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Merge remote-tracking branch 'qatar/master'

* qatar/master:
  mss3: use standard zigzag table
  mss3: split DSP functions that are used in MTS2(MSS4) into separate file
  motion-test: do not use getopt()
  tcp: add initial timeout limit for incoming connections
  configure: Change the rdtsc check to a linker check
  avconv: propagate fatal errors from lavfi.
  lavfi: add error handling to filter_samples().
  fate-run: make avconv() properly deal with multiple inputs.
  asplit: don't leak the input buffer.
  af_resample: fix request_frame() behavior.
  af_asyncts: fix request_frame() behavior.
  libx264: support aspect ratio switching
  matroskadec: honor error_recognition when encountering unknown elements.
  lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
  lavr: resampling: add filter type and Kaiser window beta to AVOptions
  lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format
  lavr: mix: validate internal sample format in ff_audio_mix_init()

Conflicts:
	ffmpeg.c
	ffplay.c
	libavcodec/libx264.c
	libavfilter/audio.c
	libavfilter/split.c
	libavformat/tcp.c
	tests/fate-run.sh

Merged-by: Michael Niedermayer <michaelni@gmx.at>
  • Loading branch information...
commit f8911b987de4a84ff8ae92f41ff492ece4acadb9 2 parents bf53863 + 5467742
@michaelni michaelni authored
Showing with 648 additions and 339 deletions.
  1. +1 −1  libavcodec/Makefile
  2. +3 −3 libavcodec/libx264.c
  3. +3 −9 libavcodec/motion-test.c
  4. +5 −103 libavcodec/mss3.c
  5. +114 −0 libavcodec/mss34dsp.c
  6. +45 −0 libavcodec/mss34dsp.h
  7. +4 −2 libavfilter/af_aconvert.c
  8. +3 −3 libavfilter/af_amerge.c
  9. +13 −9 libavfilter/af_amix.c
  10. +5 −3 libavfilter/af_aresample.c
  11. +3 −2 libavfilter/af_asetnsamples.c
  12. +2 −2 libavfilter/af_ashowinfo.c
  13. +6 −3 libavfilter/af_astreamsync.c
  14. +31 −17 libavfilter/af_asyncts.c
  15. +2 −1  libavfilter/af_atempo.c
  16. +6 −4 libavfilter/af_channelmap.c
  17. +10 −5 libavfilter/af_channelsplit.c
  18. +5 −2 libavfilter/af_earwax.c
  19. +5 −3 libavfilter/af_join.c
  20. +4 −2 libavfilter/af_pan.c
  21. +31 −14 libavfilter/af_resample.c
  22. +2 −2 libavfilter/af_silencedetect.c
  23. +2 −2 libavfilter/af_volume.c
  24. +4 −1 libavfilter/asink_anullsink.c
  25. +14 −12 libavfilter/audio.c
  26. +5 −2 libavfilter/audio.h
  27. +2 −1  libavfilter/avf_showwaves.c
  28. +5 −1 libavfilter/avfilter.h
  29. +7 −1 libavfilter/buffersink.c
  30. +3 −2 libavfilter/buffersrc.c
  31. +2 −2 libavfilter/f_settb.c
  32. +18 −8 libavfilter/fifo.c
  33. +5 −1 libavfilter/internal.h
  34. +2 −1  libavfilter/sink_buffer.c
  35. +9 −5 libavfilter/split.c
  36. +4 −1 libavformat/matroskadec.c
  37. +10 −1 libavformat/tcp.c
  38. +8 −0 libavresample/audio_mix.c
  39. +7 −0 libavresample/avresample.h
  40. +2 −0  libavresample/internal.h
  41. +6 −1 libavresample/options.c
  42. +85 −95 libavresample/resample.c
  43. +102 −0 libavresample/resample_template.c
  44. +36 −11 libavresample/utils.c
  45. +7 −1 tests/fate-run.sh
View
2  libavcodec/Makefile
@@ -327,7 +327,7 @@ OBJS-$(CONFIG_MSMPEG4V3_ENCODER) += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
h263dec.o h263.o ituh263dec.o \
mpeg4videodec.o
OBJS-$(CONFIG_MSRLE_DECODER) += msrle.o msrledec.o
-OBJS-$(CONFIG_MSA1_DECODER) += mss3.o
+OBJS-$(CONFIG_MSA1_DECODER) += mss3.o mss34dsp.o
OBJS-$(CONFIG_MSS1_DECODER) += mss1.o
OBJS-$(CONFIG_MSVIDEO1_DECODER) += msvideo1.o
OBJS-$(CONFIG_MSVIDEO1_ENCODER) += msvideo1enc.o elbg.o
View
6 libavcodec/libx264.c
@@ -175,10 +175,10 @@ static int X264_frame(AVCodecContext *ctx, AVPacket *pkt, const AVFrame *frame,
x4->params.b_tff = frame->top_field_first;
x264_encoder_reconfig(x4->enc, &x4->params);
}
- if (x4->params.vui.i_sar_height != ctx->sample_aspect_ratio.den
- || x4->params.vui.i_sar_width != ctx->sample_aspect_ratio.num) {
+ if (x4->params.vui.i_sar_height != ctx->sample_aspect_ratio.den ||
+ x4->params.vui.i_sar_width != ctx->sample_aspect_ratio.num) {
x4->params.vui.i_sar_height = ctx->sample_aspect_ratio.den;
- x4->params.vui.i_sar_width = ctx->sample_aspect_ratio.num;
+ x4->params.vui.i_sar_width = ctx->sample_aspect_ratio.num;
x264_encoder_reconfig(x4->enc, &x4->params);
}
}
View
12 libavcodec/motion-test.c
@@ -119,15 +119,9 @@ int main(int argc, char **argv)
int flags[2] = { AV_CPU_FLAG_MMX, AV_CPU_FLAG_MMX2 };
int flags_size = HAVE_MMX2 ? 2 : 1;
- for(;;) {
- c = getopt(argc, argv, "h");
- if (c == -1)
- break;
- switch(c) {
- case 'h':
- help();
- return 1;
- }
+ if (argc > 1) {
+ help();
+ return 1;
}
printf("ffmpeg motion test\n");
View
108 libavcodec/mss3.c
@@ -26,6 +26,8 @@
#include "avcodec.h"
#include "bytestream.h"
+#include "dsputil.h"
+#include "mss34dsp.h"
#define HEADER_SIZE 27
@@ -119,39 +121,6 @@ typedef struct MSS3Context {
int hblock[16 * 16];
} MSS3Context;
-static const uint8_t mss3_luma_quant[64] = {
- 16, 11, 10, 16, 24, 40, 51, 61,
- 12, 12, 14, 19, 26, 58, 60, 55,
- 14, 13, 16, 24, 40, 57, 69, 56,
- 14, 17, 22, 29, 51, 87, 80, 62,
- 18, 22, 37, 56, 68, 109, 103, 77,
- 24, 35, 55, 64, 81, 104, 113, 92,
- 49, 64, 78, 87, 103, 121, 120, 101,
- 72, 92, 95, 98, 112, 100, 103, 99
-};
-
-static const uint8_t mss3_chroma_quant[64] = {
- 17, 18, 24, 47, 99, 99, 99, 99,
- 18, 21, 26, 66, 99, 99, 99, 99,
- 24, 26, 56, 99, 99, 99, 99, 99,
- 47, 66, 99, 99, 99, 99, 99, 99,
- 99, 99, 99, 99, 99, 99, 99, 99,
- 99, 99, 99, 99, 99, 99, 99, 99,
- 99, 99, 99, 99, 99, 99, 99, 99,
- 99, 99, 99, 99, 99, 99, 99, 99
-};
-
-static const uint8_t zigzag_scan[64] = {
- 0, 1, 8, 16, 9, 2, 3, 10,
- 17, 24, 32, 25, 18, 11, 4, 5,
- 12, 19, 26, 33, 40, 48, 41, 34,
- 27, 20, 13, 6, 7, 14, 21, 28,
- 35, 42, 49, 56, 57, 50, 43, 36,
- 29, 22, 15, 23, 30, 37, 44, 51,
- 58, 59, 52, 45, 38, 31, 39, 46,
- 53, 60, 61, 54, 47, 55, 62, 63
-};
-
static void model2_reset(Model2 *m)
{
@@ -578,7 +547,7 @@ static int decode_dct(RangeCoder *c, DCTBlockCoder *bc, int *block,
if (!sign)
val = -val;
- zz_pos = zigzag_scan[pos];
+ zz_pos = ff_zigzag_direct[pos];
block[zz_pos] = val * bc->qmat[zz_pos];
pos++;
}
@@ -586,58 +555,6 @@ static int decode_dct(RangeCoder *c, DCTBlockCoder *bc, int *block,
return pos == 64 ? 0 : -1;
}
