falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
Browse Source

Merge remote-tracking branch 'qatar/master' 

* qatar/master:
  ac3enc: use correct alignment and length in channel coupling dsp functions.
  ffmpeg: don't abuse a global for passing framerate from input to output
  ffmpeg: don't abuse a global for passing channels from input to output
  ffmpeg: don't abuse a global for passing samplerate from input to output
  ARM: update ff_h264_idct8_add4_neon for 4:4:4 changes
  swscale: use SwsContext for av_log when available
  swscale: Remove HAVE_MMX from files that are only compiled with MMX enabled.
  swscale: Fix compilation with --disable-mmx2.

Conflicts:
	ffmpeg.c
	libswscale/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
tags/n0.8
Michael Niedermayer 14 years ago
parent
a0bafaabb0 b203f65451
commit
7a02527b05
11 changed files with 92 additions and 71 deletions
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. +35
    -26
      ffmpeg.c
  2. +13
    -11
      libavcodec/ac3enc_template.c
  3. +1
    -2
      libavcodec/arm/h264dsp_init_arm.c
  4. +24
    -17
      libavcodec/arm/h264idct_neon.S
  5. +4
    -4
      libswscale/utils.c
  6. +1
    -1
      libswscale/x86/rgb2rgb.c
  7. +2
    -0
      libswscale/x86/swscale_mmx.c
  8. +4
    -2
      libswscale/x86/yuv2rgb_mmx.c
  9. +1
    -1
      tests/fate2.mak
  10. +6
    -6
      tests/lavf-regression.sh
  11. +1
    -1
      tests/regression-funcs.sh

+ 35
- 26
ffmpeg.c View File

@@ -173,12 +173,12 @@ static char *vfilters = NULL;
#endif

static int intra_only = 0;
static int audio_sample_rate = 44100;
static int audio_sample_rate = 0;
static int64_t channel_layout = 0;
#define QSCALE_NONE -99999
static float audio_qscale = QSCALE_NONE;
static int audio_disable = 0;
static int audio_channels = 1;
static int audio_channels = 0;
static char *audio_codec_name = NULL;
static unsigned int audio_codec_tag = 0;
static char *audio_language = NULL;
@@ -283,6 +283,7 @@ typedef struct AVOutputStream {
int resample_height;
int resample_width;
int resample_pix_fmt;
AVRational frame_rate;

float frame_aspect_ratio;

@@ -2267,6 +2268,17 @@ static int transcode(AVFormatContext **output_files,
if(!ost->fifo)
goto fail;
ost->reformat_pair = MAKE_SFMT_PAIR(AV_SAMPLE_FMT_NONE,AV_SAMPLE_FMT_NONE);
if (!codec->sample_rate) {
codec->sample_rate = icodec->sample_rate;
if (icodec->lowres)
codec->sample_rate >>= icodec->lowres;
}
choose_sample_rate(ost->st, codec->codec);
codec->time_base = (AVRational){1, codec->sample_rate};
if (!codec->channels)
codec->channels = icodec->channels;
if (av_get_channel_layout_nb_channels(codec->channel_layout) != codec->channels)
codec->channel_layout = 0;
ost->audio_resample = codec->sample_rate != icodec->sample_rate || audio_sync_method > 1;
icodec->request_channels = codec->channels;
ist->decoding_needed = 1;
@@ -2292,6 +2304,14 @@ static int transcode(AVFormatContext **output_files,
ost->encoding_needed = 1;
ist->decoding_needed = 1;

if (!ost->frame_rate.num)
ost->frame_rate = ist->st->r_frame_rate.num ? ist->st->r_frame_rate : (AVRational){25,1};
if (codec->codec && codec->codec->supported_framerates && !force_fps) {
int idx = av_find_nearest_q_idx(ost->frame_rate, codec->codec->supported_framerates);
ost->frame_rate = codec->codec->supported_framerates[idx];
}
codec->time_base = (AVRational){ost->frame_rate.den, ost->frame_rate.num};

