/// Whether the LSBs of the matrix output are encoded in the bitstream.
uint8_t lsb_bypass[MAX_MATRICES];
/// Matrix coefficients, stored as 2.14 fixed point.
- int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
+ DECLARE_ALIGNED(32, int32_t, matrix_coeff)[MAX_MATRICES][MAX_CHANNELS];
/// Left shift to apply to noise values in 0x31eb substreams.
uint8_t matrix_noise_shift[MAX_MATRICES];
//@}
/// Current access unit being read has a major sync.
int is_major_sync_unit;
+ /// Size of the major sync unit, in bytes
+ int major_sync_header_size;
+
/// Set if a valid major sync block has been read. Otherwise no decoding is possible.
uint8_t params_valid;
int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
- int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];
+ DECLARE_ALIGNED(32, int32_t, sample_buffer)[MAX_BLOCKSIZE][MAX_CHANNELS];
MLPDSPContext dsp;
} MLPDecodeContext;
return AVERROR_PATCHWELCOME;
}
+ m->major_sync_header_size = mh.header_size;
+
m->access_unit_size = mh.access_unit_size;
m->access_unit_size_pow2 = mh.access_unit_size_pow2;
m->num_substreams = mh.num_substreams;
- m->max_decoded_substream = m->num_substreams - 1;
+
+ /* limit to decoding 3 substreams, as the 4th is used by Dolby Atmos for non-audio data */
+ m->max_decoded_substream = FFMIN(m->num_substreams - 1, 2);
m->avctx->sample_rate = mh.group1_samplerate;
m->avctx->frame_size = mh.access_unit_size;
ch_assign = av_get_channel_layout_channel_index(s->ch_layout,
channel);
}
- if ((unsigned)ch_assign > s->max_matrix_channel) {
+ if (ch_assign < 0 || ch_assign > s->max_matrix_channel) {
avpriv_request_sample(m->avctx,
"Assignment of matrix channel %d to invalid output channel %d",
ch, ch_assign);
s->noisegen_seed = seed;
}
+/** Write the audio data into the output buffer. */
-/** Apply the channel matrices in turn to reconstruct the original audio
- * samples. */
-
-static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
+static int output_data(MLPDecodeContext *m, unsigned int substr,
+ AVFrame *frame, int *got_frame_ptr)
{
+ AVCodecContext *avctx = m->avctx;
SubStream *s = &m->substream[substr];
unsigned int mat;
unsigned int maxchan;
+ int ret;
+ int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
+
+ if (m->avctx->channels != s->max_matrix_channel + 1) {
+ av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (!s->blockpos) {
+ av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
+ return AVERROR_INVALIDDATA;
+ }
maxchan = s->max_matrix_channel;
if (!s->noise_type) {
fill_noise_buffer(m, substr);
}
+ /* Apply the channel matrices in turn to reconstruct the original audio
+ * samples. */
for (mat = 0; mat < s->num_primitive_matrices; mat++) {
unsigned int dest_ch = s->matrix_out_ch[mat];
m->dsp.mlp_rematrix_channel(&m->sample_buffer[0][0],
m->access_unit_size_pow2,
MSB_MASK(s->quant_step_size[dest_ch]));
}
-}
-
-/** Write the audio data into the output buffer. */
-
-static int output_data(MLPDecodeContext *m, unsigned int substr,
- AVFrame *frame, int *got_frame_ptr)
-{
- AVCodecContext *avctx = m->avctx;
- SubStream *s = &m->substream[substr];
- int ret;
- int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
-
- if (m->avctx->channels != s->max_matrix_channel + 1) {
- av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
- return AVERROR_INVALIDDATA;
- }
-
- if (!s->blockpos) {
- av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
- return AVERROR_INVALIDDATA;
- }
/* get output buffer */
frame->nb_samples = s->blockpos;
if (read_major_sync(m, &gb) < 0)
goto error;
m->is_major_sync_unit = 1;
- header_size += 28;
+ header_size += m->major_sync_header_size;
}
if (!m->params_valid) {
buf += substream_data_len[substr];
}
- rematrix_channels(m, m->max_decoded_substream);
-
if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
return ret;