Skip to content

F
ffmpeg.wasm-core
Commit 79b77475 authored by Justin Ruggles's avatar Justin Ruggles
Browse files
Options

vorbisdec: use float planar sample format

parent 89923fce
Show whitespace changes
Inline Side-by-side
Showing with 31 additions and 50 deletions
libavcodec/vorbisdec.c
View file @ 79b77475
......@@ -151,9 +151,7 @@ typedef struct vorbis_context_s {
uint8_t mode_number; // mode number for the current packet
uint8_t previous_window;
float *channel_residues;
float *channel_floors;
float *saved;
float scale_bias; // for float->int conversion
} vorbis_context;
/* Helper functions */
......@@ -192,7 +190,6 @@ static void vorbis_free(vorbis_context *vc)
int i;
av_freep(&vc->channel_residues);
av_freep(&vc->channel_floors);
av_freep(&vc->saved);
for (i = 0; i < vc->residue_count; i++)
......@@ -951,12 +948,11 @@ static int vorbis_parse_id_hdr(vorbis_context *vc)
}
vc->channel_residues = av_malloc((vc->blocksize[1] / 2) * vc->audio_channels * sizeof(*vc->channel_residues));
vc->channel_floors = av_malloc((vc->blocksize[1] / 2) * vc->audio_channels * sizeof(*vc->channel_floors));
vc->saved = av_mallocz((vc->blocksize[1] / 4) * vc->audio_channels * sizeof(*vc->saved));
vc->previous_window = 0;
ff_mdct_init(&vc->mdct[0], bl0, 1, -vc->scale_bias);
ff_mdct_init(&vc->mdct[1], bl1, 1, -vc->scale_bias);
ff_mdct_init(&vc->mdct[0], bl0, 1, -1.0);
ff_mdct_init(&vc->mdct[1], bl1, 1, -1.0);
av_dlog(NULL, " vorbis version %d \n audio_channels %d \n audio_samplerate %d \n bitrate_max %d \n bitrate_nom %d \n bitrate_min %d \n blk_0 %d blk_1 %d \n ",
vc->version, vc->audio_channels, vc->audio_samplerate, vc->bitrate_maximum, vc->bitrate_nominal, vc->bitrate_minimum, vc->blocksize[0], vc->blocksize[1]);
......@@ -988,13 +984,7 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
avpriv_float_dsp_init(&vc->fdsp, avccontext->flags & CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&vc->fmt_conv, avccontext);
if (avccontext->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avccontext->sample_fmt = AV_SAMPLE_FMT_FLT;
vc->scale_bias = 1.0f;
} else {
avccontext->sample_fmt = AV_SAMPLE_FMT_S16;
vc->scale_bias = 32768.0f;
}
avccontext->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (!headers_len) {
av_log(avccontext, AV_LOG_ERROR, "Extradata missing.\n");
......@@ -1485,7 +1475,7 @@ void ff_vorbis_inverse_coupling(float *mag, float *ang, int blocksize)
// Decode the audio packet using the functions above
static int vorbis_parse_audio_packet(vorbis_context *vc)
static int vorbis_parse_audio_packet(vorbis_context *vc, float **floor_ptr)
{
GetBitContext *gb = &vc->gb;
FFTContext *mdct;
......@@ -1496,7 +1486,6 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
uint8_t do_not_decode[255];
vorbis_mapping *mapping;
float *ch_res_ptr = vc->channel_residues;
float *ch_floor_ptr = vc->channel_floors;
uint8_t res_chan[255];
unsigned res_num = 0;
int retlen = 0;
......@@ -1528,7 +1517,8 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
}
memset(ch_res_ptr, 0, sizeof(float) * vc->audio_channels * vlen); //FIXME can this be removed ?
memset(ch_floor_ptr, 0, sizeof(float) * vc->audio_channels * vlen); //FIXME can this be removed ?
for (i = 0; i < vc->audio_channels; ++i)
memset(floor_ptr[i], 0, vlen * sizeof(floor_ptr[0][0])); //FIXME can this be removed ?
// Decode floor
......@@ -1541,14 +1531,13 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
floor = &vc->floors[mapping->submap_floor[0]];
}