#if CONFIG_AVFILTER
if (configure_video_filters(ist, ost)) {
fprintf(stderr, "Error opening filters!\n");
@@ -3369,16 +3389,9 @@ static int opt_input_file(const char *opt, const char *filename)
input_codecs[nb_input_codecs-1] = avcodec_find_decoder(dec->codec_id);
set_context_opts(dec, avcodec_opts[AVMEDIA_TYPE_AUDIO], AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM, input_codecs[nb_input_codecs-1]);
channel_layout = dec->channel_layout;
audio_channels = dec->channels;
audio_sample_rate = dec->sample_rate;
audio_sample_fmt = dec->sample_fmt;
if(audio_disable)
st->discard= AVDISCARD_ALL;
/* Note that av_find_stream_info can add more streams, and we
* currently have no chance of setting up lowres decoding
* early enough for them. */
if (dec->lowres)
audio_sample_rate >>= dec->lowres;
break;
case AVMEDIA_TYPE_VIDEO:
input_codecs[nb_input_codecs-1] = avcodec_find_decoder_by_name(video_codec_name);
@@ -3408,9 +3421,6 @@ static int opt_input_file(const char *opt, const char *filename)

(float)rfps / rfps_base, rfps, rfps_base);
}
/* update the current frame rate to match the stream frame rate */
frame_rate.num = rfps;
frame_rate.den = rfps_base;

if(video_disable)
st->discard= AVDISCARD_ALL;
@@ -3445,6 +3455,9 @@ static int opt_input_file(const char *opt, const char *filename)

video_channel = 0;
top_field_first = -1;
frame_rate = (AVRational){0, 0};
audio_sample_rate = 0;
audio_channels = 0;

av_freep(&video_codec_name);
av_freep(&audio_codec_name);
@@ -3555,16 +3568,12 @@ static void new_video_stream(AVFormatContext *oc, int file_idx)
} else {
const char *p;
int i;
AVRational fps= frame_rate.num ? frame_rate : (AVRational){25,1};

if (frame_rate.num)
ost->frame_rate = frame_rate;
video_enc->codec_id = codec_id;
set_context_opts(video_enc, avcodec_opts[AVMEDIA_TYPE_VIDEO], AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, codec);

if (codec && codec->supported_framerates && !force_fps)
fps = codec->supported_framerates[av_find_nearest_q_idx(fps, codec->supported_framerates)];
video_enc->time_base.den = fps.num;
video_enc->time_base.num = fps.den;

video_enc->width = frame_width;
video_enc->height = frame_height;
video_enc->pix_fmt = frame_pix_fmt;
@@ -3691,8 +3700,6 @@ static void new_audio_stream(AVFormatContext *oc, int file_idx)
}
if (audio_stream_copy) {
st->stream_copy = 1;
audio_enc->channels = audio_channels;
audio_enc->sample_rate = audio_sample_rate;
} else {
audio_enc->codec_id = codec_id;
set_context_opts(audio_enc, avcodec_opts[AVMEDIA_TYPE_AUDIO], AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, codec);
@@ -3701,16 +3708,14 @@ static void new_audio_stream(AVFormatContext *oc, int file_idx)
audio_enc->flags |= CODEC_FLAG_QSCALE;
audio_enc->global_quality = st->quality = FF_QP2LAMBDA * audio_qscale;
}
audio_enc->channels = audio_channels;
if (audio_channels)
audio_enc->channels = audio_channels;
audio_enc->sample_fmt = audio_sample_fmt;
audio_enc->sample_rate = audio_sample_rate;
if (audio_sample_rate)
audio_enc->sample_rate = audio_sample_rate;
audio_enc->channel_layout = channel_layout;
if (av_get_channel_layout_nb_channels(channel_layout) != audio_channels)
audio_enc->channel_layout = 0;
choose_sample_fmt(st, codec);
choose_sample_rate(st, codec);
}
audio_enc->time_base= (AVRational){1, audio_sample_rate};
if (audio_language) {
av_dict_set(&st->metadata, "language", audio_language, 0);
av_freep(&audio_language);
@@ -3983,6 +3988,10 @@ static int opt_output_file(const char *opt, const char *filename)

set_context_opts(oc, avformat_opts, AV_OPT_FLAG_ENCODING_PARAM, NULL);

frame_rate = (AVRational){0, 0};
audio_sample_rate = 0;
audio_channels = 0;

av_freep(&forced_key_frames);
uninit_opts();
init_opts();


+ 13
- 11
libavcodec/ac3enc_template.c View File

@@ -134,36 +134,38 @@ void AC3_NAME(apply_channel_coupling)(AC3EncodeContext *s)
LOCAL_ALIGNED_16(int32_t, fixed_cpl_coords, [AC3_MAX_BLOCKS], [AC3_MAX_CHANNELS][16]);
int blk, ch, bnd, i, j;
CoefSumType energy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][16] = {{{0}}};
int num_cpl_coefs = s->num_cpl_subbands * 12;
int cpl_start, num_cpl_coefs;

memset(cpl_coords, 0, AC3_MAX_BLOCKS * sizeof(*cpl_coords));
memset(fixed_cpl_coords, 0, AC3_MAX_BLOCKS * sizeof(*fixed_cpl_coords));

