[FFmpeg-devel] [PATCH] avformat/flacdec: Implement decoding of 32-bit PCM
Martijn van Beurden
mvanb1 at gmail.com
Mon Jun 20 23:14:35 EEST 2022
Recently libFLAC gained the ability (not in any released version yet
though) to create FLAC files containing 32-bit int PCM samples. To
keep complexity reasonable, the choice was made to limit residuals
to 32-bit integers, which the encoder must make sure of. In case
the encoder cannot find any predictor of which the residuals fit
this limit, it must default to using a verbatim subframe. Tests have
shown that this does not happen often (<0.1% of subframes on a
music corpus of various styles). See also discussion here:
https://github.com/ietf-wg-cellar/flac-specification/pull/148
This patch adds decoding of these files to ffmpeg.
---
libavcodec/flac.c | 4 +-
libavcodec/flacdec.c | 248 ++++++++++++++++++++++++++++++++++++++----
libavcodec/get_bits.h | 12 ++
libavcodec/mathops.h | 9 ++
4 files changed, 250 insertions(+), 23 deletions(-)
diff --git a/libavcodec/flac.c b/libavcodec/flac.c
index dd68830622..f326d8fa5c 100644
--- a/libavcodec/flac.c
+++ b/libavcodec/flac.c
@@ -27,7 +27,7 @@
#include "flac.h"
#include "flacdata.h"
-static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 };
+static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 32 };
static const AVChannelLayout flac_channel_layouts[8] = {
AV_CHANNEL_LAYOUT_MONO,
@@ -81,7 +81,7 @@ int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
/* bits per sample */
bps_code = get_bits(gb, 3);
- if (bps_code == 3 || bps_code == 7) {
+ if (bps_code == 3) {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"invalid sample size code (%d)\n",
bps_code);
diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c
index 87f20c7425..49952ce120 100644
--- a/libavcodec/flacdec.c
+++ b/libavcodec/flacdec.c
@@ -63,6 +63,9 @@ typedef struct FLACContext {
int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples
uint8_t *decoded_buffer;
unsigned int decoded_buffer_size;
+ int64_t *decoded_33bps; ///< decoded samples for a 33 bps subframe
+ uint8_t *decoded_buffer_33bps;
+ unsigned int decoded_buffer_size_33bps;
int buggy_lpc; ///< use workaround for old lavc encoded files
FLACDSPContext dsp;
@@ -154,6 +157,24 @@ static int allocate_buffers(FLACContext *s)
s->flac_stream_info.channels,
s->flac_stream_info.max_blocksize,
AV_SAMPLE_FMT_S32P, 0);
+ if (ret >= 0 && s->flac_stream_info.bps == 32 && s->flac_stream_info.channels == 2) {
+ buf_size = av_samples_get_buffer_size(NULL, 1,
+ s->flac_stream_info.max_blocksize,
+ AV_SAMPLE_FMT_S64P, 0);
+ if (buf_size < 0)
+ return buf_size;
+
+ av_fast_malloc(&s->decoded_buffer_33bps, &s->decoded_buffer_size_33bps, buf_size);
+ if (!s->decoded_buffer)
+ return AVERROR(ENOMEM);
+
+ ret = av_samples_fill_arrays((uint8_t **)&s->decoded_33bps, NULL,
+ s->decoded_buffer_33bps,
+ 1,
+ s->flac_stream_info.max_blocksize,
+ AV_SAMPLE_FMT_S64P, 0);
+
+ }
return ret < 0 ? ret : 0;
}
@@ -331,6 +352,94 @@ static int decode_subframe_fixed(FLACContext *s, int32_t *decoded,
return 0;
}
+static int decode_subframe_fixed_wide(FLACContext *s, int32_t *decoded,
+ int pred_order, int bps)
+{
+ const int blocksize = s->blocksize;
+ int i;
+ int ret;
+
+ /* warm up samples */
+ for (i = 0; i < pred_order; i++) {
+ decoded[i] = get_sbits_long(&s->gb, bps);
+ }
+
+ if ((ret = decode_residuals(s, decoded, pred_order)) < 0)
+ return ret;
+
+ switch (pred_order) {
+ case 0:
+ break;
+ case 1:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] += decoded[i-1];
+ break;
+ case 2:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = (int64_t)decoded[i] + 2*(int64_t)decoded[i-1] - (int64_t)decoded[i-2];
+ break;
+ case 3:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = (int64_t)decoded[i] + 3*(int64_t)decoded[i-1] - 3*(int64_t)decoded[i-2] + (int64_t)decoded[i-3];
+ break;
+ case 4:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = (int64_t)decoded[i] + 4*(int64_t)decoded[i-1] - 6*(int64_t)decoded[i-2] + 4*(int64_t)decoded[i-3] - (int64_t)decoded[i-4];
