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af_hdcd: add experimental 20 and 24-bit decoding support 

I don't have any legitimate 20 or 24-bit HDCD to test. It is known
that the PM Model Two would insert packets into 20 and 24-bit output,
but I have no idea what differences in behavior existed when decoding
20 or 24-bit. For now, as with 16-bit, PE (if enabled) will expand the
top 3dB into 9dB and LLE (gain adjust) will be applied if signaled.

Signed-off-by: Burt P <pburt0@gmail.com>
tags/n3.2
Burt P 8 years ago
parent
4f94f01414
commit
f51ddbf83c
1 changed files with 68 additions and 20 deletions
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  1. +68
    -20
      libavfilter/af_hdcd.c

+ 68
- 20
libavfilter/af_hdcd.c View File

@@ -964,6 +964,8 @@ typedef struct HDCDContext {
int cdt_ms; /**< code detect timer period in ms */

int disable_autoconvert; /**< disable any format conversion or resampling in the filter graph */

int bits_per_sample; /**< bits per sample 16, 20, or 24 */
/* end AVOption members */

/** config_input() and config_output() scan links for any resampling
@@ -997,6 +999,11 @@ static const AVOption hdcd_options[] = {
{ "pe", HDCD_ANA_PE_DESC, 0, AV_OPT_TYPE_CONST, {.i64=HDCD_ANA_PE}, 0, 0, A, "analyze_mode" },
{ "cdt", HDCD_ANA_CDT_DESC, 0, AV_OPT_TYPE_CONST, {.i64=HDCD_ANA_CDT}, 0, 0, A, "analyze_mode" },
{ "tgm", HDCD_ANA_TGM_DESC, 0, AV_OPT_TYPE_CONST, {.i64=HDCD_ANA_TGM}, 0, 0, A, "analyze_mode" },
{ "bits_per_sample", "Valid bits per sample (location of the true LSB).",
OFFSET(bits_per_sample), AV_OPT_TYPE_INT, { .i64=16 }, 16, 24, A, "bits_per_sample"},
{ "16", "16-bit (in s32 or s16)", 0, AV_OPT_TYPE_CONST, {.i64=16}, 0, 0, A, "bits_per_sample" },
{ "20", "20-bit (in s32)", 0, AV_OPT_TYPE_CONST, {.i64=20}, 0, 0, A, "bits_per_sample" },
{ "24", "24-bit (in s32)", 0, AV_OPT_TYPE_CONST, {.i64=24}, 0, 0, A, "bits_per_sample" },
{NULL}
};

@@ -1253,29 +1260,34 @@ static int hdcd_analyze(int32_t *samples, int count, int stride, int gain, int t
}

/** apply HDCD decoding parameters to a series of samples */
static int hdcd_envelope(int32_t *samples, int count, int stride, int gain, int target_gain, int extend)
static int hdcd_envelope(int32_t *samples, int count, int stride, int vbits, int gain, int target_gain, int extend)
{
static const int max_asample = sizeof(peaktab) / sizeof(peaktab[0]) - 1;
int32_t *samples_end = samples + stride * count;
int i;

int pe_level = PEAK_EXT_LEVEL, shft = 15;
if (vbits != 16) {
pe_level = (1 << (vbits - 1)) - (0x8000 - PEAK_EXT_LEVEL);
shft = 32 - vbits - 1;
}
av_assert0(PEAK_EXT_LEVEL + max_asample == 0x8000);

if (extend) {
for (i = 0; i < count; i++) {
int32_t sample = samples[i * stride];
int32_t asample = abs(sample) - PEAK_EXT_LEVEL;
int32_t asample = abs(sample) - pe_level;
if (asample >= 0) {
av_assert0(asample <= max_asample);
sample = sample >= 0 ? peaktab[asample] : -peaktab[asample];
} else
sample <<= 15;
sample <<= shft;

samples[i * stride] = sample;
}
} else {
for (i = 0; i < count; i++)
samples[i * stride] <<= 15;
samples[i * stride] <<= shft;
}

if (gain <= target_gain) {
@@ -1370,7 +1382,7 @@ static void hdcd_process(HDCDContext *ctx, hdcd_state *state, int32_t *samples,
if (ctx->analyze_mode)
gain = hdcd_analyze(samples, envelope_run, stride, gain, target_gain, peak_extend, ctx->analyze_mode, state->sustain, -1);
else
gain = hdcd_envelope(samples, envelope_run, stride, gain, target_gain, peak_extend);
gain = hdcd_envelope(samples, envelope_run, stride, ctx->bits_per_sample, gain, target_gain, peak_extend);

samples += envelope_run * stride;
count -= envelope_run;
@@ -1382,7 +1394,7 @@ static void hdcd_process(HDCDContext *ctx, hdcd_state *state, int32_t *samples,
if (ctx->analyze_mode)
gain = hdcd_analyze(samples, lead, stride, gain, target_gain, peak_extend, ctx->analyze_mode, state->sustain, -1);
else
gain = hdcd_envelope(samples, lead, stride, gain, target_gain, peak_extend);
gain = hdcd_envelope(samples, lead, stride, ctx->bits_per_sample, gain, target_gain, peak_extend);
}