ret = floor->decode(vc, &floor->data, ch_floor_ptr);
ret = floor->decode(vc, &floor->data, floor_ptr[i]);
if (ret < 0) {
av_log(vc->avccontext, AV_LOG_ERROR, "Invalid codebook in vorbis_floor_decode.\n");
return AVERROR_INVALIDDATA;
}
no_residue[i] = ret;
ch_floor_ptr += vlen;
}
// Nonzero vector propagate
......@@ -1612,10 +1601,9 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
mdct = &vc->mdct[blockflag];
for (j = vc->audio_channels-1;j >= 0; j--) {
ch_floor_ptr = vc->channel_floors + j * blocksize / 2;
ch_res_ptr = vc->channel_residues + res_chan[j] * blocksize / 2;
vc->fdsp.vector_fmul(ch_floor_ptr, ch_floor_ptr, ch_res_ptr, blocksize / 2);
mdct->imdct_half(mdct, ch_res_ptr, ch_floor_ptr);
vc->fdsp.vector_fmul(floor_ptr[j], floor_ptr[j], ch_res_ptr, blocksize / 2);
mdct->imdct_half(mdct, ch_res_ptr, floor_ptr[j]);
}
// Overlap/add, save data for next overlapping
......@@ -1626,7 +1614,7 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
unsigned bs1 = vc->blocksize[1];
float *residue = vc->channel_residues + res_chan[j] * blocksize / 2;
float *saved = vc->saved + j * bs1 / 4;
float *ret = vc->channel_floors + j * retlen;
float *ret = floor_ptr[j];
float *buf = residue;
const float *win = vc->win[blockflag & previous_window];
......@@ -1655,14 +1643,31 @@ static int vorbis_decode_frame(AVCodecContext *avccontext, void *data,
int buf_size = avpkt->size;
vorbis_context *vc = avccontext->priv_data;
GetBitContext *gb = &vc->gb;
const float *channel_ptrs[255];
float *channel_ptrs[255];
int i, len, ret;
av_dlog(NULL, "packet length %d \n", buf_size);
/* get output buffer */
vc->frame.nb_samples = vc->blocksize[1] / 2;
if ((ret = avccontext->get_buffer(avccontext, &vc->frame)) < 0) {
av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
if (vc->audio_channels > 8) {
for (i = 0; i < vc->audio_channels; i++)
channel_ptrs[i] = (float *)vc->frame.extended_data[i];
} else {
for (i = 0; i < vc->audio_channels; i++) {
int ch = ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i];
channel_ptrs[ch] = (float *)vc->frame.extended_data[i];
}
}
init_get_bits(gb, buf, buf_size*8);
if ((len = vorbis_parse_audio_packet(vc)) <= 0)
if ((len = vorbis_parse_audio_packet(vc, channel_ptrs)) <= 0)
return len;
if (!vc->first_frame) {
......@@ -1674,30 +1679,7 @@ static int vorbis_decode_frame(AVCodecContext *avccontext, void *data,
av_dlog(NULL, "parsed %d bytes %d bits, returned %d samples (*ch*bits) \n",
get_bits_count(gb) / 8, get_bits_count(gb) % 8, len);
/* get output buffer */
vc->frame.nb_samples = len;
if ((ret = avccontext->get_buffer(avccontext, &vc->frame)) < 0) {
av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
if (vc->audio_channels > 8) {
for (i = 0; i < vc->audio_channels; i++)
channel_ptrs[i] = vc->channel_floors + i * len;
} else {
for (i = 0; i < vc->audio_channels; i++)
channel_ptrs[i] = vc->channel_floors +
len * ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i];
}
if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT)
vc->fmt_conv.float_interleave((float *)vc->frame.data[0], channel_ptrs,
len, vc->audio_channels);
else
vc->fmt_conv.float_to_int16_interleave((int16_t *)vc->frame.data[0],
channel_ptrs, len,
vc->audio_channels);
*got_frame_ptr = 1;
*(AVFrame *)data = vc->frame;
......@@ -1738,7 +1720,6 @@ AVCodec ff_vorbis_decoder = {
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
.channel_layouts = ff_vorbis_channel_layouts,
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment