LEVEL_MINUS_9DB
};
-/**
- * Table for center mix levels
- * reference: Section 5.4.2.4 cmixlev
- */
-static const uint8_t center_levels[4] = { 4, 5, 6, 5 };
-
-/**
- * Table for surround mix levels
- * reference: Section 5.4.2.5 surmixlev
- */
-static const uint8_t surround_levels[4] = { 4, 6, 7, 6 };
-
/**
* Table for default stereo downmixing coefficients
* reference: Section 7.8.2 Downmixing Into Two Channels
static av_cold int ac3_decode_init(AVCodecContext *avctx)
{
AC3DecodeContext *s = avctx->priv_data;
- s->avctx = avctx;
+ int i;
-#if FF_API_DRC_SCALE
- if (avctx->drc_scale)
- s->drc_scale = avctx->drc_scale;
-#endif
+ s->avctx = avctx;
ff_ac3_common_init();
ac3_tables_init();
ff_mdct_init(&s->imdct_256, 8, 1, 1.0);
ff_mdct_init(&s->imdct_512, 9, 1, 1.0);
ff_kbd_window_init(s->window, 5.0, 256);
- dsputil_init(&s->dsp, avctx);
+ ff_dsputil_init(&s->dsp, avctx);
+ avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&s->fmt_conv, avctx);
av_lfg_init(&s->dith_state, 0);
- /* set scale value for float to int16 conversion */
- if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
- s->mul_bias = 1.0f;
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
- } else {
- s->mul_bias = 32767.0f;
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- }
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
/* allow downmixing to stereo or mono */
if (avctx->channels > 0 && avctx->request_channels > 0 &&
}
s->downmixed = 1;
- avcodec_get_frame_defaults(&s->frame);
- avctx->coded_frame = &s->frame;
+ for (i = 0; i < AC3_MAX_CHANNELS; i++) {
+ s->xcfptr[i] = s->transform_coeffs[i];
+ s->dlyptr[i] = s->delay[i];
+ }
return 0;
}
int i;
/* read the rest of the bsi. read twice for dual mono mode. */
- i = !(s->channel_mode);
+ i = !s->channel_mode;
do {
skip_bits(gbc, 5); // skip dialog normalization
if (get_bits1(gbc))
static void set_downmix_coeffs(AC3DecodeContext *s)
{
int i;
- float cmix = gain_levels[center_levels[s->center_mix_level]];
- float smix = gain_levels[surround_levels[s->surround_mix_level]];
+ float cmix = gain_levels[s-> center_mix_level];
+ float smix = gain_levels[s->surround_mix_level];
float norm0, norm1;
for (i = 0; i < s->fbw_channels; i++) {
int mantissa;
switch (bap) {
case 0:
+ /* random noise with approximate range of -0.707 to 0.707 */
if (dither)
- mantissa = (av_lfg_get(&s->dith_state) & 0x7FFFFF) - 0x400000;
+ mantissa = (av_lfg_get(&s->dith_state) / 362) - 5932275;
else
mantissa = 0;
break;
for (ch = 1; ch <= s->channels; ch++) {
/* transform coefficients for full-bandwidth channel */
decode_transform_coeffs_ch(s, blk, ch, &m);
- /* tranform coefficients for coupling channel come right after the
+ /* transform coefficients for coupling channel come right after the
coefficients for the first coupled channel*/
if (s->channel_in_cpl[ch]) {
if (!got_cplchan) {
for (i = 0; i < 128; i++)
x[i] = s->transform_coeffs[ch][2 * i];
s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x);
- s->dsp.vector_fmul_window(s->output[ch - 1], s->delay[ch - 1],
- s->tmp_output, s->window, 128);
+ s->fdsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
+ s->tmp_output, s->window, 128);
for (i = 0; i < 128; i++)
x[i] = s->transform_coeffs[ch][2 * i + 1];
s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch - 1], x);
} else {
s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
- s->dsp.vector_fmul_window(s->output[ch - 1], s->delay[ch - 1],
- s->tmp_output, s->window, 128);
+ s->fdsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
+ s->tmp_output, s->window, 128);
memcpy(s->delay[ch - 1], s->tmp_output + 128, 128 * sizeof(float));
}
}
}
-/**
- * Downmix the output to mono or stereo.
