wrapper/FFmpeg
1
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity Actions

Merge remote-tracking branch 'qatar/master'

Browse source 
* qatar/master:
  lavr: add x86-optimized functions for mixing 1-to-2 s16p with flt coeffs
  lavr: add x86-optimized functions for mixing 1-to-2 fltp with flt coeffs
  Add Dolby/DPLII downmix support to libavresample
  vorbisdec: replace div/mod in loop with a counter
  fate: vorbis: add 5.1 surround test
  rtpenc: Allow requesting H264 RTP packetization mode 0
  configure: Sort the library listings in the help text alphabetically
  dwt: remove variable-length arrays
  RTMPT protocol support
  http: Properly handle chunked transfer-encoding for replies to post data
  http: Fail reading if the connection has gone away
  amr: Mark an array const
  amr: More space cleanup
  rtpenc: Fix memory leaks in the muxer open function

Conflicts:
	Changelog
	configure
	doc/APIchanges
	libavformat/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-06-18 20:05:32 +02:00
commit 82edf6727f
33 changed files with 589 additions and 112 deletions
Download patch file Download diff file Expand all files Collapse all files

3
libavresample/audio_mix.c
View file

@ -320,7 +320,8 @@ int ff_audio_mix_init(AVAudioResampleContext *avr)
avr->center_mix_level,
avr->surround_mix_level,
avr->lfe_mix_level, 1, matrix_dbl,
avr->in_channels);
avr->in_channels,
avr->matrix_encoding);
if (ret < 0) {
av_free(matrix_dbl);
return ret;

60
libavresample/audio_mix_matrix.c
View file

@ -54,6 +54,8 @@
#define SURROUND_DIRECT_LEFT 33
#define SURROUND_DIRECT_RIGHT 34
#define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
static av_always_inline int even(uint64_t layout)
{
return (!layout || (layout & (layout - 1)));
@ -83,14 +85,21 @@ static int sane_layout(uint64_t layout)
int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, int normalize,
double *matrix_out, int stride)
double *matrix_out, int stride,
enum AVMatrixEncoding matrix_encoding)
{
int i, j, out_i, out_j;
double matrix[64][64] = {{0}};
int64_t unaccounted = in_layout & ~out_layout;
int64_t unaccounted;
double maxcoef = 0;
int in_channels, out_channels;
if ((out_layout & AV_CH_LAYOUT_STEREO_DOWNMIX) == AV_CH_LAYOUT_STEREO_DOWNMIX) {
out_layout = AV_CH_LAYOUT_STEREO;
}
unaccounted = in_layout & ~out_layout;
in_channels = av_get_channel_layout_nb_channels( in_layout);
out_channels = av_get_channel_layout_nb_channels(out_layout);
@ -140,8 +149,19 @@ int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
matrix[SIDE_LEFT ][BACK_CENTER] += M_SQRT1_2;
matrix[SIDE_RIGHT][BACK_CENTER] += M_SQRT1_2;
} else if (out_layout & AV_CH_FRONT_LEFT) {
matrix[FRONT_LEFT ][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY ||
matrix_encoding == AV_MATRIX_ENCODING_DPLII) {
if (unaccounted & (AV_CH_BACK_LEFT | AV_CH_SIDE_LEFT)) {
matrix[FRONT_LEFT ][BACK_CENTER] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
} else {
matrix[FRONT_LEFT ][BACK_CENTER] -= surround_mix_level;
matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level;
}
} else {
matrix[FRONT_LEFT ][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
}
} else if (out_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
} else
@ -163,8 +183,20 @@ int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
matrix[SIDE_RIGHT][BACK_RIGHT] += 1.0;
}
} else if (out_layout & AV_CH_FRONT_LEFT) {
matrix[FRONT_LEFT ][BACK_LEFT ] += surround_mix_level;
matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level;
if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY) {
matrix[FRONT_LEFT ][BACK_LEFT ] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_LEFT ][BACK_RIGHT] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level * M_SQRT1_2;
} else if (matrix_encoding == AV_MATRIX_ENCODING_DPLII) {
matrix[FRONT_LEFT ][BACK_LEFT ] -= surround_mix_level * SQRT3_2;
matrix[FRONT_LEFT ][BACK_RIGHT] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level * SQRT3_2;
} else {
matrix[FRONT_LEFT ][BACK_LEFT ] += surround_mix_level;
matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level;
}
} else if (out_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][BACK_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_CENTER][BACK_RIGHT] += surround_mix_level * M_SQRT1_2;
@ -187,8 +219,20 @@ int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
matrix[BACK_CENTER][SIDE_LEFT ] += M_SQRT1_2;
matrix[BACK_CENTER][SIDE_RIGHT] += M_SQRT1_2;
} else if (out_layout & AV_CH_FRONT_LEFT) {
matrix[FRONT_LEFT ][SIDE_LEFT ] += surround_mix_level;
matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level;
if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY) {
matrix[FRONT_LEFT ][SIDE_LEFT ] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_LEFT ][SIDE_RIGHT] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level * M_SQRT1_2;
} else if (matrix_encoding == AV_MATRIX_ENCODING_DPLII) {
matrix[FRONT_LEFT ][SIDE_LEFT ] -= surround_mix_level * SQRT3_2;
matrix[FRONT_LEFT ][SIDE_RIGHT] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level * SQRT3_2;
} else {
matrix[FRONT_LEFT ][SIDE_LEFT ] += surround_mix_level;
matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level;
}
} else if (out_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_CENTER][SIDE_RIGHT] += surround_mix_level * M_SQRT1_2;