-#define DCT_TEMPLATE(blk, step, SOP, shift) \
- const int t0 = -39409 * blk[7 * step] - 58980 * blk[1 * step]; \
- const int t1 = 39410 * blk[1 * step] - 58980 * blk[7 * step]; \
- const int t2 = -33410 * blk[5 * step] - 167963 * blk[3 * step]; \
- const int t3 = 33410 * blk[3 * step] - 167963 * blk[5 * step]; \
- const int t4 = blk[3 * step] + blk[7 * step]; \
- const int t5 = blk[1 * step] + blk[5 * step]; \
- const int t6 = 77062 * t4 + 51491 * t5; \
- const int t7 = 77062 * t5 - 51491 * t4; \
- const int t8 = 35470 * blk[2 * step] - 85623 * blk[6 * step]; \
- const int t9 = 35470 * blk[6 * step] + 85623 * blk[2 * step]; \
- const int tA = SOP(blk[0 * step] - blk[4 * step]); \
- const int tB = SOP(blk[0 * step] + blk[4 * step]); \
- \
- blk[0 * step] = ( t1 + t6 + t9 + tB) >> shift; \
- blk[1 * step] = ( t3 + t7 + t8 + tA) >> shift; \
- blk[2 * step] = ( t2 + t6 - t8 + tA) >> shift; \
- blk[3 * step] = ( t0 + t7 - t9 + tB) >> shift; \
- blk[4 * step] = (-(t0 + t7) - t9 + tB) >> shift; \
- blk[5 * step] = (-(t2 + t6) - t8 + tA) >> shift; \
- blk[6 * step] = (-(t3 + t7) + t8 + tA) >> shift; \
- blk[7 * step] = (-(t1 + t6) + t9 + tB) >> shift; \
-
-#define SOP_ROW(a) ((a) << 16) + 0x2000
-#define SOP_COL(a) ((a + 32) << 16)
-
-static void dct_put(uint8_t *dst, int stride, int *block)
-{
- int i, j;
- int *ptr;
-
- ptr = block;
- for (i = 0; i < 8; i++) {
- DCT_TEMPLATE(ptr, 1, SOP_ROW, 13);
- ptr += 8;
- }
-
- ptr = block;
- for (i = 0; i < 8; i++) {
- DCT_TEMPLATE(ptr, 8, SOP_COL, 22);
- ptr++;
- }
-
- ptr = block;
- for (j = 0; j < 8; j++) {
- for (i = 0; i < 8; i++)
- dst[i] = av_clip_uint8(ptr[i] + 128);
- dst += stride;
- ptr += 8;
- }
-}
-
static void decode_dct_block(RangeCoder *c, DCTBlockCoder *bc,
uint8_t *dst, int stride, int block_size,
int *block, int mb_x, int mb_y)
@@ -655,7 +572,7 @@ static void decode_dct_block(RangeCoder *c, DCTBlockCoder *bc,
c->got_error = 1;
return;
}
- dct_put(dst + i * 8, stride, block);
+ ff_mss34_dct_put(dst + i * 8, stride, block);
}
dst += 8 * stride;
}
@@ -702,14 +619,6 @@ static void decode_haar_block(RangeCoder *c, HaarBlockCoder *hc,
}
}
-static void gen_quant_mat(uint16_t *qmat, const uint8_t *ref, float scale)
-{
- int i;
-
- for (i = 0; i < 64; i++)
- qmat[i] = (uint16_t)(ref[i] * scale + 50.0) / 100;
-}
-
static void reset_coders(MSS3Context *ctx, int quality)
{
int i, j;
@@ -726,15 +635,8 @@ static void reset_coders(MSS3Context *ctx, int quality)
for (j = 0; j < 125; j++)
model_reset(&ctx->image_coder[i].vq_model[j]);
if (ctx->dct_coder[i].quality != quality) {
- float scale;
ctx->dct_coder[i].quality = quality;
- if (quality > 50)
- scale = 200.0f - 2 * quality;
- else
- scale = 5000.0f / quality;
- gen_quant_mat(ctx->dct_coder[i].qmat,
- i ? mss3_chroma_quant : mss3_luma_quant,
- scale);
+ ff_mss34_gen_quant_mat(ctx->dct_coder[i].qmat, quality, !i);
}
memset(ctx->dct_coder[i].prev_dc, 0,
sizeof(*ctx->dct_coder[i].prev_dc) *
View
114 libavcodec/mss34dsp.c
@@ -0,0 +1,114 @@
+/*
+ * Common stuff for some Microsoft Screen codecs
+ * Copyright (C) 2012 Konstantin Shishkov
+ *
+ * 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 <stdint.h>
+#include "libavutil/common.h"
+#include "mss34dsp.h"
+
+static const uint8_t luma_quant[64] = {
+ 16, 11, 10, 16, 24, 40, 51, 61,
+ 12, 12, 14, 19, 26, 58, 60, 55,
+ 14, 13, 16, 24, 40, 57, 69, 56,
+ 14, 17, 22, 29, 51, 87, 80, 62,
+ 18, 22, 37, 56, 68, 109, 103, 77,
+ 24, 35, 55, 64, 81, 104, 113, 92,
+ 49, 64, 78, 87, 103, 121, 120, 101,
+ 72, 92, 95, 98, 112, 100, 103, 99
+};
+
+static const uint8_t chroma_quant[64] = {
+ 17, 18, 24, 47, 99, 99, 99, 99,
+ 18, 21, 26, 66, 99, 99, 99, 99,
+ 24, 26, 56, 99, 99, 99, 99, 99,
+ 47, 66, 99, 99, 99, 99, 99, 99,
+ 99, 99, 99, 99, 99, 99, 99, 99,
+ 99, 99, 99, 99, 99, 99, 99, 99,
+ 99, 99, 99, 99, 99, 99, 99, 99,
+ 99, 99, 99, 99, 99, 99, 99, 99
+};
+
+void ff_mss34_gen_quant_mat(uint16_t *qmat, int quality, int luma)
+{
+ int i;
+ const uint8_t *qsrc = luma ? luma_quant : chroma_quant;
+
+ if (quality >= 50) {
+ int scale = 200 - 2 * quality;
+
+ for (i = 0; i < 64; i++)
+ qmat[i] = (qsrc[i] * scale + 50) / 100;
+ } else {
+ for (i = 0; i < 64; i++)
+ qmat[i] = (5000 * qsrc[i] / quality + 50) / 100;
+ }
+}
+
+#define DCT_TEMPLATE(blk, step, SOP, shift) \
+ const int t0 = -39409 * blk[7 * step] - 58980 * blk[1 * step]; \
+ const int t1 = 39410 * blk[1 * step] - 58980 * blk[7 * step]; \
+ const int t2 = -33410 * blk[5 * step] - 167963 * blk[3 * step]; \
+ const int t3 = 33410 * blk[3 * step] - 167963 * blk[5 * step]; \
+ const int t4 = blk[3 * step] + blk[7 * step]; \
+ const int t5 = blk[1 * step] + blk[5 * step]; \
+ const int t6 = 77062 * t4 + 51491 * t5; \
+ const int t7 = 77062 * t5 - 51491 * t4; \
+ const int t8 = 35470 * blk[2 * step] - 85623 * blk[6 * step]; \
+ const int t9 = 35470 * blk[6 * step] + 85623 * blk[2 * step]; \
+ const int tA = SOP(blk[0 * step] - blk[4 * step]); \
+ const int tB = SOP(blk[0 * step] + blk[4 * step]); \
+ \
+ blk[0 * step] = ( t1 + t6 + t9 + tB) >> shift; \
+ blk[1 * step] = ( t3 + t7 + t8 + tA) >> shift; \
+ blk[2 * step] = ( t2 + t6 - t8 + tA) >> shift; \
+ blk[3 * step] = ( t0 + t7 - t9 + tB) >> shift; \
+ blk[4 * step] = (-(t0 + t7) - t9 + tB) >> shift; \
+ blk[5 * step] = (-(t2 + t6) - t8 + tA) >> shift; \
+ blk[6 * step] = (-(t3 + t7) + t8 + tA) >> shift; \
+ blk[7 * step] = (-(t1 + t6) + t9 + tB) >> shift; \
+
+#define SOP_ROW(a) ((a) << 16) + 0x2000
+#define SOP_COL(a) ((a + 32) << 16)
+
+void ff_mss34_dct_put(uint8_t *dst, int stride, int *block)
+{
+ int i, j;
+ int *ptr;
+
+ ptr = block;
+ for (i = 0; i < 8; i++) {
+ DCT_TEMPLATE(ptr, 1, SOP_ROW, 13);
+ ptr += 8;
+ }
+
+ ptr = block;
+ for (i = 0; i < 8; i++) {
+ DCT_TEMPLATE(ptr, 8, SOP_COL, 22);
+ ptr++;
+ }
+
+ ptr = block;
+ for (j = 0; j < 8; j++) {
+ for (i = 0; i < 8; i++)
+ dst[i] = av_clip_uint8(ptr[i] + 128);
+ dst += stride;
+ ptr += 8;
+ }
+}
View
45 libavcodec/mss34dsp.h
@@ -0,0 +1,45 @@
+/*
+ * Common stuff for some Microsoft Screen codecs
+ * Copyright (C) 2012 Konstantin Shishkov
+ *
+ * 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_MSS34DSP_H
+#define AVCODEC_MSS34DSP_H
+
+#include <stdint.h>
+
+/**
+ * Generate quantisation matrix for given quality.
