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Merge remote-tracking branch 'qatar/master'

Browse Source 
* qatar/master:
  h264: slice-mt: check master context for valid current_picture_ptr
  h264: slice-mt: get last_pic_dropable from master context
  alacenc: add support for multi-channel encoding

Conflicts:
	Changelog
	libavcodec/alac.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-12-06 16:03:13 +01:00
commit 71949ef715
6 changed files with 178 additions and 81 deletions
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1
Changelog
View File

@ -14,6 +14,7 @@ version <next>:
- FFM2 support
- X-Face image encoder and decoder
- 24-bit FLAC encoding
- multi-channel ALAC encoding up to 7.1
- metadata (INFO tag) support in WAV muxer
- subtitles raw text decoder
- support for building DLLs using MSVC

4
libavcodec/Makefile
View File

@ -89,8 +89,8 @@ OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_float.o ac3enc.o ac3tab.o \
ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o
OBJS-$(CONFIG_ALAC_DECODER) += alac.o
OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o
OBJS-$(CONFIG_ALAC_DECODER) += alac.o alac_data.o
OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o alac_data.o
OBJS-$(CONFIG_ALS_DECODER) += alsdec.o bgmc.o mpeg4audio.o
OBJS-$(CONFIG_AMRNB_DECODER) += amrnbdec.o celp_filters.o \
celp_math.o acelp_filters.o \

48
libavcodec/alac.c
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@ -52,9 +52,9 @@
#include "internal.h"
#include "unary.h"
#include "mathops.h"
#include "alac_data.h"
#define ALAC_EXTRADATA_SIZE 36
#define MAX_CHANNELS 8
typedef struct {
AVCodecContext *avctx;
@ -78,40 +78,6 @@ typedef struct {
int direct_output;
} ALACContext;
enum RawDataBlockType {
/* At the moment, only SCE, CPE, LFE, and END are recognized. */
TYPE_SCE,
TYPE_CPE,
TYPE_CCE,
TYPE_LFE,
TYPE_DSE,
TYPE_PCE,
TYPE_FIL,
TYPE_END
};
static const uint8_t alac_channel_layout_offsets[8][8] = {
{ 0 },
{ 0, 1 },
{ 2, 0, 1 },
{ 2, 0, 1, 3 },
{ 2, 0, 1, 3, 4 },
{ 2, 0, 1, 4, 5, 3 },
{ 2, 0, 1, 4, 5, 6, 3 },
{ 2, 6, 7, 0, 1, 4, 5, 3 }
};
static const uint16_t alac_channel_layouts[8] = {
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
AV_CH_LAYOUT_4POINT0,
AV_CH_LAYOUT_5POINT0_BACK,
AV_CH_LAYOUT_5POINT1_BACK,
AV_CH_LAYOUT_6POINT1_BACK,
AV_CH_LAYOUT_7POINT1_WIDE_BACK
};
static inline unsigned int decode_scalar(GetBitContext *gb, int k, int bps)
{
unsigned int x = get_unary_0_9(gb);
@ -475,7 +441,7 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
ALACContext *alac = avctx->priv_data;
enum RawDataBlockType element;
enum AlacRawDataBlockType element;
int channels;
int ch, ret, got_end;
@ -497,14 +463,14 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
channels = (element == TYPE_CPE) ? 2 : 1;
if ( ch + channels > alac->channels
|| alac_channel_layout_offsets[alac->channels - 1][ch] + channels > alac->channels
|| ff_alac_channel_layout_offsets[alac->channels - 1][ch] + channels > alac->channels
) {
av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n");
return AVERROR_INVALIDDATA;
}
ret = decode_element(avctx, data,
alac_channel_layout_offsets[alac->channels - 1][ch],
ff_alac_channel_layout_offsets[alac->channels - 1][ch],
channels);
if (ret < 0 && get_bits_left(&alac->gb))
return ret;
@ -634,17 +600,17 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");
alac->channels = avctx->channels;
} else {
if (alac->channels > MAX_CHANNELS)
if (alac->channels > ALAC_MAX_CHANNELS)
alac->channels = avctx->channels;
else
avctx->channels = alac->channels;
}
if (avctx->channels > MAX_CHANNELS || avctx->channels <= 0 ) {
if (avctx->channels > ALAC_MAX_CHANNELS || avctx->channels <= 0 ) {
av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n",
avctx->channels);
return AVERROR_PATCHWELCOME;
}
avctx->channel_layout = alac_channel_layouts[alac->channels - 1];
avctx->channel_layout = ff_alac_channel_layouts[alac->channels - 1];
if ((ret = allocate_buffers(alac)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n");