- */
-void ff_ac3_downmix_c(float (*samples)[256], float (*matrix)[2],
- int out_ch, int in_ch, int len)
-{
- int i, j;
- float v0, v1;
- if (out_ch == 2) {
- for (i = 0; i < len; i++) {
- v0 = v1 = 0.0f;
- for (j = 0; j < in_ch; j++) {
- v0 += samples[j][i] * matrix[j][0];
- v1 += samples[j][i] * matrix[j][1];
- }
- samples[0][i] = v0;
- samples[1][i] = v1;
- }
- } else if (out_ch == 1) {
- for (i = 0; i < len; i++) {
- v0 = 0.0f;
- for (j = 0; j < in_ch; j++)
- v0 += samples[j][i] * matrix[j][0];
- samples[0][i] = v0;
- }
- }
-}
-
/**
* Upmix delay samples from stereo to original channel layout.
*/
int downmix_output;
int cpl_in_use;
GetBitContext *gbc = &s->gbc;
- uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
-
- memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
+ uint8_t bit_alloc_stages[AC3_MAX_CHANNELS] = { 0 };
/* block switch flags */
different_transforms = 0;
}
/* dynamic range */
- i = !(s->channel_mode);
+ i = !s->channel_mode;
do {
if (get_bits1(gbc)) {
s->dynamic_range[i] = ((dynamic_range_tab[get_bits(gbc, 8)] - 1.0) *
if (s->eac3 && get_bits1(gbc)) {
/* TODO: parse enhanced coupling strategy info */
av_log_missing_feature(s->avctx, "Enhanced coupling", 1);
- return -1;
+ return AVERROR_PATCHWELCOME;
}
/* determine which channels are coupled */
/* apply scaling to coefficients (headroom, dynrng) */
for (ch = 1; ch <= s->channels; ch++) {
- float gain = s->mul_bias / 4194304.0f;
+ float gain = 1.0 / 4194304.0f;
if (s->channel_mode == AC3_CHMODE_DUALMONO) {
gain *= s->dynamic_range[2 - ch];
} else {
do_imdct(s, s->channels);
if (downmix_output) {
- s->dsp.ac3_downmix(s->output, s->downmix_coeffs,
- s->out_channels, s->fbw_channels, 256);
+ s->ac3dsp.downmix(s->outptr, s->downmix_coeffs,
+ s->out_channels, s->fbw_channels, 256);
}
} else {
if (downmix_output) {
- s->dsp.ac3_downmix(s->transform_coeffs + 1, s->downmix_coeffs,
- s->out_channels, s->fbw_channels, 256);
+ s->ac3dsp.downmix(s->xcfptr + 1, s->downmix_coeffs,
+ s->out_channels, s->fbw_channels, 256);
}
if (downmix_output && !s->downmixed) {
s->downmixed = 1;
- s->dsp.ac3_downmix(s->delay, s->downmix_coeffs, s->out_channels,
- s->fbw_channels, 128);
+ s->ac3dsp.downmix(s->dlyptr, s->downmix_coeffs, s->out_channels,
+ s->fbw_channels, 128);
}
do_imdct(s, s->out_channels);
static int ac3_decode_frame(AVCodecContext * avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
+ AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
- float *out_samples_flt;
- int16_t *out_samples_s16;
int blk, ch, err, ret;
const uint8_t *channel_map;
const float *output[AC3_MAX_CHANNELS];
if (!err) {
avctx->sample_rate = s->sample_rate;
avctx->bit_rate = s->bit_rate;
+ }
- /* channel config */
+ /* channel config */
+ if (!err || (s->channels && s->out_channels != s->channels)) {
s->out_channels = s->channels;
s->output_mode = s->channel_mode;
if (s->lfe_on)
avctx->request_channels < s->channels) {
s->out_channels = avctx->request_channels;
s->output_mode = avctx->request_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
- s->channel_layout = ff_ac3_channel_layout_tab[s->output_mode];
+ s->channel_layout = avpriv_ac3_channel_layout_tab[s->output_mode];
}
avctx->channels = s->out_channels;
avctx->channel_layout = s->channel_layout;
s->fbw_channels == s->out_channels)) {
set_downmix_coeffs(s);
