diff options
Diffstat (limited to 'libavcodec/alac.c')
| -rw-r--r-- | libavcodec/alac.c | 104 |
1 files changed, 80 insertions, 24 deletions
diff --git a/libavcodec/alac.c b/libavcodec/alac.c index f972531195..3f37f61883 100644 --- a/libavcodec/alac.c +++ b/libavcodec/alac.c @@ -2,20 +2,20 @@ * ALAC (Apple Lossless Audio Codec) decoder * Copyright (c) 2005 David Hammerton * - * 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 */ @@ -36,8 +36,8 @@ * 8bit compatible version (0) * 8bit sample size * 8bit history mult (40) - * 8bit initial history (14) - * 8bit rice param limit (10) + * 8bit initial history (10) + * 8bit rice param limit (14) * 8bit channels * 16bit maxRun (255) * 32bit max coded frame size (0 means unknown) @@ -50,6 +50,7 @@ #include "get_bits.h" #include "bytestream.h" #include "internal.h" +#include "thread.h" #include "unary.h" #include "mathops.h" #include "alac_data.h" @@ -73,6 +74,8 @@ typedef struct { int extra_bits; /**< number of extra bits beyond 16-bit */ int nb_samples; /**< number of samples in the current frame */ + + int direct_output; } ALACContext; static inline unsigned int decode_scalar(GetBitContext *gb, int k, int bps) @@ -97,7 +100,7 @@ static inline unsigned int decode_scalar(GetBitContext *gb, int k, int bps) return x; } -static void rice_decompress(ALACContext *alac, int32_t *output_buffer, +static int rice_decompress(ALACContext *alac, int32_t *output_buffer, int nb_samples, int bps, int rice_history_mult) { int i; @@ -108,6 +111,9 @@ static void rice_decompress(ALACContext *alac, int32_t *output_buffer, int k; unsigned int x; + if(get_bits_left(&alac->gb) <= 0) + return -1; + /* calculate rice param and decode next value */ k = av_log2((history >> 9) + 3); k = FFMIN(k, alac->rice_limit); @@ -148,6 +154,7 @@ static void rice_decompress(ALACContext *alac, int32_t *output_buffer, history = 0; } } + return 0; } static inline int sign_only(int v) @@ -184,7 +191,7 @@ static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out, } /* read warm-up samples */ - for (i = 1; i <= lpc_order; i++) + for (i = 1; i <= lpc_order && i < nb_samples; i++) buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i], bps); /* NOTE: 4 and 8 are very common cases that could be optimized. */ @@ -263,7 +270,7 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index, alac->extra_bits = get_bits(&alac->gb, 2) << 3; bps = alac->sample_size - alac->extra_bits + channels - 1; - if (bps > 32) { + if (bps > 32U) { av_log(avctx, AV_LOG_ERROR, "bps is unsupported: %d\n", bps); return AVERROR_PATCHWELCOME; } @@ -281,19 +288,18 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index, return AVERROR_INVALIDDATA; } if (!alac->nb_samples) { + ThreadFrame tframe = { .f = frame }; /* get output buffer */ frame->nb_samples = output_samples; - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) { - av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + if ((ret = ff_thread_get_buffer(avctx, &tframe, 0)) < 0) return ret; - } } else if (output_samples != alac->nb_samples) { av_log(avctx, AV_LOG_ERROR, "sample count mismatch: %u != %d\n", output_samples, alac->nb_samples); return AVERROR_INVALIDDATA; } alac->nb_samples = output_samples; - if (alac->sample_size > 16) { + if (alac->direct_output) { for (ch = 0; ch < channels; ch++) alac->output_samples_buffer[ch] = (int32_t *)frame->extended_data[ch_index + ch]; } @@ -324,14 +330,18 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index, if (alac->extra_bits) { for (i = 0; i < alac->nb_samples; i++) { + if(get_bits_left(&alac->gb) <= 0) + return -1; for (ch = 0; ch < channels; ch++) alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits); } } for (ch = 0; ch < channels; ch++) { - rice_decompress(alac, alac->predict_error_buffer[ch], + int ret=rice_decompress(alac, alac->predict_error_buffer[ch], alac->nb_samples, bps, rice_history_mult[ch] * alac->rice_history_mult / 4); + if(ret<0) + return ret; /* adaptive FIR filter */ if (prediction_type[ch] == 15) { @@ -356,6 +366,8 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index, } else { /* not compressed, easy case */ for (i = 0; i < alac->nb_samples; i++) { + if(get_bits_left(&alac->gb) <= 0) + return -1; for (ch = 0; ch < channels; ch++) { alac->output_samples_buffer[ch][i] = get_sbits_long(&alac->gb, alac->sample_size); @@ -376,6 +388,7 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index, alac->extra_bits, channels, alac->nb_samples); } + if(av_sample_fmt_is_planar(avctx->sample_fmt)) { switch(alac->sample_size) { case 16: { for (ch = 0; ch < channels; ch++) { @@ -391,6 +404,37 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index, }} break; } + }else{ + switch(alac->sample_size) { + case 16: { + int16_t *outbuffer = ((int16_t *)frame->extended_data[0]) + ch_index; + for (i = 0; i < alac->nb_samples; i++) { + for (ch = 0; ch < channels; ch++) + *outbuffer++ = alac->output_samples_buffer[ch][i]; + outbuffer += alac->channels - channels; + } + } + break; + case 24: { + int32_t *outbuffer = ((int32_t *)frame->extended_data[0]) + ch_index; + for (i = 0; i < alac->nb_samples; i++) { + for (ch = 0; ch < channels; ch++) + *outbuffer++ = alac->output_samples_buffer[ch][i] << 8; + outbuffer += alac->channels - channels; + } + } + break; + case 32: { + int32_t *outbuffer = ((int32_t *)frame->extended_data[0]) + ch_index; + for (i = 0; i < alac->nb_samples; i++) { + for (ch = 0; ch < channels; ch++) + *outbuffer++ = alac->output_samples_buffer[ch][i]; + outbuffer += alac->channels - channels; + } + } + break; + } + } return 0; } @@ -404,7 +448,8 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data, int channels; int ch, ret, got_end; - init_get_bits(&alac->gb, avpkt->data, avpkt->size * 8); + if ((ret = init_get_bits8(&alac->gb, avpkt->data, avpkt->size)) < 0) + return ret; got_end = 0; alac->nb_samples = 0; @@ -416,7 +461,7 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data, break; } if (element > TYPE_CPE && element != TYPE_LFE) { - av_log(avctx, AV_LOG_ERROR, "syntax element unsupported: %d", element); + av_log(avctx, AV_LOG_ERROR, "syntax element unsupported: %d\n", element); return AVERROR_PATCHWELCOME; } @@ -457,7 +502,7 @@ static av_cold int alac_decode_close(AVCodecContext *avctx) int ch; for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) { av_freep(&alac->predict_error_buffer[ch]); - if (alac->sample_size == 16) + if (!alac->direct_output) av_freep(&alac->output_samples_buffer[ch]); av_freep(&alac->extra_bits_buffer[ch]); } @@ -474,7 +519,8 @@ static int allocate_buffers(ALACContext *alac) FF_ALLOC_OR_GOTO(alac->avctx, alac->predict_error_buffer[ch], buf_size, buf_alloc_fail); - if (alac->sample_size == 16) { + alac->direct_output = alac->sample_size > 16 && av_sample_fmt_is_planar(alac->avctx->sample_fmt); + if (!alac->direct_output) { FF_ALLOC_OR_GOTO(alac->avctx, alac->output_samples_buffer[ch], buf_size, buf_alloc_fail); } @@ -521,24 +567,26 @@ static int alac_set_info(ALACContext *alac) static av_cold int alac_decode_init(AVCodecContext * avctx) { int ret; + int req_packed; ALACContext *alac = avctx->priv_data; alac->avctx = avctx; /* initialize from the extradata */ if (alac->avctx->extradata_size < ALAC_EXTRADATA_SIZE) { - av_log(avctx, AV_LOG_ERROR, "alac: extradata is too small\n"); + av_log(avctx, AV_LOG_ERROR, "extradata is too small\n"); return AVERROR_INVALIDDATA; } if (alac_set_info(alac)) { - av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n"); + av_log(avctx, AV_LOG_ERROR, "set_info failed\n"); return -1; } + req_packed = LIBAVCODEC_VERSION_MAJOR < 55 && !av_sample_fmt_is_planar(avctx->request_sample_fmt); switch (alac->sample_size) { - case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16P; + case 16: avctx->sample_fmt = req_packed ? AV_SAMPLE_FMT_S16 : AV_SAMPLE_FMT_S16P; break; case 24: - case 32: avctx->sample_fmt = AV_SAMPLE_FMT_S32P; + case 32: avctx->sample_fmt = req_packed ? AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S32P; break; default: avpriv_request_sample(avctx, "Sample depth %d", alac->sample_size); return AVERROR_PATCHWELCOME; @@ -554,7 +602,7 @@ static av_cold int alac_decode_init(AVCodecContext * avctx) else avctx->channels = alac->channels; } - if (avctx->channels > ALAC_MAX_CHANNELS) { + if (avctx->channels > ALAC_MAX_CHANNELS || avctx->channels <= 0 ) { av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n", avctx->channels); return AVERROR_PATCHWELCOME; @@ -569,6 +617,13 @@ static av_cold int alac_decode_init(AVCodecContext * avctx) return 0; } +static int init_thread_copy(AVCodecContext *avctx) +{ + ALACContext *alac = avctx->priv_data; + alac->avctx = avctx; + return allocate_buffers(alac); +} + AVCodec ff_alac_decoder = { .name = "alac", .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), @@ -578,5 +633,6 @@ AVCodec ff_alac_decoder = { .init = alac_decode_init, .close = alac_decode_close, .decode = alac_decode_frame, - .capabilities = CODEC_CAP_DR1, + .init_thread_copy = ONLY_IF_THREADS_ENABLED(init_thread_copy), + .capabilities = CODEC_CAP_DR1 | CODEC_CAP_FRAME_THREADS, }; |