+ *
+ * @param qmat destination matrix
+ * @param quality quality setting (1-100)
+ * @param luma generate quantisation matrix for luma or chroma
+ */
+void ff_mss34_gen_quant_mat(uint16_t *qmat, int quality, int luma);
+
+/**
+ * Transform and output DCT block.
+ *
+ * @param dst output plane
+ * @param stride output plane stride
+ * @param block block to transform and output
+ */
+void ff_mss34_dct_put(uint8_t *dst, int stride, int *block);
+
+#endif /* AVCODEC_MSS34DSP_H */
View
6 libavfilter/af_aconvert.c
@@ -135,12 +135,13 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AConvertContext *aconvert = inlink->dst->priv;
const int n = insamplesref->audio->nb_samples;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n);
+ int ret;
swr_convert(aconvert->swr, outsamplesref->data, n,
(void *)insamplesref->data, n);
@@ -148,8 +149,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
outsamplesref->audio->channel_layout = outlink->channel_layout;
- ff_filter_samples(outlink, outsamplesref);
+ ret = ff_filter_samples(outlink, outsamplesref);
avfilter_unref_buffer(insamplesref);
+ return ret;
}
AVFilter avfilter_af_aconvert = {
View
6 libavfilter/af_amerge.c
@@ -212,7 +212,7 @@ static inline void copy_samples(int nb_inputs, struct amerge_input in[],
}
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AMergeContext *am = ctx->priv;
@@ -232,7 +232,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
for (i = 1; i < am->nb_inputs; i++)
nb_samples = FFMIN(nb_samples, am->in[i].nb_samples);
if (!nb_samples)
- return;
+ return 0;
outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE, nb_samples);
outs = outbuf->data[0];
@@ -285,7 +285,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
}
}
- ff_filter_samples(ctx->outputs[0], outbuf);
+ return ff_filter_samples(ctx->outputs[0], outbuf);
}
static av_cold int init(AVFilterContext *ctx, const char *args)
View
22 libavfilter/af_amix.c
@@ -305,9 +305,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
if (s->next_pts != AV_NOPTS_VALUE)
s->next_pts += nb_samples;
- ff_filter_samples(outlink, out_buf);
-
- return 0;
+ return ff_filter_samples(outlink, out_buf);
}
/**
@@ -448,31 +446,37 @@ static int request_frame(AVFilterLink *outlink)
return output_frame(outlink, available_samples);
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
MixContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
- int i;
+ int i, ret = 0;
for (i = 0; i < ctx->nb_inputs; i++)
if (ctx->inputs[i] == inlink)
break;
if (i >= ctx->nb_inputs) {
av_log(ctx, AV_LOG_ERROR, "unknown input link\n");
- return;
+ ret = AVERROR(EINVAL);
+ goto fail;
}
if (i == 0) {
int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
outlink->time_base);
- frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
+ ret = frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
+ if (ret < 0)
+ goto fail;
}
- av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
- buf->audio->nb_samples);
+ ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
+ buf->audio->nb_samples);
+fail:
avfilter_unref_buffer(buf);
+
+ return ret;
}
static int init(AVFilterContext *ctx, const char *args)
View
8 libavfilter/af_aresample.c
@@ -168,13 +168,14 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AResampleContext *aresample = inlink->dst->priv;
const int n_in = insamplesref->audio->nb_samples;
int n_out = n_in * aresample->ratio * 2 ;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
+ int ret;
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
@@ -193,15 +194,16 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
if (n_out <= 0) {
avfilter_unref_buffer(outsamplesref);
avfilter_unref_buffer(insamplesref);
- return;
+ return 0;
}
outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out;
- ff_filter_samples(outlink, outsamplesref);
+ ret = ff_filter_samples(outlink, outsamplesref);
aresample->req_fullfilled= 1;
avfilter_unref_buffer(insamplesref);
+ return ret;
}
static int request_frame(AVFilterLink *outlink)
View
5 libavfilter/af_asetnsamples.c
@@ -131,7 +131,7 @@ static int push_samples(AVFilterLink *outlink)
return nb_out_samples;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
ASNSContext *asns = ctx->priv;
@@ -145,7 +145,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
if (ret < 0) {
av_log(ctx, AV_LOG_ERROR,
"Stretching audio fifo failed, discarded %d samples\n", nb_samples);
- return;
+ return -1;
}
}
av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
@@ -155,6 +155,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
if (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
push_samples(outlink);
+ return 0;
}
static int request_frame(AVFilterLink *outlink)
View
4 libavfilter/af_ashowinfo.c
@@ -40,7 +40,7 @@ static av_cold int init(AVFilterContext *ctx, const char *args)
return 0;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterContext *ctx = inlink->dst;
ShowInfoContext *showinfo = ctx->priv;
@@ -83,7 +83,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
av_log(ctx, AV_LOG_INFO, "]\n");
showinfo->frame++;
- ff_filter_samples(inlink->dst->outputs[0], samplesref);
+ return ff_filter_samples(inlink->dst->outputs[0], samplesref);
}
AVFilter avfilter_af_ashowinfo = {
View
9 libavfilter/af_astreamsync.c
@@ -107,11 +107,12 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
-static void send_out(AVFilterContext *ctx, int out_id)
+static int send_out(AVFilterContext *ctx, int out_id)
{
AStreamSyncContext *as = ctx->priv;
struct buf_queue *queue = &as->queue[out_id];
AVFilterBufferRef *buf = queue->buf[queue->tail];
+ int ret;
queue->buf[queue->tail] = NULL;
as->var_values[VAR_B1 + out_id]++;
@@ -121,11 +122,12 @@ static void send_out(AVFilterContext *ctx, int out_id)
av_q2d(ctx->outputs[out_id]->time_base) * buf->pts;
as->var_values[VAR_T1 + out_id] += buf->audio->nb_samples /
(double)ctx->inputs[out_id]->sample_rate;
- ff_filter_samples(ctx->outputs[out_id], buf);
+ ret = ff_filter_samples(ctx->outputs[out_id], buf);
queue->nb--;
queue->tail = (queue->tail + 1) % QUEUE_SIZE;
if (as->req[out_id])
as->req[out_id]--;
+ return ret;
}
static void send_next(AVFilterContext *ctx)
@@ -165,7 +167,7 @@ static int request_frame(AVFilterLink *outlink)
return 0;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AStreamSyncContext *as = ctx->priv;
@@ -175,6 +177,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
insamples;
as->eof &= ~(1 << id);
send_next(ctx);
+ return 0;
}
AVFilter avfilter_af_astreamsync = {
View
48 libavfilter/af_asyncts.c
@@ -37,6 +37,9 @@ typedef struct ASyncContext {
int resample;
float min_delta_sec;
int max_comp;
+
+ /* set by filter_samples() to signal an output frame to request_frame() */
+ int got_output;
} ASyncContext;
#define OFFSET(x) offsetof(ASyncContext, x)
@@ -112,9 +115,13 @@ static int request_frame(AVFilterLink *link)
{
AVFilterContext *ctx = link->src;
ASyncContext *s = ctx->priv;
- int ret = ff_request_frame(ctx->inputs[0]);
+ int ret = 0;
int nb_samples;
+ s->got_output = 0;
+ while (ret >= 0 && !s->got_output)
+ ret = ff_request_frame(ctx->inputs[0]);
+
/* flush the fifo */
if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) {
AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
@@ -124,18 +131,18 @@ static int request_frame(AVFilterLink *link)
avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0],
nb_samples, NULL, 0, 0);
buf->pts = s->pts;
- ff_filter_samples(link, buf);
- return 0;
+ return ff_filter_samples(link, buf);
}
return ret;
}
-static void write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
+static int write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
{
- avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
- buf->linesize[0], buf->audio->nb_samples);
+ int ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
+ buf->linesize[0], buf->audio->nb_samples);
avfilter_unref_buffer(buf);
+ return ret;
}
/* get amount of data currently buffered, in samples */
@@ -144,7 +151,7 @@ static int64_t get_delay(ASyncContext *s)
return avresample_available(s->avr) + avresample_get_delay(s->avr);
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ASyncContext *s = ctx->priv;
@@ -152,7 +159,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
- int out_size;
+ int out_size, ret;
int64_t delta;
/* buffer data until we get the first timestamp */
@@ -160,14 +167,12 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
if (pts != AV_NOPTS_VALUE) {
s->pts = pts - get_delay(s);
}
- write_to_fifo(s, buf);
- return;
+ return write_to_fifo(s, buf);
}
/* now wait for the next timestamp */
if (pts == AV_NOPTS_VALUE) {
- write_to_fifo(s, buf);
- return;
+ return write_to_fifo(s, buf);
}
/* when we have two timestamps, compute how many samples would we have
@@ -190,8 +195,10 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
if (out_size > 0) {
AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
out_size);
- if (!buf_out)
- return;
+ if (!buf_out) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
avresample_read(s->avr, (void**)buf_out->extended_data, out_size);
buf_out->pts = s->pts;
@@ -200,7 +207,10 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
av_samples_set_silence(buf_out->extended_data, out_size - delta,
delta, nb_channels, buf->format);
}
- ff_filter_samples(outlink, buf_out);
+ ret = ff_filter_samples(outlink, buf_out);
+ if (ret < 0)
+ goto fail;
+ s->got_output = 1;
} else {
av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
"whole buffer.\n");
@@ -210,9 +220,13 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
avresample_read(s->avr, NULL, avresample_available(s->avr));
s->pts = pts - avresample_get_delay(s->avr);
- avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
- buf->linesize[0], buf->audio->nb_samples);
+ ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
+ buf->linesize[0], buf->audio->nb_samples);
+
+fail:
avfilter_unref_buffer(buf);
+
+ return ret;
}
AVFilter avfilter_af_asyncts = {
View
3  libavfilter/af_atempo.c
@@ -1040,7 +1040,7 @@ static void push_samples(ATempoContext *atempo,
atempo->nsamples_out += n_out;
}
-static void filter_samples(AVFilterLink *inlink,
+static int filter_samples(AVFilterLink *inlink,
AVFilterBufferRef *src_buffer)
{
AVFilterContext *ctx = inlink->dst;
@@ -1074,6 +1074,7 @@ static void filter_samples(AVFilterLink *inlink,
atempo->nsamples_in += n_in;
avfilter_unref_bufferp(&src_buffer);
+ return 0;
}
static int request_frame(AVFilterLink *outlink)
View
10 libavfilter/af_channelmap.c
@@ -313,7 +313,7 @@ static int channelmap_query_formats(AVFilterContext *ctx)
return 0;
}
-static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
@@ -330,8 +330,10 @@ static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *b
if (nch_out > FF_ARRAY_ELEMS(buf->data)) {
uint8_t **new_extended_data =
av_mallocz(nch_out * sizeof(*buf->extended_data));
- if (!new_extended_data)
- return;
+ if (!new_extended_data) {
+ avfilter_unref_buffer(buf);
+ return AVERROR(ENOMEM);
+ }
if (buf->extended_data == buf->data) {
buf->extended_data = new_extended_data;
} else {
@@ -353,7 +355,7 @@ static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *b
memcpy(buf->data, buf->extended_data,
FFMIN(FF_ARRAY_ELEMS(buf->data), nch_out) * sizeof(buf->data[0]));
- ff_filter_samples(outlink, buf);
+ return ff_filter_samples(outlink, buf);
}
static int channelmap_config_input(AVFilterLink *inlink)
View
15 libavfilter/af_channelsplit.c
@@ -105,24 +105,29 @@ static int query_formats(AVFilterContext *ctx)
return 0;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
- int i;
+ int i, ret = 0;
for (i = 0; i < ctx->nb_outputs; i++) {
AVFilterBufferRef *buf_out = avfilter_ref_buffer(buf, ~AV_PERM_WRITE);
- if (!buf_out)
- return;
+ if (!buf_out) {
+ ret = AVERROR(ENOMEM);
+ break;
+ }
buf_out->data[0] = buf_out->extended_data[0] = buf_out->extended_data[i];
buf_out->audio->channel_layout =
av_channel_layout_extract_channel(buf->audio->channel_layout, i);
- ff_filter_samples(ctx->outputs[i], buf_out);
+ ret = ff_filter_samples(ctx->outputs[i], buf_out);
+ if (ret < 0)
+ break;
}
avfilter_unref_buffer(buf);
+ return ret;
}
AVFilter avfilter_af_channelsplit = {
View
7 libavfilter/af_earwax.c
@@ -120,13 +120,15 @@ static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, in
return out;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterLink *outlink = inlink->dst->outputs[0];
int16_t *taps, *endin, *in, *out;
AVFilterBufferRef *outsamples =
ff_get_audio_buffer(inlink, AV_PERM_WRITE,
insamples->audio->nb_samples);
+ int ret;
+
avfilter_copy_buffer_ref_props(outsamples, insamples);
taps = ((EarwaxContext *)inlink->dst->priv)->taps;
@@ -144,8 +146,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
// save part of input for next round
memcpy(taps, endin, NUMTAPS * sizeof(*taps));
- ff_filter_samples(outlink, outsamples);
+ ret = ff_filter_samples(outlink, outsamples);
avfilter_unref_buffer(insamples);
+ return ret;
}
AVFilter avfilter_af_earwax = {
View
8 libavfilter/af_join.c
@@ -92,7 +92,7 @@ static const AVClass join_class = {
.version = LIBAVUTIL_VERSION_INT,
};
-static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = link->dst;
JoinContext *s = ctx->priv;
@@ -104,6 +104,8 @@ static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
av_assert0(i < ctx->nb_inputs);
av_assert0(!s->input_frames[i]);
s->input_frames[i] = buf;
+
+ return 0;
}
static int parse_maps(AVFilterContext *ctx)
@@ -468,11 +470,11 @@ static int join_request_frame(AVFilterLink *outlink)
priv->nb_in_buffers = ctx->nb_inputs;
buf->buf->priv = priv;
- ff_filter_samples(outlink, buf);
+ ret = ff_filter_samples(outlink, buf);
memset(s->input_frames, 0, sizeof(*s->input_frames) * ctx->nb_inputs);
- return 0;
+ return ret;
fail:
avfilter_unref_buffer(buf);
View
6 libavfilter/af_pan.c
@@ -343,8 +343,9 @@ static int config_props(AVFilterLink *link)
return 0;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
+ int ret;
int n = insamples->audio->nb_samples;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamples = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n);
@@ -354,8 +355,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
avfilter_copy_buffer_ref_props(outsamples, insamples);
outsamples->audio->channel_layout = outlink->channel_layout;
- ff_filter_samples(outlink, outsamples);
+ ret = ff_filter_samples(outlink, outsamples);
avfilter_unref_buffer(insamples);
+ return ret;
}
static av_cold void uninit(AVFilterContext *ctx)
View
45 libavfilter/af_resample.c
@@ -38,6 +38,9 @@ typedef struct ResampleContext {
AVAudioResampleContext *avr;
