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Merge remote-tracking branch 'qatar/master'

Browse Source 
* qatar/master:
  doc/APIchanges: add an entry for codec descriptors.
  vorbisenc: set AVCodecContext.bit_rate to 0
  vorbisenc: fix quality parameter
  FATE: add ALAC encoding tests
  lpc: fix alignment of windowed samples for odd maximum LPC order
  alacenc: use s16p sample format as input
  alacenc: remove unneeded sample_fmt check
  alacenc: fix max_frame_size calculation for the final frame
  adpcm_swf: Use correct sample offsets when using trellis.
  rtmp: support strict rtmp servers
  mjpegdec: support AVRn interlaced
  x86: remove FASTDIV inline asm

Conflicts:
	doc/APIchanges
	libavcodec/mjpegdec.c
	libavcodec/vorbisenc.c
	libavutil/x86/intmath.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2012-08-23 14:23:08 +02:00
parent c2271fa7f9 7b44061f4b
commit 104f42e694
10 changed files with 71 additions and 43 deletions
Download Patch File Download Diff File Expand all files Collapse all files

7
libavcodec/adpcmenc.c
View File

@@ -616,10 +616,11 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
if (avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
adpcm_compress_trellis(avctx, samples + 2, buf, &c->status[0], n);
adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
&c->status[0], n);
if (avctx->channels == 2)
adpcm_compress_trellis(avctx, samples + 3, buf + n,
&c->status[1], n);
adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
buf + n, &c->status[1], n);
for (i = 0; i < n; i++) {
put_bits(&pb, 4, buf[i]);
if (avctx->channels == 2)

32
libavcodec/alacenc.c
View File

@@ -78,17 +78,15 @@ typedef struct AlacEncodeContext {
} AlacEncodeContext;
static void init_sample_buffers(AlacEncodeContext *s,
const int16_t *input_samples)
static void init_sample_buffers(AlacEncodeContext *s, int16_t **input_samples)
{
int ch, i;
for (ch = 0; ch < s->avctx->channels; ch++) {
const int16_t *sptr = input_samples + ch;
for (i = 0; i < s->frame_size; i++) {
s->sample_buf[ch][i] = *sptr;
sptr += s->avctx->channels;
}
int32_t *bptr = s->sample_buf[ch];
const int16_t *sptr = input_samples[ch];
for (i = 0; i < s->frame_size; i++)
bptr[i] = sptr[i];
}
}
@@ -347,8 +345,7 @@ static void alac_entropy_coder(AlacEncodeContext *s)
}
}
static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
const int16_t *samples)
static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, int16_t **samples)
{
int i, j;
int prediction_type = 0;
@@ -358,8 +355,10 @@ static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
if (s->verbatim) {
write_frame_header(s);
for (i = 0; i < s->frame_size * s->avctx->channels; i++)
put_sbits(pb, 16, *samples++);
/* samples are channel-interleaved in verbatim mode */
for (i = 0; i < s->frame_size; i++)
for (j = 0; j < s->avctx->channels; j++)
put_sbits(pb, 16, samples[j][i]);
} else {
init_sample_buffers(s, samples);
write_frame_header(s);
@@ -426,11 +425,6 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;
if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
return -1;
}
/* 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. */
@@ -542,11 +536,11 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
{
AlacEncodeContext *s = avctx->priv_data;
int out_bytes, max_frame_size, ret;
const int16_t *samples = (const int16_t *)frame->data[0];
int16_t **samples = (int16_t **)frame->extended_data;
s->frame_size = frame->nb_samples;
if (avctx->frame_size < DEFAULT_FRAME_SIZE)
if (frame->nb_samples < DEFAULT_FRAME_SIZE)
max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,
DEFAULT_SAMPLE_SIZE);
else
@@ -580,7 +574,7 @@ AVCodec ff_alac_encoder = {
.encode2 = alac_encode_frame,
.close = alac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};

15
libavcodec/lpc.c
View File

@@ -179,11 +179,9 @@ int ff_lpc_calc_coefs(LPCContext *s,
}
if (lpc_type == FF_LPC_TYPE_LEVINSON) {
double *windowed_samples = s->windowed_samples + max_order;
s->lpc_apply_welch_window(samples, blocksize, s->windowed_samples);
s->lpc_apply_welch_window(samples, blocksize, windowed_samples);
s->lpc_compute_autocorr(windowed_samples, blocksize, max_order, autoc);
s->lpc_compute_autocorr(s->windowed_samples, blocksize, max_order, autoc);
compute_lpc_coefs(autoc, max_order, &lpc[0][0], MAX_LPC_ORDER, 0, 1);
@@ -252,10 +250,11 @@ av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order,
s->lpc_type = lpc_type;
if (lpc_type == FF_LPC_TYPE_LEVINSON) {
s->windowed_samples = av_mallocz((blocksize + max_order + 2) *
sizeof(*s->windowed_samples));
if (!s->windowed_samples)
s->windowed_buffer = av_mallocz((blocksize + 2 + FFALIGN(max_order, 4)) *
sizeof(*s->windowed_samples));
if (!s->windowed_buffer)
return AVERROR(ENOMEM);
s->windowed_samples = s->windowed_buffer + FFALIGN(max_order, 4);
} else {
s->windowed_samples = NULL;
}
@@ -271,5 +270,5 @@ av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order,
av_cold void ff_lpc_end(LPCContext *s)
{
av_freep(&s->windowed_samples);
av_freep(&s->windowed_buffer);
}

1
libavcodec/lpc.h
View File

@@ -51,6 +51,7 @@ typedef struct LPCContext {
int blocksize;
int max_order;
enum FFLPCType lpc_type;
double *windowed_buffer;
double *windowed_samples;
/**

12
libavcodec/mjpegdec.c
View File

@@ -1235,6 +1235,7 @@ int ff_mjpeg_decode_sos(MJpegDecodeContext *s, const uint8_t *mb_bitmask,
/* mjpeg-b can have padding bytes between sos and image data, skip them */
for (i = s->mjpb_skiptosod; i > 0; i--)
skip_bits(&s->gb, 8);
next_field:
for (i = 0; i < nb_components; i++)
s->last_dc[i] = 1024;
@@ -1271,11 +1272,14 @@ next_field:
return ret;
}
}
if(s->interlaced && get_bits_left(&s->gb) > 32 && show_bits(&s->gb, 8) == 0xFF) {
GetBitContext bak= s->gb;
if (s->interlaced &&
get_bits_left(&s->gb) > 32 &&
show_bits(&s->gb, 8) == 0xFF) {
GetBitContext bak = s->gb;
align_get_bits(&bak);
if(show_bits(&bak, 16) == 0xFFD1) {
av_log(s->avctx, AV_LOG_DEBUG, "AVRn ingterlaced picture\n");
if (show_bits(&bak, 16) == 0xFFD1) {
av_log(s->avctx, AV_LOG_DEBUG, "AVRn interlaced picture marker found\n");
s->gb = bak;
skip_bits(&s->gb, 16);
s->bottom_field ^= 1;

3
libavcodec/vorbisenc.c
View File

@@ -1177,8 +1177,9 @@ static av_cold int vorbis_encode_init(AVCodecContext *avccontext)
if ((ret = create_vorbis_context(venc, avccontext)) < 0)
goto error;
avccontext->bit_rate = 0;
if (avccontext->flags & CODEC_FLAG_QSCALE)
venc->quality = avccontext->global_quality / (float)FF_QP2LAMBDA / 10.;
venc->quality = avccontext->global_quality / (float)FF_QP2LAMBDA;
else
venc->quality = 8;
venc->quality *= venc->quality;