/* align start to 16-byte boundary. align length to multiple of 32.
note: coupling start bin % 4 will always be 1 */
cpl_start = s->start_freq[CPL_CH] - 1;
num_cpl_coefs = FFALIGN(s->num_cpl_subbands * 12 + 1, 32);
cpl_start = FFMIN(256, cpl_start + num_cpl_coefs) - num_cpl_coefs;

/* calculate coupling channel from fbw channels */
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
CoefType *cpl_coef = &block->mdct_coef[CPL_CH][s->start_freq[CPL_CH]];
CoefType *cpl_coef = &block->mdct_coef[CPL_CH][cpl_start];
if (!block->cpl_in_use)
continue;
memset(cpl_coef-1, 0, (num_cpl_coefs+4) * sizeof(*cpl_coef));
memset(cpl_coef, 0, num_cpl_coefs * sizeof(*cpl_coef));
for (ch = 1; ch <= s->fbw_channels; ch++) {
CoefType *ch_coef = &block->mdct_coef[ch][s->start_freq[CPL_CH]];
CoefType *ch_coef = &block->mdct_coef[ch][cpl_start];
if (!block->channel_in_cpl[ch])
continue;
for (i = 0; i < num_cpl_coefs; i++)
cpl_coef[i] += ch_coef[i];
}
/* note: coupling start bin % 4 will always be 1 and num_cpl_coefs
will always be a multiple of 12, so we need to subtract 1 from
the start and add 4 to the length when using optimized
functions which require 16-byte alignment. */

/* coefficients must be clipped to +/- 1.0 in order to be encoded */
s->dsp.vector_clipf(cpl_coef-1, cpl_coef-1, -1.0f, 1.0f, num_cpl_coefs+4);
s->dsp.vector_clipf(cpl_coef, cpl_coef, -1.0f, 1.0f, num_cpl_coefs);

/* scale coupling coefficients from float to 24-bit fixed-point */
s->ac3dsp.float_to_fixed24(&block->fixed_coef[CPL_CH][s->start_freq[CPL_CH]-1],
cpl_coef-1, num_cpl_coefs+4);
s->ac3dsp.float_to_fixed24(&block->fixed_coef[CPL_CH][cpl_start],
cpl_coef, num_cpl_coefs);
}

/* calculate energy in each band in coupling channel and each fbw channel */


+ 1
- 2
libavcodec/arm/h264dsp_init_arm.c View File

@@ -122,8 +122,7 @@ static void ff_h264dsp_init_neon(H264DSPContext *c, const int bit_depth)
c->h264_idct_dc_add = ff_h264_idct_dc_add_neon;
c->h264_idct_add16 = ff_h264_idct_add16_neon;
c->h264_idct_add16intra = ff_h264_idct_add16intra_neon;
//FIXME: reenable when asm is updated.
//c->h264_idct_add8 = ff_h264_idct_add8_neon;
c->h264_idct_add8 = ff_h264_idct_add8_neon;
c->h264_idct8_add = ff_h264_idct8_add_neon;
c->h264_idct8_dc_add = ff_h264_idct8_dc_add_neon;
c->h264_idct8_add4 = ff_h264_idct8_add4_neon;


+ 24
- 17
libavcodec/arm/h264idct_neon.S View File

@@ -148,24 +148,27 @@ function ff_h264_idct_add8_neon, export=1
add r5, r1, #16*4
add r1, r2, #16*32
mov r2, r3
mov r3, r1
ldr r6, [sp, #32]
movrel r7, scan8+16
mov ip, #7
1: ldrb r8, [r7], #1
ldr r0, [r5], #4
mov r12, #0
1: ldrb r8, [r7, r12]
ldr r0, [r5, r12, lsl #2]
ldrb r8, [r6, r8]
tst ip, #4
addne r0, r0, r4
addeq r0, r0, r9
add r0, r0, r4
add r1, r3, r12, lsl #5
cmp r8, #0
ldrsh r8, [r1]
adrne lr, ff_h264_idct_add_neon
adreq lr, ff_h264_idct_dc_add_neon
cmpeq r8, #0
blxne lr
subs ip, ip, #1
add r1, r1, #32
bge 1b
add r12, r12, #1
cmp r12, #4
moveq r12, #16
moveq r4, r9
cmp r12, #20
blt 1b
pop {r4-r10,pc}
endfunc

@@ -374,11 +377,15 @@ function ff_h264_idct8_add4_neon, export=1
endfunc

.section .rodata
scan8: .byte 4+1*8, 5+1*8, 4+2*8, 5+2*8
.byte 6+1*8, 7+1*8, 6+2*8, 7+2*8
.byte 4+3*8, 5+3*8, 4+4*8, 5+4*8
.byte 6+3*8, 7+3*8, 6+4*8, 7+4*8
.byte 1+1*8, 2+1*8
.byte 1+2*8, 2+2*8
.byte 1+4*8, 2+4*8
.byte 1+5*8, 2+5*8
scan8: .byte 4+ 1*8, 5+ 1*8, 4+ 2*8, 5+ 2*8
.byte 6+ 1*8, 7+ 1*8, 6+ 2*8, 7+ 2*8
.byte 4+ 3*8, 5+ 3*8, 4+ 4*8, 5+ 4*8
.byte 6+ 3*8, 7+ 3*8, 6+ 4*8, 7+ 4*8
.byte 4+ 6*8, 5+ 6*8, 4+ 7*8, 5+ 7*8
.byte 6+ 6*8, 7+ 6*8, 6+ 7*8, 7+ 7*8
.byte 4+ 8*8, 5+ 8*8, 4+ 9*8, 5+ 9*8
.byte 6+ 8*8, 7+ 8*8, 6+ 9*8, 7+ 9*8
.byte 4+11*8, 5+11*8, 4+12*8, 5+12*8
.byte 6+11*8, 7+11*8, 6+12*8, 7+12*8
.byte 4+13*8, 5+13*8, 4+14*8, 5+14*8
.byte 6+13*8, 7+13*8, 6+14*8, 7+14*8

+ 4
- 4
libswscale/utils.c View File

@@ -790,11 +790,11 @@ int sws_init_context(SwsContext *c, SwsFilter *srcFilter, SwsFilter *dstFilter)
unscaled = (srcW == dstW && srcH == dstH);

if (!isSupportedIn(srcFormat)) {
av_log(NULL, AV_LOG_ERROR, "swScaler: %s is not supported as input pixel format\n", av_get_pix_fmt_name(srcFormat));
av_log(c, AV_LOG_ERROR, "%s is not supported as input pixel format\n", av_get_pix_fmt_name(srcFormat));
return AVERROR(EINVAL);
}
if (!isSupportedOut(dstFormat)) {
av_log(NULL, AV_LOG_ERROR, "swScaler: %s is not supported as output pixel format\n", av_get_pix_fmt_name(dstFormat));
av_log(c, AV_LOG_ERROR, "%s is not supported as output pixel format\n", av_get_pix_fmt_name(dstFormat));
return AVERROR(EINVAL);
}

@@ -810,12 +810,12 @@ int sws_init_context(SwsContext *c, SwsFilter *srcFilter, SwsFilter *dstFilter)
|SWS_SPLINE
|SWS_BICUBLIN);
if(!i || (i & (i-1))) {
av_log(NULL, AV_LOG_ERROR, "swScaler: Exactly one scaler algorithm must be chosen\n");
av_log(c, AV_LOG_ERROR, "Exactly one scaler algorithm must be chosen\n");
return AVERROR(EINVAL);
}
/* sanity check */
if (srcW<4 || srcH<1 || dstW<8 || dstH<1) { //FIXME check if these are enough and try to lowwer them after fixing the relevant parts of the code
av_log(NULL, AV_LOG_ERROR, "swScaler: %dx%d -> %dx%d is invalid scaling dimension\n",
av_log(c, AV_LOG_ERROR, "%dx%d -> %dx%d is invalid scaling dimension\n",
srcW, srcH, dstW, dstH);
return AVERROR(EINVAL);
}


+ 1
- 1
libswscale/x86/rgb2rgb.c View File

@@ -127,7 +127,7 @@ void rgb2rgb_init_x86(void)
{
int cpu_flags = av_get_cpu_flags();

if (HAVE_MMX && cpu_flags & AV_CPU_FLAG_MMX)
if (cpu_flags & AV_CPU_FLAG_MMX)
rgb2rgb_init_MMX();
if (HAVE_AMD3DNOW && cpu_flags & AV_CPU_FLAG_3DNOW)
rgb2rgb_init_3DNOW();


