* 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
*/
* 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)
#include "get_bits.h"
#include "bytestream.h"
#include "internal.h"
+#include "thread.h"
#include "unary.h"
#include "mathops.h"
#include "alac_data.h"
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)
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;
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);
history = 0;
}
}
+ return 0;
}
static inline int sign_only(int v)
}
/* 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. */
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;
}
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];
}
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) {
} 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);
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++) {
}}
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;
}
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;
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;
}
channels = (element == TYPE_CPE) ? 2 : 1;
- if (ch + channels > alac->channels) {
+ if ( ch + channels > alac->channels
+ || ff_alac_channel_layout_offsets[alac->channels - 1][ch] + channels > alac->channels
+ ) {
av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n");
return AVERROR_INVALIDDATA;
}
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]);
}
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);
}
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;
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;
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",
.type = AVMEDIA_TYPE_AUDIO,
.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,
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
projects / ffmpeg / blobdiff