[FFmpeg-cvslog] af_hdcd: add experimental 20 and 24-bit decoding support
Burt P
git at videolan.org
Wed Oct 5 20:57:53 EEST 2016
ffmpeg | branch: master | Burt P <pburt0 at gmail.com> | Sun Oct 2 02:25:13 2016 -0500| [f51ddbf83cb7193f7aee4cb971dc1f2ca8539e94] | committer: Burt P
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 at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=f51ddbf83cb7193f7aee4cb971dc1f2ca8539e94
---
libavfilter/af_hdcd.c | 88 +++++++++++++++++++++++++++++++++++++++------------
1 file changed, 68 insertions(+), 20 deletions(-)
diff --git a/libavfilter/af_hdcd.c b/libavfilter/af_hdcd.c
index b5aad85..4754329 100644
--- a/libavfilter/af_hdcd.c
+++ b/libavfilter/af_hdcd.c
@@ -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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