56
libavcodec/alac_data.c Normal file
View File

@ -0,0 +1,56 @@
/*
* ALAC encoder and decoder common data
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/channel_layout.h"
#include "alac_data.h"
const uint8_t ff_alac_channel_layout_offsets[ALAC_MAX_CHANNELS][ALAC_MAX_CHANNELS] = {
{ 0 },
{ 0, 1 },
{ 2, 0, 1 },
{ 2, 0, 1, 3 },
{ 2, 0, 1, 3, 4 },
{ 2, 0, 1, 4, 5, 3 },
{ 2, 0, 1, 4, 5, 6, 3 },
{ 2, 6, 7, 0, 1, 4, 5, 3 }
};
const uint64_t ff_alac_channel_layouts[ALAC_MAX_CHANNELS + 1] = {
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
AV_CH_LAYOUT_4POINT0,
AV_CH_LAYOUT_5POINT0_BACK,
AV_CH_LAYOUT_5POINT1_BACK,
AV_CH_LAYOUT_6POINT1_BACK,
AV_CH_LAYOUT_7POINT1_WIDE_BACK,
0
};
const enum AlacRawDataBlockType ff_alac_channel_elements[ALAC_MAX_CHANNELS][5] = {
{ TYPE_SCE, },
{ TYPE_CPE, },
{ TYPE_SCE, TYPE_CPE, },
{ TYPE_SCE, TYPE_CPE, TYPE_SCE },
{ TYPE_SCE, TYPE_CPE, TYPE_CPE, },
{ TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE, },
{ TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_SCE, },
{ TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE, },
};

46
libavcodec/alac_data.h Normal file
View File

@ -0,0 +1,46 @@
/*
* ALAC encoder and decoder common data
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_ALAC_DATA_H
#define AVCODEC_ALAC_DATA_H
#include <stdint.h>
enum AlacRawDataBlockType {
/* At the moment, only SCE, CPE, LFE, and END are recognized. */
TYPE_SCE,
TYPE_CPE,
TYPE_CCE,
TYPE_LFE,
TYPE_DSE,
TYPE_PCE,
TYPE_FIL,
TYPE_END
};
#define ALAC_MAX_CHANNELS 8
extern const uint8_t ff_alac_channel_layout_offsets[ALAC_MAX_CHANNELS][ALAC_MAX_CHANNELS];
extern const uint64_t ff_alac_channel_layouts[ALAC_MAX_CHANNELS + 1];
extern const enum AlacRawDataBlockType ff_alac_channel_elements[ALAC_MAX_CHANNELS][5];
#endif /* AVCODEC_ALAC_DATA_H */