int64_t next_pts;
+
+ /* set by filter_samples() to signal an output frame to request_frame() */
+ int got_output;
} ResampleContext;
static av_cold void uninit(AVFilterContext *ctx)
@@ -102,12 +105,6 @@ static int config_output(AVFilterLink *outlink)
av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
- /* if both the input and output formats are s16 or u8, use s16 as
- the internal sample format */
- if (av_get_bytes_per_sample(inlink->format) <= 2 &&
- av_get_bytes_per_sample(outlink->format) <= 2)
- av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
-
if ((ret = avresample_open(s->avr)) < 0)
return ret;
@@ -130,7 +127,11 @@ static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ResampleContext *s = ctx->priv;
- int ret = ff_request_frame(ctx->inputs[0]);
+ int ret = 0;
+
+ s->got_output = 0;
+ while (ret >= 0 && !s->got_output)
+ ret = ff_request_frame(ctx->inputs[0]);
/* flush the lavr delay buffer */
if (ret == AVERROR_EOF && s->avr) {
@@ -156,21 +157,21 @@ static int request_frame(AVFilterLink *outlink)
}
buf->pts = s->next_pts;
- ff_filter_samples(outlink, buf);
- return 0;
+ return ff_filter_samples(outlink, buf);
}
return ret;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ResampleContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
+ int ret;
if (s->avr) {
AVFilterBufferRef *buf_out;
- int delay, nb_samples, ret;
+ int delay, nb_samples;
/* maximum possible samples lavr can output */
delay = avresample_get_delay(s->avr);
@@ -179,10 +180,19 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
AV_ROUND_UP);
buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+ if (!buf_out) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+
ret = avresample_convert(s->avr, (void**)buf_out->extended_data,
buf_out->linesize[0], nb_samples,
(void**)buf->extended_data, buf->linesize[0],
buf->audio->nb_samples);
+ if (ret < 0) {
+ avfilter_unref_buffer(buf_out);
+ goto fail;
+ }
av_assert0(!avresample_available(s->avr));
@@ -208,11 +218,18 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
- ff_filter_samples(outlink, buf_out);
+ ret = ff_filter_samples(outlink, buf_out);
+ s->got_output = 1;
}
+
+fail:
avfilter_unref_buffer(buf);
- } else
- ff_filter_samples(outlink, buf);
+ } else {
+ ret = ff_filter_samples(outlink, buf);
+ s->got_output = 1;
+ }
+
+ return ret;
}
AVFilter avfilter_af_resample = {
View
4 libavfilter/af_silencedetect.c
@@ -78,7 +78,7 @@ static av_cold int init(AVFilterContext *ctx, const char *args)
return 0;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
int i;
SilenceDetectContext *silence = inlink->dst->priv;
@@ -118,7 +118,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
}
- ff_filter_samples(inlink->dst->outputs[0], insamples);
+ return ff_filter_samples(inlink->dst->outputs[0], insamples);
}
static int query_formats(AVFilterContext *ctx)
View
4 libavfilter/af_volume.c
@@ -110,7 +110,7 @@ static int query_formats(AVFilterContext *ctx)
return 0;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
VolumeContext *vol = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
@@ -169,7 +169,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
}
}
- ff_filter_samples(outlink, insamples);
+ return ff_filter_samples(outlink, insamples);
}
AVFilter avfilter_af_volume = {
View
5 libavfilter/asink_anullsink.c
@@ -21,7 +21,10 @@
#include "avfilter.h"
#include "internal.h"
-static void null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) { }
+static int null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+{
+ return 0;
+}
AVFilter avfilter_asink_anullsink = {
.name = "anullsink",
View
26 libavfilter/audio.c
@@ -150,19 +150,19 @@ AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data,
return NULL;
}
-static void default_filter_samples(AVFilterLink *link,
- AVFilterBufferRef *samplesref)
+static int default_filter_samples(AVFilterLink *link,
+ AVFilterBufferRef *samplesref)
{
- ff_filter_samples(link->dst->outputs[0], samplesref);
+ return ff_filter_samples(link->dst->outputs[0], samplesref);
}
-void ff_filter_samples_framed(AVFilterLink *link,
- AVFilterBufferRef *samplesref)
+int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
- void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
+ int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
AVFilterPad *dst = link->dstpad;
int64_t pts;
AVFilterBufferRef *buf_out;
+ int ret;
FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);
@@ -193,21 +193,22 @@ void ff_filter_samples_framed(AVFilterLink *link,
link->cur_buf = buf_out;
pts = buf_out->pts;
- filter_samples(link, buf_out);
+ ret = filter_samples(link, buf_out);
ff_update_link_current_pts(link, pts);
+ return ret;
}
-void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
AVFilterBufferRef *pbuf = link->partial_buf;
int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
+ int ret = 0;
if (!link->min_samples ||
(!pbuf &&
insamples >= link->min_samples && insamples <= link->max_samples)) {
- ff_filter_samples_framed(link, samplesref);
- return;
+ return ff_filter_samples_framed(link, samplesref);
}
/* Handle framing (min_samples, max_samples) */
while (insamples) {
@@ -218,7 +219,7 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
if (!pbuf) {
av_log(link->dst, AV_LOG_WARNING,
"Samples dropped due to memory allocation failure.\n");
- return;
+ return 0;
}
avfilter_copy_buffer_ref_props(pbuf, samplesref);
pbuf->pts = samplesref->pts +
@@ -234,10 +235,11 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
insamples -= nb_samples;
pbuf->audio->nb_samples += nb_samples;
if (pbuf->audio->nb_samples >= link->min_samples) {
- ff_filter_samples_framed(link, pbuf);
+ ret = ff_filter_samples_framed(link, pbuf);
pbuf = NULL;
}
}
avfilter_unref_buffer(samplesref);
link->partial_buf = pbuf;
+ return ret;
}
View
7 libavfilter/audio.h
@@ -70,14 +70,17 @@ AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
* @param samplesref a reference to the buffer of audio samples being sent. The
* receiving filter will free this reference when it no longer
* needs it or pass it on to the next filter.
+ *
+ * @return >= 0 on success, a negative AVERROR on error. The receiving filter
+ * is responsible for unreferencing samplesref in case of error.
*/
-void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
+int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
/**
* Send a buffer of audio samples to the next link, without checking
* min_samples.
*/
-void ff_filter_samples_framed(AVFilterLink *link,
+int ff_filter_samples_framed(AVFilterLink *link,
AVFilterBufferRef *samplesref);
#endif /* AVFILTER_AUDIO_H */
View
3  libavfilter/avf_showwaves.c
@@ -180,7 +180,7 @@ static int request_frame(AVFilterLink *outlink)
#define MAX_INT16 ((1<<15) -1)
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
@@ -225,6 +225,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
avfilter_unref_buffer(insamples);
+ return 0;
}
AVFilter avfilter_avf_showwaves = {
View
6 libavfilter/avfilter.h
@@ -301,8 +301,12 @@ struct AVFilterPad {
* and should do its processing.
*
* Input audio pads only.
+ *
+ * @return >= 0 on success, a negative AVERROR on error. This function
+ * must ensure that samplesref is properly unreferenced on error if it
+ * hasn't been passed on to another filter.