state->running_gain = gain;
@@ -1422,8 +1434,8 @@ static void hdcd_process_stereo(HDCDContext *ctx, int32_t *samples, int count)
ctx->state[1].sustain,
(ctlret == HDCD_TG_MISMATCH) );
} else {
gain[0] = hdcd_envelope(samples, envelope_run, stride, gain[0], ctx->val_target_gain, peak_extend[0]);
gain[1] = hdcd_envelope(samples + 1, envelope_run, stride, gain[1], ctx->val_target_gain, peak_extend[1]);
gain[0] = hdcd_envelope(samples, envelope_run, stride, ctx->bits_per_sample, gain[0], ctx->val_target_gain, peak_extend[0]);
gain[1] = hdcd_envelope(samples + 1, envelope_run, stride, ctx->bits_per_sample, gain[1], ctx->val_target_gain, peak_extend[1]);
}

samples += envelope_run * stride;
@@ -1444,8 +1456,8 @@ static void hdcd_process_stereo(HDCDContext *ctx, int32_t *samples, int count)
ctx->state[1].sustain,
(ctlret == HDCD_TG_MISMATCH) );
} else {
gain[0] = hdcd_envelope(samples, lead, stride, gain[0], ctx->val_target_gain, peak_extend[0]);
gain[1] = hdcd_envelope(samples + 1, lead, stride, gain[1], ctx->val_target_gain, peak_extend[1]);
gain[0] = hdcd_envelope(samples, lead, stride, ctx->bits_per_sample, gain[0], ctx->val_target_gain, peak_extend[0]);
gain[1] = hdcd_envelope(samples + 1, lead, stride, ctx->bits_per_sample, gain[1], ctx->val_target_gain, peak_extend[1]);
}
}

@@ -1516,8 +1528,10 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out;
const int16_t *in_data;
const int32_t *in_data32;
int32_t *out_data;
int n, c, result;
int a = 32 - s->bits_per_sample;

out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
@@ -1533,16 +1547,32 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
out->format = outlink->format; // is this needed?

out_data = (int32_t*)out->data[0];
if (inlink->format == AV_SAMPLE_FMT_S16P) {
for (n = 0; n < in->nb_samples; n++)
for (c = 0; c < in->channels; c++) {
in_data = (int16_t*)in->extended_data[c];
out_data[(n * in->channels) + c] = in_data[n];
}
} else {
in_data = (int16_t*)in->data[0];
for (n = 0; n < in->nb_samples * in->channels; n++)
out_data[n] = in_data[n];
switch (inlink->format) {
case AV_SAMPLE_FMT_S16P:
for (n = 0; n < in->nb_samples; n++)
for (c = 0; c < in->channels; c++) {
in_data = (int16_t*)in->extended_data[c];
out_data[(n * in->channels) + c] = in_data[n];
}
break;
case AV_SAMPLE_FMT_S16:
in_data = (int16_t*)in->data[0];
for (n = 0; n < in->nb_samples * in->channels; n++)
out_data[n] = in_data[n];
break;

case AV_SAMPLE_FMT_S32P:
for (n = 0; n < in->nb_samples; n++)
for (c = 0; c < in->channels; c++) {
in_data32 = (int32_t*)in->extended_data[c];
out_data[(n * in->channels) + c] = in_data32[n] >> a;
}
break;
case AV_SAMPLE_FMT_S32:
in_data32 = (int32_t*)in->data[0];
for (n = 0; n < in->nb_samples * in->channels; n++)
out_data[n] = in_data32[n] >> a;
break;
}

if (s->process_stereo) {
@@ -1583,6 +1613,8 @@ static int query_formats(AVFilterContext *ctx)
static const enum AVSampleFormat sample_fmts_in[] = {
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_NONE
};
static const enum AVSampleFormat sample_fmts_out[] = {
@@ -1684,6 +1716,22 @@ static int config_input(AVFilterLink *inlink) {
av_log(ctx, AV_LOG_VERBOSE, "Auto-convert: %s\n",
(ctx->graph->disable_auto_convert) ? "disabled" : "enabled");

if ((inlink->format == AV_SAMPLE_FMT_S16 ||
inlink->format == AV_SAMPLE_FMT_S16P) &&
s->bits_per_sample != 16) {
av_log(ctx, AV_LOG_WARNING, "bits_per_sample %d does not fit into sample format %s, falling back to 16\n",
s->bits_per_sample, av_get_sample_fmt_name(inlink->format) );
s->bits_per_sample = 16;
} else {
av_log(ctx, AV_LOG_VERBOSE, "Looking for %d-bit HDCD in sample format %s\n",
s->bits_per_sample, av_get_sample_fmt_name(inlink->format) );
}

if (s->bits_per_sample != 16)
av_log(ctx, AV_LOG_WARNING, "20 and 24-bit HDCD decoding is experimental\n");
if (inlink->sample_rate != 44100)
av_log(ctx, AV_LOG_WARNING, "HDCD decoding for sample rates other than 44100 is experimental\n");

hdcd_detect_reset(&s->detect);
for (c = 0; c < HDCD_MAX_CHANNELS; c++) {
hdcd_reset(&s->state[c], inlink->sample_rate, s->cdt_ms);


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