104
libavcodec/alacenc.c
View File

@ -25,9 +25,9 @@
#include "internal.h"
#include "lpc.h"
#include "mathops.h"
#include "alac_data.h"
#define DEFAULT_FRAME_SIZE 4096
#define MAX_CHANNELS 8
#define ALAC_EXTRADATA_SIZE 36
#define ALAC_FRAME_HEADER_SIZE 55
#define ALAC_FRAME_FOOTER_SIZE 3
@ -66,27 +66,27 @@ typedef struct AlacEncodeContext {
int max_coded_frame_size;
int write_sample_size;
int extra_bits;
int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
int32_t sample_buf[2][DEFAULT_FRAME_SIZE];
int32_t predictor_buf[DEFAULT_FRAME_SIZE];
int interlacing_shift;
int interlacing_leftweight;
PutBitContext pbctx;
RiceContext rc;
AlacLPCContext lpc[MAX_CHANNELS];
AlacLPCContext lpc[2];
LPCContext lpc_ctx;
AVCodecContext *avctx;
} AlacEncodeContext;
static void init_sample_buffers(AlacEncodeContext *s,
uint8_t * const *samples)
static void init_sample_buffers(AlacEncodeContext *s, int channels,
uint8_t const *samples[2])
{
int ch, i;
int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -
s->avctx->bits_per_raw_sample;
#define COPY_SAMPLES(type) do { \
for (ch = 0; ch < s->avctx->channels; ch++) { \
for (ch = 0; ch < channels; ch++) { \
int32_t *bptr = s->sample_buf[ch]; \
const type *sptr = (const type *)samples[ch]; \
for (i = 0; i < s->frame_size; i++) \
@ -128,15 +128,18 @@ static void encode_scalar(AlacEncodeContext *s, int x,
}
}
static void write_frame_header(AlacEncodeContext *s)
static void write_element_header(AlacEncodeContext *s,
enum AlacRawDataBlockType element,
int instance)
{
int encode_fs = 0;
if (s->frame_size < DEFAULT_FRAME_SIZE)
encode_fs = 1;
put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1
put_bits(&s->pbctx, 16, 0); // Seems to be zero
put_bits(&s->pbctx, 3, element); // element type
put_bits(&s->pbctx, 4, instance); // element instance
put_bits(&s->pbctx, 12, 0); // unused header bits
put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header
put_bits(&s->pbctx, 2, s->extra_bits >> 3); // Extra bytes (for 24-bit)
put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim
@ -355,42 +358,51 @@ static void alac_entropy_coder(AlacEncodeContext *s)
}
}
static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
uint8_t * const *samples)
static void write_element(AlacEncodeContext *s,
enum AlacRawDataBlockType element, int instance,
const uint8_t *samples0, const uint8_t *samples1)
{
int i, j;
uint8_t const *samples[2] = { samples0, samples1 };
int i, j, channels;
int prediction_type = 0;
PutBitContext *pb = &s->pbctx;
init_put_bits(pb, avpkt->data, avpkt->size);
channels = element == TYPE_CPE ? 2 : 1;
if (s->verbatim) {
write_frame_header(s);
write_element_header(s, element, instance);
/* samples are channel-interleaved in verbatim mode */
if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
int shift = 32 - s->avctx->bits_per_raw_sample;
int32_t * const *samples_s32 = (int32_t * const *)samples;
int32_t const *samples_s32[2] = { (const int32_t *)samples0,
(const int32_t *)samples1 };
for (i = 0; i < s->frame_size; i++)
for (j = 0; j < s->avctx->channels; j++)
for (j = 0; j < channels; j++)
put_sbits(pb, s->avctx->bits_per_raw_sample,
samples_s32[j][i] >> shift);
} else {
int16_t * const *samples_s16 = (int16_t * const *)samples;
int16_t const *samples_s16[2] = { (const int16_t *)samples0,
(const int16_t *)samples1 };
for (i = 0; i < s->frame_size; i++)
for (j = 0; j < s->avctx->channels; j++)
for (j = 0; j < channels; j++)
put_sbits(pb, s->avctx->bits_per_raw_sample,
samples_s16[j][i]);
}
} else {
init_sample_buffers(s, samples);
write_frame_header(s);
s->write_sample_size = s->avctx->bits_per_raw_sample - s->extra_bits +
channels - 1;
if (s->avctx->channels == 2)
init_sample_buffers(s, channels, samples);
write_element_header(s, element, instance);
if (channels == 2)
alac_stereo_decorrelation(s);
else
s->interlacing_shift = s->interlacing_leftweight = 0;
put_bits(pb, 8, s->interlacing_shift);
put_bits(pb, 8, s->interlacing_leftweight);
for (i = 0; i < s->avctx->channels; i++) {
for (i = 0; i < channels; i++) {
calc_predictor_params(s, i);
put_bits(pb, 4, prediction_type);
@ -407,7 +419,7 @@ static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
if (s->extra_bits) {
uint32_t mask = (1 << s->extra_bits) - 1;
for (i = 0; i < s->frame_size; i++) {
for (j = 0; j < s->avctx->channels; j++) {
for (j = 0; j < channels; j++) {
put_bits(pb, s->extra_bits, s->sample_buf[j][i] & mask);
s->sample_buf[j][i] >>= s->extra_bits;
}
@ -415,8 +427,7 @@ static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
}
// apply lpc and entropy coding to audio samples
for (i = 0; i < s->avctx->channels; i++) {
for (i = 0; i < channels; i++) {
alac_linear_predictor(s, i);
// TODO: determine when this will actually help. for now it's not used.
@ -425,12 +436,39 @@ static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
for (j = s->frame_size - 1; j > 0; j--)
s->predictor_buf[j] -= s->predictor_buf[j - 1];
}
alac_entropy_coder(s);
}
}
put_bits(pb, 3, 7);
}
static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
uint8_t * const *samples)
{
PutBitContext *pb = &s->pbctx;
const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1];
const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1];
int ch, element, sce, cpe;
init_put_bits(pb, avpkt->data, avpkt->size);
ch = element = sce = cpe = 0;
while (ch < s->avctx->channels) {
if (ch_elements[element] == TYPE_CPE) {
write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]],
samples[ch_map[ch + 1]]);
cpe++;
ch += 2;
} else {
write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL);
sce++;
ch++;
}
element++;
}
put_bits(pb, 3, TYPE_END);
flush_put_bits(pb);
return put_bits_count(pb) >> 3;
}
@ -458,14 +496,6 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;
/* TODO: Correctly implement multi-channel ALAC.
It is similar to multi-channel AAC, in that it has a series of
single-channel (SCE), channel-pair (CPE), and LFE elements. */
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "only mono or stereo input is currently supported\n");
return AVERROR_PATCHWELCOME;
}
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
if (avctx->bits_per_raw_sample != 24)
av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n");
@ -595,8 +625,6 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
s->verbatim = 1;
s->extra_bits = 0;
}
s->write_sample_size = avctx->bits_per_raw_sample - s->extra_bits +
avctx->channels - 1;
out_bytes = write_frame(s, avpkt, frame->extended_data);
@ -604,7 +632,6 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
/* frame too large. use verbatim mode */
s->verbatim = 1;
s->extra_bits = 0;
s->write_sample_size = avctx->bits_per_raw_sample + avctx->channels - 1;
out_bytes = write_frame(s, avpkt, frame->extended_data);
}
@ -622,6 +649,7 @@ AVCodec ff_alac_encoder = {
.encode2 = alac_encode_frame,
.close = alac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.channel_layouts = ff_alac_channel_layouts,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },