Merge remote-tracking branch 'qatar/master'

* qatar/master:
  lavfi: autoinsert resample filter when necessary.
  lavfi: add lavr-based audio resampling filter.
  x86: vc1: drop MMX loop filter implementation, which uses MMX2 instructions.

Conflicts:
	configure
	doc/filters.texi
	libavcodec/x86/vc1dsp_mmx.c
	libavfilter/Makefile
	libavfilter/allfilters.c
	libavfilter/avfiltergraph.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

configure

@@ -1690,6 +1690,7 @@ movie_filter_deps="avcodec avformat"
 mp_filter_deps="gpl avcodec swscale postproc"
 mptestsrc_filter_deps="gpl"
 negate_filter_deps="lut_filter"
+resample_filter_deps="avresample"
 ocv_filter_deps="libopencv"
 pan_filter_deps="swresample"
 removelogo_filter_deps="avcodec avformat swscale"

doc/filters.texi

@@ -502,6 +502,10 @@ volume=-12dB
 @end example
 @end itemize
 
+@section resample
+Convert the audio sample format, sample rate and channel layout. This filter is
+not meant to be used directly.
+
 @c man end AUDIO FILTERS
 
 @chapter Audio Sources

libavcodec/x86/vc1dsp_mmx.c

@@ -701,7 +701,6 @@ static void vc1_h_loop_filter16_ ## EXT(uint8_t *src, int stride, int pq) \
 }
 
 #if HAVE_YASM
-LOOP_FILTER(mmx)
 LOOP_FILTER(mmx2)
 LOOP_FILTER(sse2)
 LOOP_FILTER(ssse3)
@@ -803,7 +802,6 @@ void ff_vc1dsp_init_mmx(VC1DSPContext *dsp)
 
 #if HAVE_YASM
     if (mm_flags & AV_CPU_FLAG_MMX) {
-        ASSIGN_LF(mmx);
     }
     return;
     if (mm_flags & AV_CPU_FLAG_MMX2) {

libavcodec/x86/vc1dsp_yasm.asm

@@ -227,13 +227,6 @@ section .text
     imul r2, 0x01010101
 %endmacro
 
-; I do not know why the sign extension is needed...
-%macro PSIGNW_SRA_MMX 2
-    psraw %2, 15
-    PSIGNW_MMX %1, %2
-%endmacro
-
-
 %macro VC1_LF_MMX 1
 INIT_MMX
 cglobal vc1_v_loop_filter_internal_%1
@@ -274,10 +267,6 @@ cglobal vc1_h_loop_filter8_%1, 3,5,0
     RET
 %endmacro
 
-%define PABSW PABSW_MMX
-%define PSIGNW PSIGNW_SRA_MMX
-VC1_LF_MMX mmx
-
 %define PABSW PABSW_MMX2
 VC1_LF_MMX mmx2
 

libavfilter/Makefile

@@ -2,6 +2,7 @@ include $(SUBDIR)../config.mak
 
 NAME = avfilter
 FFLIBS = avutil swscale
+FFLIBS-$(CONFIG_RESAMPLE_FILTER) += avresample
 
 FFLIBS-$(CONFIG_ACONVERT_FILTER)             += swresample
 FFLIBS-$(CONFIG_AMOVIE_FILTER)               += avformat avcodec
@@ -48,6 +49,7 @@ OBJS-$(CONFIG_ASPLIT_FILTER)                 += af_asplit.o
 OBJS-$(CONFIG_ASTREAMSYNC_FILTER)            += af_astreamsync.o
 OBJS-$(CONFIG_EARWAX_FILTER)                 += af_earwax.o
 OBJS-$(CONFIG_PAN_FILTER)                    += af_pan.o
+OBJS-$(CONFIG_RESAMPLE_FILTER)               += af_resample.o
 OBJS-$(CONFIG_SILENCEDETECT_FILTER)          += af_silencedetect.o
 OBJS-$(CONFIG_VOLUME_FILTER)                 += af_volume.o
 