*/
- void (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
+ int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
/**
* Frame poll callback. This returns the number of immediately available
View
8 libavfilter/buffersink.c
@@ -56,6 +56,12 @@ static void start_frame(AVFilterLink *link, AVFilterBufferRef *buf)
link->cur_buf = NULL;
};
+static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
+{
+ start_frame(link, buf);
+ return 0;
+}
+
int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf)
{
BufferSinkContext *s = ctx->priv;
@@ -160,7 +166,7 @@ AVFilter avfilter_asink_abuffer = {
.inputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
- .filter_samples = start_frame,
+ .filter_samples = filter_samples,
.min_perms = AV_PERM_READ,
.needs_fifo = 1 },
{ .name = NULL }},
View
5 libavfilter/buffersrc.c
@@ -408,6 +408,7 @@ static int request_frame(AVFilterLink *link)
{
BufferSourceContext *c = link->src->priv;
AVFilterBufferRef *buf;
+ int ret = 0;
if (!av_fifo_size(c->fifo)) {
if (c->eof)
@@ -424,7 +425,7 @@ static int request_frame(AVFilterLink *link)
ff_end_frame(link);
break;
case AVMEDIA_TYPE_AUDIO:
- ff_filter_samples(link, avfilter_ref_buffer(buf, ~0));
+ ret = ff_filter_samples(link, avfilter_ref_buffer(buf, ~0));
break;
default:
return AVERROR(EINVAL);
@@ -432,7 +433,7 @@ static int request_frame(AVFilterLink *link)
avfilter_unref_buffer(buf);
- return 0;
+ return ret;
}
static int poll_frame(AVFilterLink *link)
View
4 libavfilter/f_settb.c
@@ -117,7 +117,7 @@ static void start_frame(AVFilterLink *inlink, AVFilterBufferRef *picref)
ff_start_frame(outlink, picref2);
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
@@ -132,7 +132,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
avfilter_unref_buffer(insamples);
}
- ff_filter_samples(outlink, outsamples);
+ return ff_filter_samples(outlink, outsamples);
}
#if CONFIG_SETTB_FILTER
View
26 libavfilter/fifo.c
@@ -72,13 +72,25 @@ static av_cold void uninit(AVFilterContext *ctx)
avfilter_unref_buffer(fifo->buf_out);
}
-static void add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
FifoContext *fifo = inlink->dst->priv;
fifo->last->next = av_mallocz(sizeof(Buf));
+ if (!fifo->last->next) {
+ avfilter_unref_buffer(buf);
+ return AVERROR(ENOMEM);
+ }
+
fifo->last = fifo->last->next;
fifo->last->buf = buf;
+
+ return 0;
+}
+
+static void start_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+ add_to_queue(inlink, buf);
}
static void queue_pop(FifoContext *s)
@@ -210,15 +222,13 @@ static int return_audio_frame(AVFilterContext *ctx)
buf_out = s->buf_out;
s->buf_out = NULL;
}
- ff_filter_samples(link, buf_out);
-
- return 0;
+ return ff_filter_samples(link, buf_out);
}
static int request_frame(AVFilterLink *outlink)
{
FifoContext *fifo = outlink->src->priv;
- int ret;
+ int ret = 0;
if (!fifo->root.next) {
if ((ret = ff_request_frame(outlink->src->inputs[0])) < 0)
@@ -238,7 +248,7 @@ static int request_frame(AVFilterLink *outlink)
if (outlink->request_samples) {
return return_audio_frame(outlink->src);
} else {
- ff_filter_samples(outlink, fifo->root.next->buf);
+ ret = ff_filter_samples(outlink, fifo->root.next->buf);
queue_pop(fifo);
}
break;
@@ -246,7 +256,7 @@ static int request_frame(AVFilterLink *outlink)
return AVERROR(EINVAL);
}
- return 0;
+ return ret;
}
AVFilter avfilter_vf_fifo = {
@@ -261,7 +271,7 @@ AVFilter avfilter_vf_fifo = {
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_VIDEO,
.get_video_buffer= ff_null_get_video_buffer,
- .start_frame = add_to_queue,
+ .start_frame = start_frame,
.draw_slice = draw_slice,
.end_frame = end_frame,
.rej_perms = AV_PERM_REUSE2, },
View
6 libavfilter/internal.h
@@ -135,8 +135,12 @@ struct AVFilterPad {
* and should do its processing.
*
* Input audio pads only.
+ *
+ * @return >= 0 on success, a negative AVERROR on error. This function
+ * must ensure that samplesref is properly unreferenced on error if it
+ * hasn't been passed on to another filter.
*/
- void (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
+ int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
/**
* Frame poll callback. This returns the number of immediately available
View
3  libavfilter/sink_buffer.c
@@ -244,9 +244,10 @@ AVFilter avfilter_vsink_buffersink = {
#if CONFIG_ABUFFERSINK_FILTER
-static void filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+static int filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
end_frame(link);
+ return 0;
}
static av_cold int asink_init(AVFilterContext *ctx, const char *args)
View
14 libavfilter/split.c
@@ -110,15 +110,19 @@ AVFilter avfilter_vf_split = {
.outputs = (AVFilterPad[]) {{ .name = NULL}},
};
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterContext *ctx = inlink->dst;
- int i;
+ int i, ret = 0;
- for (i = 0; i < ctx->nb_outputs; i++)
- ff_filter_samples(inlink->dst->outputs[i],
- avfilter_ref_buffer(samplesref, ~AV_PERM_WRITE));
+ for (i = 0; i < ctx->nb_outputs; i++) {
+ ret = ff_filter_samples(inlink->dst->outputs[i],
+ avfilter_ref_buffer(samplesref, ~AV_PERM_WRITE));
+ if (ret < 0)
+ break;
+ }
avfilter_unref_buffer(samplesref);
+ return ret;
}
AVFilter avfilter_af_asplit = {
View
5 libavformat/matroskadec.c
@@ -842,8 +842,11 @@ static int ebml_parse_id(MatroskaDemuxContext *matroska, EbmlSyntax *syntax,
matroska->num_levels > 0 &&
matroska->levels[matroska->num_levels-1].length == 0xffffffffffffff)
return 0; // we reached the end of an unknown size cluster
- if (!syntax[i].id && id != EBML_ID_VOID && id != EBML_ID_CRC32)
+ if (!syntax[i].id && id != EBML_ID_VOID && id != EBML_ID_CRC32) {
av_log(matroska->ctx, AV_LOG_INFO, "Unknown entry 0x%X\n", id);
+ if (matroska->ctx->error_recognition & AV_EF_EXPLODE)
+ return AVERROR_INVALIDDATA;
+ }
return ebml_parse_elem(matroska, &syntax[i], data);
}
View
11 libavformat/tcp.c
@@ -43,7 +43,7 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
char buf[256];
int ret;
socklen_t optlen;
- int timeout = 50;
+ int timeout = 50, listen_timeout = -1;
char hostname[1024],proto[1024],path[1024];
char portstr[10];
@@ -59,6 +59,9 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
if (av_find_info_tag(buf, sizeof(buf), "timeout", p)) {
timeout = strtol(buf, NULL, 10);
}
+ if (av_find_info_tag(buf, sizeof(buf), "listen_timeout", p)) {
+ listen_timeout = strtol(buf, NULL, 10);
+ }
}
hints.ai_family = AF_UNSPEC;
hints.ai_socktype = SOCK_STREAM;
@@ -87,6 +90,7 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
if (listen_socket) {
int fd1;
int reuse = 1;
+ struct pollfd lp = { fd, POLLIN, 0 };
setsockopt(fd, SOL_SOCKET, SO_REUSEADDR, &reuse, sizeof(reuse));
ret = bind(fd, cur_ai->ai_addr, cur_ai->ai_addrlen);
if (ret) {
@@ -98,6 +102,11 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
ret = ff_neterrno();
goto fail1;
}
+ ret = poll(&lp, 1, listen_timeout >= 0 ? listen_timeout : -1);
+ if (ret <= 0) {
+ ret = AVERROR(ETIMEDOUT);
+ goto fail1;
+ }
fd1 = accept(fd, NULL, NULL);
if (fd1 < 0) {
ret = ff_neterrno();
View
8 libavresample/audio_mix.c
@@ -305,6 +305,14 @@ int ff_audio_mix_init(AVAudioResampleContext *avr)
{
int ret;
+ if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
+ avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
+ av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
+ "mixing: %s\n",
+ av_get_sample_fmt_name(avr->internal_sample_fmt));
+ return AVERROR(EINVAL);
+ }
+
/* build matrix if the user did not already set one */
if (!avr->am->matrix) {
int i, j;
View
7 libavresample/avresample.h
@@ -45,6 +45,13 @@ enum AVMixCoeffType {
AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
};
+/** Resampling Filter Types */
+enum AVResampleFilterType {
+ AV_RESAMPLE_FILTER_TYPE_CUBIC, /**< Cubic */
+ AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
+ AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
+};
+
/**
* Return the LIBAVRESAMPLE_VERSION_INT constant.