}
- } else if (!s->out_channels) {
- s->out_channels = avctx->channels;
- if (s->out_channels < s->channels)
- s->output_mode = s->out_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
+ } else if (!s->channels) {
+ av_log(avctx, AV_LOG_ERROR, "unable to determine channel mode\n");
+ return AVERROR_INVALIDDATA;
}
+ avctx->channels = s->out_channels;
+
/* set audio service type based on bitstream mode for AC-3 */
avctx->audio_service_type = s->bitstream_mode;
if (s->bitstream_mode == 0x7 && s->channels > 1)
avctx->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
/* get output buffer */
- s->frame.nb_samples = s->num_blocks * 256;
- if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ frame->nb_samples = s->num_blocks * 256;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- out_samples_flt = (float *)s->frame.data[0];
- out_samples_s16 = (int16_t *)s->frame.data[0];
/* decode the audio blocks */
channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
- for (ch = 0; ch < s->out_channels; ch++)
- output[ch] = s->output[channel_map[ch]];
+ for (ch = 0; ch < s->channels; ch++) {
+ if (ch < s->out_channels)
+ s->outptr[channel_map[ch]] = (float *)frame->data[ch];
+ else
+ s->outptr[ch] = s->output[ch];
+ output[ch] = s->output[ch];
+ }
for (blk = 0; blk < s->num_blocks; blk++) {
if (!err && decode_audio_block(s, blk)) {
av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
err = 1;
}
- if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
- s->fmt_conv.float_interleave(out_samples_flt, output, 256,
- s->out_channels);
- out_samples_flt += 256 * s->out_channels;
- } else {
- s->fmt_conv.float_to_int16_interleave(out_samples_s16, output, 256,
- s->out_channels);
- out_samples_s16 += 256 * s->out_channels;
+ if (err)
+ for (ch = 0; ch < s->out_channels; ch++)
+ memcpy(s->outptr[channel_map[ch]], output[ch], 1024);
+ for (ch = 0; ch < s->out_channels; ch++) {
+ output[ch] = s->outptr[channel_map[ch]];
+ s->outptr[channel_map[ch]] += AC3_BLOCK_SIZE;
}
}
- *got_frame_ptr = 1;
- *(AVFrame *)data = s->frame;
+ /* keep last block for error concealment in next frame */
+ for (ch = 0; ch < s->out_channels; ch++)
+ memcpy(s->output[ch], output[ch], 1024);
+
+ *got_frame_ptr = 1;
return FFMIN(buf_size, s->frame_size);
}
#define OFFSET(x) offsetof(AC3DecodeContext, x)
#define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
static const AVOption options[] = {
- { "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), AV_OPT_TYPE_FLOAT, {1.0}, 0.0, 1.0, PAR },
+ { "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, 0.0, 1.0, PAR },
{ NULL},
};
AVCodec ff_ac3_decoder = {
.name = "ac3",
.type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_AC3,
+ .id = AV_CODEC_ID_AC3,
.priv_data_size = sizeof (AC3DecodeContext),
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
- AV_SAMPLE_FMT_S16,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.priv_class = &ac3_decoder_class,
};
AVCodec ff_eac3_decoder = {
.name = "eac3",
.type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_EAC3,
+ .id = AV_CODEC_ID_EAC3,
.priv_data_size = sizeof (AC3DecodeContext),
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
- AV_SAMPLE_FMT_S16,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.priv_class = &eac3_decoder_class,
};
projects / ffmpeg / blobdiff