* Copyright (C) 2006 Benjamin Larsson
* Copyright (C) 2007 Konstantin Shishkov
*
- * 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
*/
* All 2 channel configurations -> AV_CH_LAYOUT_STEREO
*/
-static const int64_t dca_core_channel_layout[] = {
+static const uint64_t dca_core_channel_layout[] = {
AV_CH_FRONT_CENTER, ///< 1, A
AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
typedef struct {
AVCodecContext *avctx;
+ AVFrame frame;
/* Frame header */
int frame_type; ///< type of the current frame
int samples_deficit; ///< deficit sample count
/* Primary audio coding header */
int subframes; ///< number of subframes
- int is_channels_set; ///< check for if the channel number is already set
int total_channels; ///< number of channels including extensions
int prim_channels; ///< number of primary audio channels
int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
/* Sync code */
- get_bits(&s->gb, 32);
+ skip_bits_long(&s->gb, 32);
/* Frame header */
s->frame_type = get_bits(&s->gb, 1);
}
+#ifndef decode_blockcodes
/* Very compact version of the block code decoder that does not use table
* look-up but is slightly slower */
static int decode_blockcode(int code, int levels, int *values)
code = div;
}
- if (code == 0)
- return 0;
- else {
- av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
- return AVERROR_INVALIDDATA;
- }
+ return code;
+}
+
+static int decode_blockcodes(int code1, int code2, int levels, int *values)
+{
+ return decode_blockcode(code1, levels, values) |
+ decode_blockcode(code2, levels, values + 4);
}
+#endif
static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
if (abits <= 7){
/* Block code */
- int block_code1, block_code2, size, levels;
+ int block_code1, block_code2, size, levels, err;
size = abits_sizes[abits-1];
levels = abits_levels[abits-1];
block_code1 = get_bits(&s->gb, size);
- /* FIXME Should test return value */
- decode_blockcode(block_code1, levels, block);
block_code2 = get_bits(&s->gb, size);
- decode_blockcode(block_code2, levels, &block[4]);
+ err = decode_blockcodes(block_code1, block_code2,
+ levels, block);
+ if (err) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "ERROR: block code look-up failed\n");
+ return AVERROR_INVALIDDATA;
+ }
}else{
/* no coding */
for (m = 0; m < 8; m++)
/* presumably optional information only appears in the core? */
if (!base_channel) {
if (s->timestamp)
- get_bits(&s->gb, 32);
+ skip_bits_long(&s->gb, 32);
if (s->aux_data)
aux_data_count = get_bits(&s->gb, 6);
* Main frame decoding function
* FIXME add arguments
*/
-static int dca_decode_frame(AVCodecContext * avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+static int dca_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int lfe_samples;
int num_core_channels = 0;
- int i;
- float *samples_flt = data;
- int16_t *samples_s16 = data;
- int out_size;
+ int i, ret;
+ float *samples_flt;
+ int16_t *samples_s16;
DCAContext *s = avctx->priv_data;
int channels;
int core_ss_end;
}
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
- if (dca_parse_frame_header(s) < 0) {
+ if ((ret = dca_parse_frame_header(s)) < 0) {
//seems like the frame is corrupt, try with the next one
- *data_size=0;
- return buf_size;
+ return ret;
}
//set AVCodec values with parsed data
avctx->sample_rate = s->sample_rate;
s->profile = FF_PROFILE_DTS;
for (i = 0; i < (s->sample_blocks / 8); i++) {
- dca_decode_block(s, 0, i);
+ if ((ret = dca_decode_block(s, 0, i))) {
+ av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
+ return ret;
+ }
}
/* record number of core channels incase less than max channels are requested */
dca_parse_audio_coding_header(s, s->xch_base_channel);
for (i = 0; i < (s->sample_blocks / 8); i++) {
- dca_decode_block(s, s->xch_base_channel, i);
+ if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
+ av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
+ continue;
+ }
}
s->xch_present = 1;
s->output = DCA_STEREO;
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
}
+ else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
+ static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };
+ s->channel_order_tab = dca_channel_order_native;
+ }
} else {
av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);
return AVERROR_INVALIDDATA;
}
-
- /* There is nothing that prevents a dts frame to change channel configuration
- but Libav doesn't support that so only set the channels if it is previously
- unset. Ideally during the first probe for channels the crc should be checked
- and only set avctx->channels when the crc is ok. Right now the decoder could
- set the channels based on a broken first frame.*/
- if (s->is_channels_set == 0) {
- s->is_channels_set = 1;
- avctx->channels = channels;
- }
if (avctx->channels != channels) {
- av_log(avctx, AV_LOG_ERROR, "DCA decoder does not support number of "
- "channels changing in stream. Skipping frame.\n");
- return AVERROR_PATCHWELCOME;
+ if (avctx->channels)
+ av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);
+ avctx->channels = channels;
}
- out_size = 256 / 8 * s->sample_blocks * channels *
- av_get_bytes_per_sample(avctx->sample_fmt);
- if (*data_size < out_size)
- return AVERROR(EINVAL);
- *data_size = out_size;
+ /* get output buffer */
+ s->frame.nb_samples = 256 * (s->sample_blocks / 8);
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ samples_flt = (float *)s->frame.data[0];
+ samples_s16 = (int16_t *)s->frame.data[0];
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
}
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
+
return buf_size;
}
avctx->channels = avctx->request_channels;
}
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
return 0;
}
.decode = dca_decode_frame,
.close = dca_decode_end,
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
- .capabilities = CODEC_CAP_CHANNEL_CONF,
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
projects / ffmpeg / blobdiff