8_t *data, int size, + const int16_t *samples) { int i, j; int prediction_type = 0; + PutBitContext *pb = &s->pbctx; + + init_put_bits(pb, data, 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); 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); + put_bits(pb, 8, s->interlacing_shift); + put_bits(pb, 8, s->interlacing_leftweight); 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); + 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(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); + put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]); } // apply lpc and entropy coding to audio samples @@ -382,6 +395,10 @@ static void write_compressed_frame(AlacEncodeContext *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) @@ -523,9 +540,7 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size, void *data) { AlacEncodeContext *s = avctx->priv_data; - PutBitContext *pb = &s->pbctx; - int i, out_bytes, verbatim_flag = 0; - int max_frame_size; + int out_bytes, max_frame_size; s->frame_size = avctx->frame_size; @@ -540,35 +555,15 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, return AVERROR(EINVAL); } -verbatim: - init_put_bits(pb, frame, buf_size); + /* use verbatim mode for compression_level 0 */ + s->verbatim = !s->compression_level; - if (s->compression_level == 0 || verbatim_flag) { - // Verbatim mode - const int16_t *samples = data; - write_frame_header(s, 1); - for (i = 0; i < s->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); - } - - put_bits(pb, 3, 7); - flush_put_bits(pb); - out_bytes = put_bits_count(pb) >> 3; + out_bytes = write_frame(s, frame, buf_size, data); 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 AVERROR_BUG; - } - verbatim_flag = 1; - goto verbatim; + s->verbatim = 1; + out_bytes = write_frame(s, frame, buf_size, data); } return out_bytes; From bee80054f75516cc63a1e0d50c00622a6a309832 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Fri, 3 Feb 2012 17:56:49 -0500 Subject: [PATCH 14/19] alacenc: cosmetics: indentation --- libavcodec/alacenc.c | 58 ++++++++++++++++++++++---------------------- 1 file changed, 29 insertions(+), 29 deletions(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 332356db0b..13ef291285 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -360,41 +360,41 @@ static int write_frame(AlacEncodeContext *s, uint8_t *data, int size, 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); + init_sample_buffers(s, samples); + write_frame_header(s); - if (s->avctx->channels == 2) - alac_stereo_decorrelation(s); - put_bits(pb, 8, s->interlacing_shift); - put_bits(pb, 8, s->interlacing_leftweight); + if (s->avctx->channels == 2) + alac_stereo_decorrelation(s); + put_bits(pb, 8, s->interlacing_shift); + put_bits(pb, 8, s->interlacing_leftweight); - for (i = 0; i < s->avctx->channels; i++) { - calc_predictor_params(s, i); + for (i = 0; i < s->avctx->channels; i++) { + calc_predictor_params(s, i); - put_bits(pb, 4, prediction_type); - put_bits(pb, 4, s->lpc[i].lpc_quant); + put_bits(pb, 4, prediction_type); + put_bits(pb, 4, s->lpc[i].lpc_quant); - 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 - - 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->frame_size - 1; j > 0; j--) - s->predictor_buf[j] -= s->predictor_buf[j - 1]; + 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]); } - alac_entropy_coder(s); - } + // apply lpc and entropy coding to audio samples + + 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->frame_size - 1; j > 0; j--) + s->predictor_buf[j] -= s->predictor_buf[j - 1]; + } + + alac_entropy_coder(s); + } } put_bits(pb, 3, 7); flush_put_bits(pb); From 764852d6531016491a19d73d31e619887890f04d Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Fri, 3 Feb 2012 18:04:58 -0500 Subject: [PATCH 15/19] alacenc: use AVCodec.encode2() --- libavcodec/alacenc.c | 30 +++++++++++++++++------------- 1 file changed, 17 insertions(+), 13 deletions(-) diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 13ef291285..405b85724e 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" @@ -346,14 +347,14 @@ static void alac_entropy_coder(AlacEncodeContext *s) } } -static int write_frame(AlacEncodeContext *s, uint8_t *data, int size, +static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, const int16_t *samples) { int i, j; int prediction_type = 0; PutBitContext *pb = &s->pbctx; - init_put_bits(pb, data, size); + init_put_bits(pb, avpkt->data, avpkt->size); if (s->verbatim) { write_frame_header(s); @@ -536,13 +537,14 @@ error: 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; - int out_bytes, max_frame_size; + int out_bytes, max_frame_size, ret; + const int16_t *samples = (const int16_t *)frame->data[0]; - s->frame_size = avctx->frame_size; + s->frame_size = frame->nb_samples; if (avctx->frame_size < DEFAULT_FRAME_SIZE) max_frame_size = get_max_frame_size(s->frame_size, avctx->channels, @@ -550,23 +552,25 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, else max_frame_size = s->max_coded_frame_size; - if (buf_size < 2 * max_frame_size) { - av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); - return AVERROR(EINVAL); + if ((ret = ff_alloc_packet(avpkt, 2 * max_frame_size))) { + av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + return ret; } /* use verbatim mode for compression_level 0 */ s->verbatim = !s->compression_level; - out_bytes = write_frame(s, frame, buf_size, data); + out_bytes = write_frame(s, avpkt, samples); if (out_bytes > max_frame_size) { /* frame too large. use verbatim mode */ s->verbatim = 1; - out_bytes = write_frame(s, frame, buf_size, data); + out_bytes = write_frame(s, avpkt, samples); } - return out_bytes; + avpkt->size = out_bytes; + *got_packet_ptr = 1; + return 0; } AVCodec ff_alac_encoder = { @@ -575,7 +579,7 @@ 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, From dae2ce361a2b5fd9be1d43e5e8c00bdbc5f03e3d Mon Sep 17 00:00:00 2001 From: "Ronald S. Bultje" Date: Sat, 11 Feb 2012 08:42:28 -0800 Subject: [PATCH 16/19] swscale: enforce a minimum filtersize. At very small dimensions, this calculation could lead to zero-sized filters, which leads to uninitialized output, zero-sized allocations, loop overflows in SIMD that uses do{..}while(i++ srcW-2) filterSize=srcW-2; + filterSize = av_clip(filterSize, 1, srcW - 2); FF_ALLOC_OR_GOTO(NULL, filter, dstW*sizeof(*filter)*filterSize, fail); From 4fbccfab2371eb400efc51e550bf89adb87f1070 Mon Sep 17 00:00:00 2001 From: Aneesh Dogra Date: Sat, 11 Feb 2012 10:42:45 +0530 Subject: [PATCH 17/19] sunrast: Add support for gray8 decoding. Signed-off-by: Ronald S. Bultje --- libavcodec/sunrast.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/libavcodec/sunrast.c b/libavcodec/sunrast.c index 601ba6efc9..f03f24c158 100644 --- a/libavcodec/sunrast.c +++ b/libavcodec/sunrast.c @@ -112,7 +112,7 @@ static int sunrast_decode_frame(AVCodecContext *avctx, void *data, avctx->pix_fmt = PIX_FMT_MONOWHITE; break; case 8: - avctx->pix_fmt = PIX_FMT_PAL8; + avctx->pix_fmt = maplength ? PIX_FMT_PAL8 : PIX_FMT_GRAY8; break; case 24: avctx->pix_fmt = (type == RT_FORMAT_RGB) ? PIX_FMT_RGB24 : PIX_FMT_BGR24; @@ -140,13 +140,9 @@ static int sunrast_decode_frame(AVCodecContext *avctx, void *data, if (depth != 8 && maplength) { av_log(avctx, AV_LOG_WARNING, "useless colormap found or file is corrupted, trying to recover\n"); - } else if (depth == 8) { + } else if (maplength) { unsigned int len = maplength / 3; - if (!maplength) { - av_log(avctx, AV_LOG_ERROR, "colormap expected\n"); - return -1; - } if (maplength % 3 || maplength > 768) { av_log(avctx, AV_LOG_WARNING, "invalid colormap length\n"); return -1; From 41afdd913af9ac442b37c8bd39798576c66974c0 Mon Sep 17 00:00:00 2001 From: Aneesh Dogra Date: Sun, 12 Feb 2012 01:59:56 +0530 Subject: [PATCH 18/19] sunrast: Return AVERROR values instead of -1. Signed-off-by: Ronald S. Bultje --- libavcodec/sunrast.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/libavcodec/sunrast.c b/libavcodec/sunrast.c index f03f24c158..d01a1ef381 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; 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; } @@ -119,7 +120,7 @@ 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]) @@ -127,9 +128,9 @@ 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; @@ -145,7 +146,7 @@ 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]; From b498867d6691b5f1f107afd81aff403f66b434aa Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Sat, 11 Feb 2012 15:58:04 -0500 Subject: [PATCH 19/19] FATE: update reference for seek-alac_mp4 This should have been updated in b590f3a7bf9103ac7a7a61c48568676201d6824b. --- tests/ref/seek/alac_m4a | 54 ++++++++++++++++++++--------------------- 1 file changed, 27 insertions(+), 27 deletions(-) diff --git a/tests/ref/seek/alac_m4a b/tests/ref/seek/alac_m4a index 892cfb5961..51e23e1dd4 100644 --- a/tests/ref/seek/alac_m4a +++ b/tests/ref/seek/alac_m4a @@ -1,53 +1,53 @@ -ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3240 +ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3236 ret: 0 st:-1 flags:0 ts:-1.000000 -ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3240 +ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3236 ret: 0 st:-1 flags:1 ts: 1.894167 -ret: 0 st: 0 flags:1 dts: 1.857596 pts: 1.857596 pos: 73731 size: 4965 +ret: 0 st: 0 flags:1 dts: 1.857596 pts: 1.857596 pos: 73651 size: 4961 ret: 0 st: 0 flags:0 ts: 0.788345 -ret: 0 st: 0 flags:1 dts: 0.835918 pts: 0.835918 pos: 29072 size: 3198 +ret: 0 st: 0 flags:1 dts: 0.835918 pts: 0.835918 pos: 29036 size: 3194 ret: 0 st: 0 flags:1 ts:-0.317506 -ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3240 +ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3236 ret: 0 st:-1 flags:0 ts: 2.576668 -ret: 0 st: 0 flags:1 dts: 2.600635 pts: 2.600635 pos: 137669 size: 12847 +ret: 0 st: 0 flags:1 dts: 2.600635 pts: 2.600635 pos: 137557 size: 12843 ret: 0 st:-1 flags:1 ts: 1.470835 -ret: 0 st: 0 flags:1 dts: 1.393197 pts: 1.393197 pos: 50219 size: 4418 +ret: 0 st: 0 flags:1 dts: 1.393197 pts: 1.393197 pos: 50159 size: 4414 ret: 0 st: 0 flags:0 ts: 0.365011 -ret: 0 st: 0 flags:1 dts: 0.371519 pts: 0.371519 pos: 12962 size: 3213 +ret: 0 st: 0 flags:1 dts: 0.371519 pts: 0.371519 pos: 12946 size: 3209 ret: 0 st: 0 flags:1 ts:-0.740839 -ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3240 +ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3236 ret: 0 st:-1 flags:0 ts: 2.153336 -ret: 0 st: 0 flags:1 dts: 2.229116 pts: 2.229116 pos: 101031 size: 7900 +ret: 0 st: 0 flags:1 dts: 2.229116 pts: 2.229116 pos: 100935 size: 7896 ret: 0 st:-1 flags:1 ts: 1.047503 -ret: 0 st: 0 flags:1 dts: 1.021678 pts: 1.021678 pos: 35362 size: 3035 +ret: 0 st: 0 flags:1 dts: 1.021678 pts: 1.021678 pos: 35318 size: 3031 ret: 0 st: 0 flags:0 ts:-0.058322 -ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3240 +ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3236 ret: 0 st: 0 flags:1 ts: 2.835828 -ret: 0 st: 0 flags:1 dts: 2.786395 pts: 2.786395 pos: 163300 size: 12769 +ret: 0 st: 0 flags:1 dts: 2.786395 pts: 2.786395 pos: 163180 size: 12765 ret: 0 st:-1 flags:0 ts: 1.730004 -ret: 0 st: 0 flags:1 dts: 1.764717 pts: 1.764717 pos: 68756 size: 4975 +ret: 0 st: 0 flags:1 dts: 1.764717 pts: 1.764717 pos: 68680 size: 4971 ret: 0 st:-1 flags:1 ts: 0.624171 -ret: 0 st: 0 flags:1 dts: 0.557279 pts: 0.557279 pos: 19365 size: 3238 +ret: 0 st: 0 flags:1 dts: 0.557279 pts: 0.557279 pos: 19341 size: 3234 ret: 0 st: 0 flags:0 ts:-0.481655 -ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3240 +ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3236 ret: 0 st: 0 flags:1 ts: 2.412494 -ret: 0 st: 0 flags:1 dts: 2.321995 pts: 2.321995 pos: 108931 size: 7890 +ret: 0 st: 0 flags:1 dts: 2.321995 pts: 2.321995 pos: 108831 size: 7886 ret: 0 st:-1 flags:0 ts: 1.306672 -ret: 0 st: 0 flags:1 dts: 1.393197 pts: 1.393197 pos: 50219 size: 4418 +ret: 0 st: 0 flags:1 dts: 1.393197 pts: 1.393197 pos: 50159 size: 4414 ret: 0 st:-1 flags:1 ts: 0.200839 -ret: 0 st: 0 flags:1 dts: 0.185760 pts: 0.185760 pos: 6478 size: 3249 +ret: 0 st: 0 flags:1 dts: 0.185760 pts: 0.185760 pos: 6470 size: 3245 ret: 0 st: 0 flags:0 ts:-0.904989 -ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3240 +ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3236 ret: 0 st: 0 flags:1 ts: 1.989184 -ret: 0 st: 0 flags:1 dts: 1.950476 pts: 1.950476 pos: 78696 size: 6518 +ret: 0 st: 0 flags:1 dts: 1.950476 pts: 1.950476 pos: 78612 size: 6514 ret: 0 st:-1 flags:0 ts: 0.883340 -ret: 0 st: 0 flags:1 dts: 0.928798 pts: 0.928798 pos: 32270 size: 3092 +ret: 0 st: 0 flags:1 dts: 0.928798 pts: 0.928798 pos: 32230 size: 3088 ret: 0 st:-1 flags:1 ts:-0.222493 -ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3240 +ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 40 size: 3236 ret: 0 st: 0 flags:0 ts: 2.671678 -ret: 0 st: 0 flags:1 dts: 2.693515 pts: 2.693515 pos: 150516 size: 12784 +ret: 0 st: 0 flags:1 dts: 2.693515 pts: 2.693515 pos: 150400 size: 12780 ret: 0 st: 0 flags:1 ts: 1.565850 -ret: 0 st: 0 flags:1 dts: 1.486077 pts: 1.486077 pos: 54637 size: 4558 +ret: 0 st: 0 flags:1 dts: 1.486077 pts: 1.486077 pos: 54573 size: 4554 ret: 0 st:-1 flags:0 ts: 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