3
libavresample/avresample.h
View file

@ -131,12 +131,13 @@ void avresample_free(AVAudioResampleContext **avr);
* the weight of input channel i in output channel o.
* @param stride distance between adjacent input channels in the
* matrix array
* @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, int normalize, double *matrix,
int stride);
int stride, enum AVMatrixEncoding matrix_encoding);
/**
* Get the current channel mixing matrix.

1
libavresample/internal.h
View file

@ -70,6 +70,7 @@ struct AVAudioResampleContext {
AudioConvert *ac_out; /**< output sample format conversion context */
ResampleContext *resample; /**< resampling context */
AudioMix *am; /**< channel mixing context */
enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
};
#endif /* AVRESAMPLE_INTERNAL_H */

4
libavresample/options.c
View file

@ -52,6 +52,10 @@ static const AVOption options[] = {
{ "phase_shift", "Resampling Phase Shift", OFFSET(phase_shift), AV_OPT_TYPE_INT, { 10 }, 0, 30, /* ??? */ PARAM },
{ "linear_interp", "Use Linear Interpolation", OFFSET(linear_interp), AV_OPT_TYPE_INT, { 0 }, 0, 1, PARAM },
{ "cutoff", "Cutoff Frequency Ratio", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, { 0.8 }, 0.0, 1.0, PARAM },
{ "matrix_encoding", "Matrixed Stereo Encoding", OFFSET(matrix_encoding), AV_OPT_TYPE_INT, { AV_MATRIX_ENCODING_NONE}, AV_MATRIX_ENCODING_NONE, AV_MATRIX_ENCODING_NB-1, PARAM, "matrix_encoding" },
{ "none", "None", 0, AV_OPT_TYPE_CONST, { AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "dolby", "Dolby", 0, AV_OPT_TYPE_CONST, { AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "dplii", "Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ NULL },
};

2
libavresample/version.h
View file

@ -21,7 +21,7 @@
#define LIBAVRESAMPLE_VERSION_MAJOR 0
#define LIBAVRESAMPLE_VERSION_MINOR 0
#define LIBAVRESAMPLE_VERSION_MICRO 2
#define LIBAVRESAMPLE_VERSION_MICRO 3
#define LIBAVRESAMPLE_VERSION_INT AV_VERSION_INT(LIBAVRESAMPLE_VERSION_MAJOR, \
LIBAVRESAMPLE_VERSION_MINOR, \