+ 2
- 0
libswscale/x86/swscale_mmx.c View File

@@ -182,6 +182,8 @@ void ff_sws_init_swScale_mmx(SwsContext *c)

if (cpu_flags & AV_CPU_FLAG_MMX)
sws_init_swScale_MMX(c);
#if HAVE_MMX2
if (cpu_flags & AV_CPU_FLAG_MMX2)
sws_init_swScale_MMX2(c);
#endif
}

+ 4
- 2
libswscale/x86/yuv2rgb_mmx.c View File

@@ -72,14 +72,16 @@ SwsFunc ff_yuv2rgb_init_mmx(SwsContext *c)
c->srcFormat != PIX_FMT_YUVA420P)
return NULL;

if (HAVE_MMX2 && cpu_flags & AV_CPU_FLAG_MMX2) {
#if HAVE_MMX2
if (cpu_flags & AV_CPU_FLAG_MMX2) {
switch (c->dstFormat) {
case PIX_FMT_RGB24: return yuv420_rgb24_MMX2;
case PIX_FMT_BGR24: return yuv420_bgr24_MMX2;
}
}
#endif

if (HAVE_MMX && cpu_flags & AV_CPU_FLAG_MMX) {
if (cpu_flags & AV_CPU_FLAG_MMX) {
switch (c->dstFormat) {
case PIX_FMT_RGB32:
if (c->srcFormat == PIX_FMT_YUVA420P) {


+ 1
- 1
tests/fate2.mak View File

@@ -165,7 +165,7 @@ fate-wmapro-2ch: CMP = oneoff
fate-wmapro-2ch: REF = $(SAMPLES)/wmapro/Beethovens_9th-1_small.pcm

FATE_TESTS += fate-ansi
fate-ansi: CMD = framecrc -i $(SAMPLES)/ansi/TRE-IOM5.ANS -pix_fmt rgb24
fate-ansi: CMD = framecrc -ar 44100 -i $(SAMPLES)/ansi/TRE-IOM5.ANS -pix_fmt rgb24

FATE_TESTS += fate-wmv8-drm
# discard last packet to avoid fails due to overread of VC-1 decoder


+ 6
- 6
tests/lavf-regression.sh View File

@@ -14,7 +14,7 @@ eval do_$test=y
do_lavf()
{
file=${outfile}lavf.$1
do_ffmpeg $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 $2
do_ffmpeg $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -ar 44100 -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 $2
do_ffmpeg_crc $file $DEC_OPTS -i $target_path/$file $3
}

@@ -39,8 +39,8 @@ do_image_formats()
do_audio_only()
{
file=${outfile}lavf.$1
do_ffmpeg $file $DEC_OPTS $2 -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 $3
do_ffmpeg_crc $file $DEC_OPTS -i $target_path/$file
do_ffmpeg $file $DEC_OPTS $2 -ar 44100 -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 $3
do_ffmpeg_crc $file $DEC_OPTS $4 -i $target_path/$file
}

rm -f "$logfile"
@@ -55,7 +55,7 @@ fi

if [ -n "$do_rm" ] ; then
file=${outfile}lavf.rm
do_ffmpeg $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 -acodec ac3_fixed
do_ffmpeg $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -ar 44100 -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 -acodec ac3_fixed
# broken
#do_ffmpeg_crc $file -i $target_path/$file
fi
@@ -181,11 +181,11 @@ do_audio_only wav
fi

if [ -n "$do_alaw" ] ; then
do_audio_only al
do_audio_only al "" "" "-ar 44100"
fi

if [ -n "$do_mulaw" ] ; then
do_audio_only ul
do_audio_only ul "" "" "-ar 44100"
fi

if [ -n "$do_au" ] ; then


+ 1
- 1
tests/regression-funcs.sh View File

@@ -114,7 +114,7 @@ do_video_encoding()
do_audio_encoding()
{
file=${outfile}$1
do_ffmpeg $file $DEC_OPTS -ac 2 -f s16le -i $pcm_src -ab 128k $ENC_OPTS $2
do_ffmpeg $file $DEC_OPTS -ac 2 -ar 44100 -f s16le -i $pcm_src -ab 128k $ENC_OPTS $2
}

do_audio_decoding()


Write Preview
Loading…
Cancel
Save