*/
View
2  libavresample/internal.h
@@ -50,6 +50,8 @@ struct AVAudioResampleContext {
int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
+ enum AVResampleFilterType filter_type; /**< resampling filter type */
+ int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
int in_channels; /**< number of input channels */
int out_channels; /**< number of output channels */
View
7 libavresample/options.c
@@ -39,7 +39,7 @@ static const AVOption options[] = {
{ "out_channel_layout", "Output Channel Layout", OFFSET(out_channel_layout), AV_OPT_TYPE_INT64, { 0 }, INT64_MIN, INT64_MAX, PARAM },
{ "out_sample_fmt", "Output Sample Format", OFFSET(out_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM },
{ "out_sample_rate", "Output Sample Rate", OFFSET(out_sample_rate), AV_OPT_TYPE_INT, { 48000 }, 1, INT_MAX, PARAM },
- { "internal_sample_fmt", "Internal Sample Format", OFFSET(internal_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_FLTP }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, PARAM },
+ { "internal_sample_fmt", "Internal Sample Format", OFFSET(internal_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_NONE }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, PARAM },
{ "mix_coeff_type", "Mixing Coefficient Type", OFFSET(mix_coeff_type), AV_OPT_TYPE_INT, { AV_MIX_COEFF_TYPE_FLT }, AV_MIX_COEFF_TYPE_Q8, AV_MIX_COEFF_TYPE_NB-1, PARAM, "mix_coeff_type" },
{ "q8", "16-bit 8.8 Fixed-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q8 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
{ "q15", "32-bit 17.15 Fixed-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q15 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
@@ -56,6 +56,11 @@ static const AVOption options[] = {
{ "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" },
+ { "filter_type", "Filter Type", OFFSET(filter_type), AV_OPT_TYPE_INT, { AV_RESAMPLE_FILTER_TYPE_KAISER }, AV_RESAMPLE_FILTER_TYPE_CUBIC, AV_RESAMPLE_FILTER_TYPE_KAISER, PARAM, "filter_type" },
+ { "cubic", "Cubic", 0, AV_OPT_TYPE_CONST, { AV_RESAMPLE_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+ { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0, AV_OPT_TYPE_CONST, { AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+ { "kaiser", "Kaiser Windowed Sinc", 0, AV_OPT_TYPE_CONST, { AV_RESAMPLE_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+ { "kaiser_beta", "Kaiser Window Beta", OFFSET(kaiser_beta), AV_OPT_TYPE_INT, { 9 }, 2, 16, PARAM },
{ NULL },
};
View
180 libavresample/resample.c
@@ -24,37 +24,10 @@
#include "internal.h"
#include "audio_data.h"
-#ifdef CONFIG_RESAMPLE_FLT
-/* float template */
-#define FILTER_SHIFT 0
-#define FELEM float
-#define FELEM2 float
-#define FELEML float
-#define WINDOW_TYPE 24
-#elifdef CONFIG_RESAMPLE_S32
-/* s32 template */
-#define FILTER_SHIFT 30
-#define FELEM int32_t
-#define FELEM2 int64_t
-#define FELEML int64_t
-#define FELEM_MAX INT32_MAX
-#define FELEM_MIN INT32_MIN
-#define WINDOW_TYPE 12
-#else
-/* s16 template */
-#define FILTER_SHIFT 15
-#define FELEM int16_t
-#define FELEM2 int32_t
-#define FELEML int64_t
-#define FELEM_MAX INT16_MAX
-#define FELEM_MIN INT16_MIN
-#define WINDOW_TYPE 9
-#endif
-
struct ResampleContext {
AVAudioResampleContext *avr;
AudioData *buffer;
- FELEM *filter_bank;
+ uint8_t *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
@@ -65,9 +38,35 @@ struct ResampleContext {
int phase_shift;
int phase_mask;
int linear;
+ enum AVResampleFilterType filter_type;
+ int kaiser_beta;
double factor;
+ void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
+ void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0,
+ int dst_index, const void *src0, int src_size,
+ int index, int frac);
};
+
+/* double template */
+#define CONFIG_RESAMPLE_DBL
+#include "resample_template.c"
+#undef CONFIG_RESAMPLE_DBL
+
+/* float template */
+#define CONFIG_RESAMPLE_FLT
+#include "resample_template.c"
+#undef CONFIG_RESAMPLE_FLT
+
+/* s32 template */
+#define CONFIG_RESAMPLE_S32
+#include "resample_template.c"
+#undef CONFIG_RESAMPLE_S32
+
+/* s16 template */
+#include "resample_template.c"
+
+
/**
* 0th order modified bessel function of the first kind.
*/
@@ -95,17 +94,17 @@ static double bessel(double x)
* @param tap_count tap count
* @param phase_count phase count
* @param scale wanted sum of coefficients for each filter
- * @param type 0->cubic
- * 1->blackman nuttall windowed sinc
- * 2..16->kaiser windowed sinc beta=2..16
+ * @param filter_type filter type
+ * @param kaiser_beta kaiser window beta
* @return 0 on success, negative AVERROR code on failure
*/
-static int build_filter(FELEM *filter, double factor, int tap_count,
- int phase_count, int scale, int type)
+static int build_filter(ResampleContext *c)
{
int ph, i;
- double x, y, w;
+ double x, y, w, factor;
double *tab;
+ int tap_count = c->filter_length;
+ int phase_count = 1 << c->phase_shift;
const int center = (tap_count - 1) / 2;
tab = av_malloc(tap_count * sizeof(*tab));
@@ -113,8 +112,7 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
return AVERROR(ENOMEM);
/* if upsampling, only need to interpolate, no filter */
- if (factor > 1.0)
- factor = 1.0;
+ factor = FFMIN(c->factor, 1.0);
for (ph = 0; ph < phase_count; ph++) {
double norm = 0;
@@ -122,39 +120,34 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else y = sin(x) / x;
- switch (type) {
- case 0: {
+ switch (c->filter_type) {
+ case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
const float d = -0.5; //first order derivative = -0.5
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * ( -x*x + x*x*x);
else y = d * (-4 + 8 * x - 5 * x*x + x*x*x);
break;
}
- case 1:
+ case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
w = 2.0 * x / (factor * tap_count) + M_PI;
y *= 0.3635819 - 0.4891775 * cos( w) +
0.1365995 * cos(2 * w) -
0.0106411 * cos(3 * w);
break;
- default:
+ case AV_RESAMPLE_FILTER_TYPE_KAISER:
w = 2.0 * x / (factor * tap_count * M_PI);
- y *= bessel(type * sqrt(FFMAX(1 - w * w, 0)));
+ y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
break;
}
tab[i] = y;
norm += y;
}
-
/* normalize so that an uniform color remains the same */
- for (i = 0; i < tap_count; i++) {
-#ifdef CONFIG_RESAMPLE_FLT
- filter[ph * tap_count + i] = tab[i] / norm;
-#else
- filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm),
- FELEM_MIN, FELEM_MAX);
-#endif
- }
+ for (i = 0; i < tap_count; i++)
+ tab[i] = tab[i] / norm;
+
+ c->set_filter(c->filter_bank, tab, ph, tap_count);
}
av_free(tab);
@@ -168,9 +161,12 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
int in_rate = avr->in_sample_rate;
double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
int phase_count = 1 << avr->phase_shift;
+ int felem_size;
- /* TODO: add support for s32 and float internal formats */
- if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
+ if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
+ avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
+ avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
+ avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
"resampling: %s\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
@@ -186,18 +182,40 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
c->linear = avr->linear_interp;
c->factor = factor;
c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
+ c->filter_type = avr->filter_type;
+ c->kaiser_beta = avr->kaiser_beta;
+
+ switch (avr->internal_sample_fmt) {
+ case AV_SAMPLE_FMT_DBLP:
+ c->resample_one = resample_one_dbl;
+ c->set_filter = set_filter_dbl;
+ break;
+ case AV_SAMPLE_FMT_FLTP:
+ c->resample_one = resample_one_flt;
+ c->set_filter = set_filter_flt;
+ break;
+ case AV_SAMPLE_FMT_S32P:
+ c->resample_one = resample_one_s32;
+ c->set_filter = set_filter_s32;
+ break;
+ case AV_SAMPLE_FMT_S16P:
+ c->resample_one = resample_one_s16;
+ c->set_filter = set_filter_s16;
+ break;
+ }
- c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * sizeof(FELEM));
+ felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