+ break;
+ default:
+ av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order);
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+
+static int decode_subframe_fixed_33bps(FLACContext *s, int64_t *decoded,
+ int32_t *residual, int pred_order)
+{
+ const int blocksize = s->blocksize;
+ int i;
+ int ret;
+
+ /* warm up samples */
+ for (i = 0; i < pred_order; i++) {
+ decoded[i] = get_sbits64(&s->gb, 33);
+ av_log(s->avctx, AV_LOG_DEBUG, "warm-up %d = %" PRId64 "\n", i, decoded[i]);
+ }
+
+ if ((ret = decode_residuals(s, residual, pred_order)) < 0)
+ return ret;
+
+ switch (pred_order) {
+ case 0:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = residual[i];
+ break;
+ case 1:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = residual[i] + decoded[i-1];
+ break;
+ case 2:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = residual[i] + 2*decoded[i-1] - decoded[i-2];
+ break;
+ case 3:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = residual[i] + 3*decoded[i-1] - 3*decoded[i-2] + decoded[i-3];
+ break;
+ case 4:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = residual[i] + 4*decoded[i-1] - 6*decoded[i-2] + 4*decoded[i-3] - decoded[i-4];
+ break;
+ default:
+ av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order);
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
static void lpc_analyze_remodulate(SUINT32 *decoded, const int coeffs[32],
int order, int qlevel, int len, int bps)
{
@@ -402,12 +511,53 @@ static int decode_subframe_lpc(FLACContext *s, int32_t *decoded, int pred_order,
return 0;
}
+static int decode_subframe_lpc_33bps(FLACContext *s, int64_t *decoded,
+ int32_t *residual, int pred_order)
+{
+ int i, j, ret;
+ int coeff_prec, qlevel;
+ int coeffs[32];
+
+ /* warm up samples */
+ for (i = 0; i < pred_order; i++) {
+ decoded[i] = get_sbits64(&s->gb, 33);
+ }
+
+ coeff_prec = get_bits(&s->gb, 4) + 1;
+ if (coeff_prec == 16) {
+ av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n");
+ return AVERROR_INVALIDDATA;
+ }
+ qlevel = get_sbits(&s->gb, 5);
+ if (qlevel < 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "qlevel %d not supported, maybe buggy stream\n",
+ qlevel);
+ return AVERROR_INVALIDDATA;
+ }
+
+ for (i = 0; i < pred_order; i++) {
+ coeffs[pred_order - i - 1] = get_sbits(&s->gb, coeff_prec);
+ }
+
+ if ((ret = decode_residuals(s, residual, pred_order)) < 0)
+ return ret;
+
+ for (i = pred_order; i < s->blocksize; i++, decoded++) {
+ int64_t sum = 0;
+ for (j = 0; j < pred_order; j++)
+ sum += (int64_t)coeffs[j] * decoded[j];
+ decoded[j] = residual[i] + (sum >> qlevel);
+ }
+
+ return 0;
+}
+
static inline int decode_subframe(FLACContext *s, int channel)
{
int32_t *decoded = s->decoded[channel];
int type, wasted = 0;
int bps = s->flac_stream_info.bps;
- int i, tmp, ret;
+ int i, ret;
if (channel == 0) {
if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE)
@@ -436,34 +586,63 @@ static inline int decode_subframe(FLACContext *s, int channel)
wasted = 1 + get_unary(&s->gb, 1, get_bits_left(&s->gb));
bps -= wasted;
}
- if (bps > 32) {
- avpriv_report_missing_feature(s->avctx, "Decorrelated bit depth > 32");
- return AVERROR_PATCHWELCOME;
- }
//FIXME use av_log2 for types
if (type == 0) {
- tmp = get_sbits_long(&s->gb, bps);
- for (i = 0; i < s->blocksize; i++)
- decoded[i] = tmp;
+ if (bps < 33) {
+ int32_t tmp = get_sbits_long(&s->gb, bps);
+ for (i = 0; i < s->blocksize; i++)
+ decoded[i] = tmp;
+ } else {
+ int64_t tmp = get_sbits64(&s->gb, 33);
+ for (i = 0; i < s->blocksize; i++)
+ s->decoded_33bps[i] = tmp;
+ }
} else if (type == 1) {
- for (i = 0; i < s->blocksize; i++)
- decoded[i] = get_sbits_long(&s->gb, bps);
+ if (bps < 33) {
+ for (i = 0; i < s->blocksize; i++)
+ decoded[i] = get_sbits_long(&s->gb, bps);
+ } else {
+ for (i = 0; i < s->blocksize; i++)
+ s->decoded_33bps[i] = get_sbits64(&s->gb, 33);
+ }
} else if ((type >= 8) && (type <= 12)) {
- if ((ret = decode_subframe_fixed(s, decoded, type & ~0x8, bps)) < 0)
- return ret;
+ int order = type & ~0x8;
+ if (bps < 33) {
+ if (bps + order <= 32) {
+ if ((ret = decode_subframe_fixed(s, decoded, order, bps)) < 0)
+ return ret;
+ } else {
+ if ((ret = decode_subframe_fixed_wide(s, decoded, order, bps)) < 0)
+ return ret;
+ }
+ } else {
+ if ((ret = decode_subframe_fixed_33bps(s, s->decoded_33bps, decoded, order)) < 0)
+ return ret;
+ }
} else if (type >= 32) {
- if ((ret = decode_subframe_lpc(s, decoded, (type & ~0x20)+1, bps)) < 0)
- return ret;
+ if (bps < 33) {
+ if ((ret = decode_subframe_lpc(s, decoded, (type & ~0x20)+1, bps)) < 0)
+ return ret;
+ } else {
+ if ((ret = decode_subframe_lpc_33bps(s, s->decoded_33bps, decoded, (type & ~0x20)+1)) < 0)
+ return ret;
+ }
} else {
av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n");
return AVERROR_INVALIDDATA;
}
- if (wasted && wasted < 32) {
- int i;
- for (i = 0; i < s->blocksize; i++)
- decoded[i] = (unsigned)decoded[i] << wasted;
+ if (wasted) {
+ if (wasted+bps == 33) {
+ int i;
+ for (i = 0; i < s->blocksize; i++)
+ s->decoded_33bps[i] = (uint64_t)decoded[i] << wasted;
+ } else if (wasted < 32) {
+ int i;
+ for (i = 0; i < s->blocksize; i++)
+ decoded[i] = (unsigned)decoded[i] << wasted;
+ }
}
return 0;
@@ -554,6 +733,26 @@ static int decode_frame(FLACContext *s)
return 0;
}
+static void decorrelate_33bps(int ch_mode, int32_t **decoded, int64_t *decoded_33bps, int len)
+{
+ int i;
+ if (ch_mode == FLAC_CHMODE_LEFT_SIDE ) {
+ for (i = 0; i < len; i++)
+ decoded[1][i] = decoded[0][i] - decoded_33bps[i];
+ } else if (ch_mode == FLAC_CHMODE_RIGHT_SIDE ) {
+ for (i = 0; i < len; i++)
+ decoded[0][i] = decoded[1][i] + decoded_33bps[i];
+ } else if (ch_mode == FLAC_CHMODE_MID_SIDE ) {
+ for (i = 0; i < len; i++) {
+ uint64_t a = decoded[0][i];
+ int64_t b = decoded_33bps[i];
+ a -= b >> 1;
+ decoded[0][i] = (a + b);
+ decoded[1][i] = a;
+ }
+ }
+}
+
static int flac_decode_frame(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, AVPacket *avpkt)
{
@@ -618,9 +817,15 @@ static int flac_decode_frame(AVCodecContext *avctx, AVFrame *frame,
if ((ret = ff_thread_get_buffer(avctx, frame, 0)) < 0)
return ret;
- s->dsp.decorrelate[s->ch_mode](frame->data, s->decoded,
- s->flac_stream_info.channels,
- s->blocksize, s->sample_shift);
+ if (s->flac_stream_info.bps == 32 && s->ch_mode > 0) {
+ decorrelate_33bps(s->ch_mode, s->decoded, s->decoded_33bps, s->blocksize);
+ s->dsp.decorrelate[0](frame->data, s->decoded, s->flac_stream_info.channels,
+ s->blocksize, s->sample_shift);
+ } else {
+ s->dsp.decorrelate[s->ch_mode](frame->data, s->decoded,
+ s->flac_stream_info.channels,
+ s->blocksize, s->sample_shift);
+ }
if (bytes_read > buf_size) {
av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
@@ -641,6 +846,7 @@ static av_cold int flac_decode_close(AVCodecContext *avctx)
FLACContext *s = avctx->priv_data;
av_freep(&s->decoded_buffer);
+ av_freep(&s->decoded_buffer_33bps);
return 0;
}
diff --git a/libavcodec/get_bits.h b/libavcodec/get_bits.h
index 16f8af5107..c12b62d4be 100644
--- a/libavcodec/get_bits.h
+++ b/libavcodec/get_bits.h
@@ -597,6 +597,18 @@ static inline int get_sbits_long(GetBitContext *s, int n)
return sign_extend(get_bits_long(s, n), n);
}
+/**
+ * Read 0-64 bits as a signed integer.
+ */
+static inline int64_t get_sbits64(GetBitContext *s, int n)
+{
+ // sign_extend(x, 0) is undefined
+ if (!n)
+ return 0;
+
+ return sign_extend64(get_bits64(s, n), n);
+}
+
/**
* Show 0-32 bits.
*/
diff --git a/libavcodec/mathops.h b/libavcodec/mathops.h
index f81d21f9c4..8a82d9d086 100644
--- a/libavcodec/mathops.h
+++ b/libavcodec/mathops.h
@@ -137,6 +137,15 @@ static inline av_const int sign_extend(int val, unsigned bits)
}
#endif
+#ifndef sign_extend64
+static inline av_const int64_t sign_extend64(int64_t val, unsigned bits)
+{
+ unsigned shift = 8 * sizeof(int64_t) - bits;
+ union { uint64_t u; int64_t s; } v = { (uint64_t) val << shift };
+ return v.s >> shift;
+}
+#endif
+
#ifndef zero_extend
static inline av_const unsigned zero_extend(unsigned val, unsigned bits)
{
--
2.30.2
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