libavfilter/af_resample.c

+/*
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * sample format and channel layout conversion audio filter
+ */
+
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/opt.h"
+
+#include "libavresample/avresample.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct ResampleContext {
+    AVAudioResampleContext *avr;
+
+    int64_t next_pts;
+} ResampleContext;
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    ResampleContext *s = ctx->priv;
+
+    if (s->avr) {
+        avresample_close(s->avr);
+        avresample_free(&s->avr);
+    }
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterLink *inlink  = ctx->inputs[0];
+    AVFilterLink *outlink = ctx->outputs[0];
+
+    AVFilterFormats        *in_formats      = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
+    AVFilterFormats        *out_formats     = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
+
+    avfilter_formats_ref(in_formats,  &inlink->out_formats);
+    avfilter_formats_ref(out_formats, &outlink->in_formats);
+
+    return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AVFilterLink *inlink = ctx->inputs[0];
+    ResampleContext   *s = ctx->priv;
+    char buf1[64], buf2[64];
+    int ret;
+
+    if (s->avr) {
+        avresample_close(s->avr);
+        avresample_free(&s->avr);
+    }
+
+    if (inlink->channel_layout == outlink->channel_layout &&
+        inlink->sample_rate    == outlink->sample_rate    &&
+        inlink->format         == outlink->format)
+        return 0;
+
+    if (!(s->avr = avresample_alloc_context()))
+        return AVERROR(ENOMEM);
+
+    av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);
+    av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
+    av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);
+    av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);
+    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;
+
+    outlink->time_base = (AVRational){ 1, outlink->sample_rate };
+    s->next_pts        = AV_NOPTS_VALUE;
+
+    av_get_channel_layout_string(buf1, sizeof(buf1),
+                                 -1, inlink ->channel_layout);
+    av_get_channel_layout_string(buf2, sizeof(buf2),
+                                 -1, outlink->channel_layout);
+    av_log(ctx, AV_LOG_VERBOSE,
+           "fmt:%s srate: %d cl:%s -> fmt:%s srate: %d cl:%s\n",
+           av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
+           av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
+
+    return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    ResampleContext   *s = ctx->priv;
+    int ret = avfilter_request_frame(ctx->inputs[0]);
+
+    /* flush the lavr delay buffer */
+    if (ret == AVERROR_EOF && s->avr) {
+        AVFilterBufferRef *buf;
+        int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
+                                        outlink->sample_rate,
+                                        ctx->inputs[0]->sample_rate,
+                                        AV_ROUND_UP);
+
+        if (!nb_samples)
+            return ret;
+
+        buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+        if (!buf)
+            return AVERROR(ENOMEM);
+
+        ret = avresample_convert(s->avr, (void**)buf->extended_data,
+                                 buf->linesize[0], nb_samples,
+                                 NULL, 0, 0);
+        if (ret <= 0) {
+            avfilter_unref_buffer(buf);
+            return (ret == 0) ? AVERROR_EOF : ret;
+        }
+
+        buf->pts = s->next_pts;
+        ff_filter_samples(outlink, buf);
+        return 0;
+    }
+    return ret;
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+    AVFilterContext  *ctx = inlink->dst;
+    ResampleContext    *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+
+    if (s->avr) {
+        AVFilterBufferRef *buf_out;
+        int delay, nb_samples, ret;
+
+        /* maximum possible samples lavr can output */
+        delay      = avresample_get_delay(s->avr);
+        nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
+                                    outlink->sample_rate, inlink->sample_rate,
+                                    AV_ROUND_UP);
+
+        buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+        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);
+
+        av_assert0(!avresample_available(s->avr));
+
+        if (s->next_pts == AV_NOPTS_VALUE) {
+            if (buf->pts == AV_NOPTS_VALUE) {
+                av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
+                       "assuming 0.\n");
+                s->next_pts = 0;
+            } else
+                s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
+                                           outlink->time_base);
+        }
+
+        if (ret > 0) {
+            buf_out->audio->nb_samples = ret;
+            if (buf->pts != AV_NOPTS_VALUE) {
+                buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
+                                            outlink->time_base) -
+                               av_rescale(delay, outlink->sample_rate,
+                                          inlink->sample_rate);
+            } else
+                buf_out->pts = s->next_pts;
+
+            s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
+
+            ff_filter_samples(outlink, buf_out);
+        }
+        avfilter_unref_buffer(buf);
+    } else
+        ff_filter_samples(outlink, buf);
+}
+
+AVFilter avfilter_af_resample = {
+    .name          = "resample",
+    .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
+    .priv_size     = sizeof(ResampleContext),
+
+    .uninit         = uninit,
+    .query_formats  = query_formats,
+
+    .inputs    = (const AVFilterPad[]) {{ .name            = "default",
+                                          .type            = AVMEDIA_TYPE_AUDIO,
+                                          .filter_samples  = filter_samples,
+                                          .min_perms       = AV_PERM_READ },
+                                        { .name = NULL}},
+    .outputs   = (const AVFilterPad[]) {{ .name          = "default",
+                                          .type          = AVMEDIA_TYPE_AUDIO,
+                                          .config_props  = config_output,
+                                          .request_frame = request_frame },
+                                        { .name = NULL}},
+};

libavfilter/allfilters.c

@@ -46,6 +46,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER (PAN,         pan,         af);
     REGISTER_FILTER (SILENCEDETECT, silencedetect, af);
     REGISTER_FILTER (VOLUME,      volume,      af);
+    REGISTER_FILTER (RESAMPLE,    resample,    af);
 
     REGISTER_FILTER (ABUFFER,     abuffer,     asrc);
     REGISTER_FILTER (AEVALSRC,    aevalsrc,    asrc);