+ c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
if (!c->filter_bank)
goto error;
- if (build_filter(c->filter_bank, factor, c->filter_length, phase_count,
- 1 << FILTER_SHIFT, WINDOW_TYPE) < 0)
+ if (build_filter(c) < 0)
goto error;
- memcpy(&c->filter_bank[c->filter_length * phase_count + 1],
- c->filter_bank, (c->filter_length - 1) * sizeof(FELEM));
- c->filter_bank[c->filter_length * phase_count] = c->filter_bank[c->filter_length - 1];
+ memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
+ c->filter_bank, (c->filter_length - 1) * felem_size);
+ memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
+ &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
c->compensation_distance = 0;
if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
@@ -311,10 +329,10 @@ int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
return ret;
}
-static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
+static int resample(ResampleContext *c, void *dst, const void *src,
int *consumed, int src_size, int dst_size, int update_ctx)
{
- int dst_index, i;
+ int dst_index;
int index = c->index;
int frac = c->frac;
int dst_incr_frac = c->dst_incr % c->src_incr;
@@ -334,7 +352,7 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
if (dst) {
for(dst_index = 0; dst_index < dst_size; dst_index++) {
- dst[dst_index] = src[index2 >> 32];
+ c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);
index2 += incr;
}
} else {
@@ -345,42 +363,14 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
} else {
for (dst_index = 0; dst_index < dst_size; dst_index++) {
- FELEM *filter = c->filter_bank +
- c->filter_length * (index & c->phase_mask);
int sample_index = index >> c->phase_shift;
- if (!dst && (sample_index + c->filter_length > src_size ||
- -sample_index >= src_size))
+ if (sample_index + c->filter_length > src_size ||
+ -sample_index >= src_size)
break;
- if (dst) {
- FELEM2 val = 0;
-
- if (sample_index < 0) {
- for (i = 0; i < c->filter_length; i++)
- val += src[FFABS(sample_index + i) % src_size] *
- (FELEM2)filter[i];
- } else if (sample_index + c->filter_length > src_size) {
- break;
- } else if (c->linear) {
- FELEM2 v2 = 0;
- for (i = 0; i < c->filter_length; i++) {
- val += src[abs(sample_index + i)] * (FELEM2)filter[i];
- v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
- }
- val += (v2 - val) * (FELEML)frac / c->src_incr;
- } else {
- for (i = 0; i < c->filter_length; i++)
- val += src[sample_index + i] * (FELEM2)filter[i];
- }
-
-#ifdef CONFIG_RESAMPLE_FLT
- dst[dst_index] = av_clip_int16(lrintf(val));
-#else
- val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
- dst[dst_index] = av_clip_int16(val);
-#endif
- }
+ if (dst)
+ c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);
frac += dst_incr_frac;
index += dst_incr;
@@ -451,8 +441,8 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
/* resample each channel plane */
for (ch = 0; ch < c->buffer->channels; ch++) {
- out_samples = resample(c, (int16_t *)dst->data[ch],
- (const int16_t *)c->buffer->data[ch], consumed,
+ out_samples = resample(c, (void *)dst->data[ch],
+ (const void *)c->buffer->data[ch], consumed,
c->buffer->nb_samples, dst->allocated_samples,
ch + 1 == c->buffer->channels);
}
View
102 libavresample/resample_template.c
@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * 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
+ */
+
+#if defined(CONFIG_RESAMPLE_DBL)
+#define SET_TYPE(func) func ## _dbl
+#define FELEM double
+#define FELEM2 double
+#define FELEML double
+#define OUT(d, v) d = v
+#define DBL_TO_FELEM(d, v) d = v
+#elif defined(CONFIG_RESAMPLE_FLT)
+#define SET_TYPE(func) func ## _flt
+#define FELEM float
+#define FELEM2 float
+#define FELEML float
+#define OUT(d, v) d = v
+#define DBL_TO_FELEM(d, v) d = v
+#elif defined(CONFIG_RESAMPLE_S32)
+#define SET_TYPE(func) func ## _s32
+#define FELEM int32_t
+#define FELEM2 int64_t
+#define FELEML int64_t
+#define OUT(d, v) d = av_clipl_int32((v + (1 << 29)) >> 30)
+#define DBL_TO_FELEM(d, v) d = av_clipl_int32(llrint(v * (1 << 30)));
+#else
+#define SET_TYPE(func) func ## _s16
+#define FELEM int16_t
+#define FELEM2 int32_t
+#define FELEML int64_t
+#define OUT(d, v) d = av_clip_int16((v + (1 << 14)) >> 15)
+#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15)))
+#endif
+
+static void SET_TYPE(resample_one)(ResampleContext *c, int no_filter,
+ void *dst0, int dst_index, const void *src0,
+ int src_size, int index, int frac)
+{
+ FELEM *dst = dst0;
+ const FELEM *src = src0;
+
+ if (no_filter) {
+ dst[dst_index] = src[index];
+ } else {
+ int i;
+ int sample_index = index >> c->phase_shift;
+ FELEM2 val = 0;
+ FELEM *filter = ((FELEM *)c->filter_bank) +
+ c->filter_length * (index & c->phase_mask);
+
+ if (sample_index < 0) {
+ for (i = 0; i < c->filter_length; i++)
+ val += src[FFABS(sample_index + i) % src_size] *
+ (FELEM2)filter[i];
+ } else if (c->linear) {
+ FELEM2 v2 = 0;
+ for (i = 0; i < c->filter_length; i++) {
+ val += src[abs(sample_index + i)] * (FELEM2)filter[i];
+ v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
+ }
+ val += (v2 - val) * (FELEML)frac / c->src_incr;
+ } else {
+ for (i = 0; i < c->filter_length; i++)
+ val += src[sample_index + i] * (FELEM2)filter[i];
+ }
+
+ OUT(dst[dst_index], val);
+ }
+}
+
+static void SET_TYPE(set_filter)(void *filter0, double *tab, int phase,
+ int tap_count)
+{
+ int i;
+ FELEM *filter = ((FELEM *)filter0) + phase * tap_count;
+ for (i = 0; i < tap_count; i++) {
+ DBL_TO_FELEM(filter[i], tab[i]);
+ }
+}
+
+#undef SET_TYPE
+#undef FELEM
+#undef FELEM2
+#undef FELEML
+#undef OUT
+#undef DBL_TO_FELEM
View
47 libavresample/utils.c
@@ -57,18 +57,43 @@ int avresample_open(AVAudioResampleContext *avr)
avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
avr->force_resampling;
- /* set sample format conversion parameters */
- /* override user-requested internal format to avoid unexpected failures
- TODO: support more internal formats */
- if (avr->resample_needed && avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
- av_log(avr, AV_LOG_WARNING, "Using s16p as internal sample format\n");
- avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
- } else if (avr->mixing_needed &&
- avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
- avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
- av_log(avr, AV_LOG_WARNING, "Using fltp as internal sample format\n");
- avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
+ /* select internal sample format if not specified by the user */
+ if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
+ (avr->mixing_needed || avr->resample_needed)) {
+ enum AVSampleFormat in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
+ enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
+ int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
+ av_get_bytes_per_sample(out_fmt));
+ if (max_bps <= 2) {
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
+ } else if (avr->mixing_needed) {
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
+ } else {
+ if (max_bps <= 4) {
+ if (in_fmt == AV_SAMPLE_FMT_S32P ||
+ out_fmt == AV_SAMPLE_FMT_S32P) {
+ if (in_fmt == AV_SAMPLE_FMT_FLTP ||
+ out_fmt == AV_SAMPLE_FMT_FLTP) {
+ /* if one is s32 and the other is flt, use dbl */
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
+ } else {
+ /* if one is s32 and the other is s32, s16, or u8, use s32 */
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
+ }
+ } else {
+ /* if one is flt and the other is flt, s16 or u8, use flt */
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
+ }
+ } else {
+ /* if either is dbl, use dbl */
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
+ }
+ }
+ av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
+ av_get_sample_fmt_name(avr->internal_sample_fmt));
}
+
+ /* set sample format conversion parameters */
if (avr->in_channels == 1)
avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);