81
libavresample/x86/audio_mix.asm
View file

@ -150,3 +150,84 @@ cglobal mix_2_to_1_s16p_q8, 3,4,6, src, matrix, len, src1
sub lend, mmsize/2
jg .loop
REP_RET
;-----------------------------------------------------------------------------
; void ff_mix_1_to_2_fltp_flt(float **src, float **matrix, int len,
; int out_ch, int in_ch);
;-----------------------------------------------------------------------------
%macro MIX_1_TO_2_FLTP_FLT 0
cglobal mix_1_to_2_fltp_flt, 3,5,4, src0, matrix0, len, src1, matrix1
mov src1q, [src0q+gprsize]
mov src0q, [src0q]
sub src1q, src0q
mov matrix1q, [matrix0q+gprsize]
mov matrix0q, [matrix0q]
VBROADCASTSS m2, [matrix0q]
VBROADCASTSS m3, [matrix1q]
ALIGN 16
.loop:
mova m0, [src0q]
mulps m1, m0, m3
mulps m0, m0, m2
mova [src0q ], m0
mova [src0q+src1q], m1
add src0q, mmsize
sub lend, mmsize/4
jg .loop
REP_RET
%endmacro
INIT_XMM sse
MIX_1_TO_2_FLTP_FLT
%if HAVE_AVX
INIT_YMM avx
MIX_1_TO_2_FLTP_FLT
%endif
;-----------------------------------------------------------------------------
; void ff_mix_1_to_2_s16p_flt(int16_t **src, float **matrix, int len,
; int out_ch, int in_ch);
;-----------------------------------------------------------------------------
%macro MIX_1_TO_2_S16P_FLT 0
cglobal mix_1_to_2_s16p_flt, 3,5,6, src0, matrix0, len, src1, matrix1
mov src1q, [src0q+gprsize]
mov src0q, [src0q]
sub src1q, src0q
mov matrix1q, [matrix0q+gprsize]
mov matrix0q, [matrix0q]
VBROADCASTSS m4, [matrix0q]
VBROADCASTSS m5, [matrix1q]
ALIGN 16
.loop:
mova m0, [src0q]
S16_TO_S32_SX 0, 2
cvtdq2ps m0, m0
cvtdq2ps m2, m2
mulps m1, m0, m5
mulps m0, m0, m4
mulps m3, m2, m5
mulps m2, m2, m4
cvtps2dq m0, m0
cvtps2dq m1, m1
cvtps2dq m2, m2
cvtps2dq m3, m3
packssdw m0, m2
packssdw m1, m3
mova [src0q ], m0
mova [src0q+src1q], m1
add src0q, mmsize
sub lend, mmsize/2
jg .loop
REP_RET
%endmacro
INIT_XMM sse2
MIX_1_TO_2_S16P_FLT
INIT_XMM sse4
MIX_1_TO_2_S16P_FLT
%if HAVE_AVX
INIT_XMM avx
MIX_1_TO_2_S16P_FLT
%endif

22
libavresample/x86/audio_mix_init.c
View file

@ -35,6 +35,18 @@ extern void ff_mix_2_to_1_s16p_flt_sse4(int16_t **src, float **matrix, int len,
extern void ff_mix_2_to_1_s16p_q8_sse2(int16_t **src, int16_t **matrix,
int len, int out_ch, int in_ch);
extern void ff_mix_1_to_2_fltp_flt_sse(float **src, float **matrix, int len,
int out_ch, int in_ch);
extern void ff_mix_1_to_2_fltp_flt_avx(float **src, float **matrix, int len,
int out_ch, int in_ch);
extern void ff_mix_1_to_2_s16p_flt_sse2(int16_t **src, float **matrix, int len,
int out_ch, int in_ch);
extern void ff_mix_1_to_2_s16p_flt_sse4(int16_t **src, float **matrix, int len,
int out_ch, int in_ch);
extern void ff_mix_1_to_2_s16p_flt_avx (int16_t **src, float **matrix, int len,
int out_ch, int in_ch);
av_cold void ff_audio_mix_init_x86(AudioMix *am)
{
#if HAVE_YASM
@ -43,20 +55,30 @@ av_cold void ff_audio_mix_init_x86(AudioMix *am)
if (mm_flags & AV_CPU_FLAG_SSE && HAVE_SSE) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 16, 8, "SSE", ff_mix_2_to_1_fltp_flt_sse);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
1, 2, 16, 4, "SSE", ff_mix_1_to_2_fltp_flt_sse);
}
if (mm_flags & AV_CPU_FLAG_SSE2 && HAVE_SSE) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
2, 1, 16, 8, "SSE2", ff_mix_2_to_1_s16p_flt_sse2);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
2, 1, 16, 8, "SSE2", ff_mix_2_to_1_s16p_q8_sse2);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
1, 2, 16, 8, "SSE2", ff_mix_1_to_2_s16p_flt_sse2);
}
if (mm_flags & AV_CPU_FLAG_SSE4 && HAVE_SSE) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
2, 1, 16, 8, "SSE4", ff_mix_2_to_1_s16p_flt_sse4);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
1, 2, 16, 8, "SSE4", ff_mix_1_to_2_s16p_flt_sse4);
}
if (mm_flags & AV_CPU_FLAG_AVX && HAVE_AVX) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 32, 16, "AVX", ff_mix_2_to_1_fltp_flt_avx);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
1, 2, 32, 8, "AVX", ff_mix_1_to_2_fltp_flt_avx);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
1, 2, 16, 8, "AVX", ff_mix_1_to_2_s16p_flt_avx);
}
#endif
}