Merge remote-tracking branch 'qatar/master'

* qatar/master:
  FATE: use updated reference for aac-latm_stereo_to_51
  avconv: use libavresample
  Add libavresample
  FATE: avoid channel mixing in lavf-dv_fmt

Conflicts:
	Changelog
	Makefile
	cmdutils.c
	configure
	doc/APIchanges
	ffmpeg.c
	tests/lavf-regression.sh
	tests/ref/lavf/dv_fmt
	tests/ref/seek/lavf_dv
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Changelog

@@ -26,6 +26,7 @@ version next:
 - drawtext video filter: fontconfig support
 - ffmpeg -benchmark_all option
 - super2xsai filter ported from libmpcodecs
+- add libavresample audio conversion library for compatibility
 
 
 version 0.10:

Makefile

@@ -31,6 +31,7 @@ ALLMANPAGES = $(BASENAMES:%=%.1)
 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
 FFLIBS-$(CONFIG_AVFORMAT) += avformat
+FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
 FFLIBS-$(CONFIG_AVCODEC)  += avcodec
 FFLIBS-$(CONFIG_POSTPROC) += postproc
 FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample

cmdutils.c

@@ -32,6 +32,7 @@
 #include "libavformat/avformat.h"
 #include "libavfilter/avfilter.h"
 #include "libavdevice/avdevice.h"
+#include "libavresample/avresample.h"
 #include "libswscale/swscale.h"
 #include "libswresample/swresample.h"
 #if CONFIG_POSTPROC
@@ -633,7 +634,8 @@ static int warned_cfg = 0;
         const char *indent = flags & INDENT? "  " : "";                 \
         if (flags & SHOW_VERSION) {                                     \
             unsigned int version = libname##_version();                 \
-            av_log(NULL, level, "%slib%-11s %2d.%3d.%3d / %2d.%3d.%3d\n",\
+            av_log(NULL, level,                                         \
+                   "%slib%-11s %2d.%3d.%3d / %2d.%3d.%3d\n",            \
                    indent, #libname,                                    \
                    LIB##LIBNAME##_VERSION_MAJOR,                        \
                    LIB##LIBNAME##_VERSION_MINOR,                        \
@@ -662,6 +664,7 @@ static void print_all_libs_info(int flags, int level)
     PRINT_LIB_INFO(avformat, AVFORMAT, flags, level);
     PRINT_LIB_INFO(avdevice, AVDEVICE, flags, level);
     PRINT_LIB_INFO(avfilter, AVFILTER, flags, level);
+//    PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level);
     PRINT_LIB_INFO(swscale,  SWSCALE,  flags, level);
     PRINT_LIB_INFO(swresample,SWRESAMPLE,  flags, level);
 #if CONFIG_POSTPROC

common.mak

@@ -20,7 +20,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
 
-ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
+ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
 
 # NASM requires -I path terminated with /
 IFLAGS     := -I. -I$(SRC_PATH)/

configure

@@ -112,6 +112,7 @@ Component options:
   --disable-swscale        disable libswscale build
   --disable-postproc       disable libpostproc build
   --disable-avfilter       disable video filter support [no]
+  --disable-avresample     disable libavresample build [no]
   --disable-pthreads       disable pthreads [auto]
   --disable-w32threads     disable Win32 threads [auto]
   --disable-os2threads     disable OS/2 threads [auto]
@@ -1013,6 +1014,7 @@ CONFIG_LIST="
     avdevice
     avfilter
     avformat
+    avresample
     avisynth
     bzlib
     crystalhd
@@ -1870,6 +1872,7 @@ enable avcodec
 enable avdevice
 enable avfilter
 enable avformat
+enable avresample
 enable avutil
 enable postproc
 enable stripping
@@ -3724,6 +3727,7 @@ get_version LIBAVCODEC  libavcodec/version.h
 get_version LIBAVDEVICE libavdevice/avdevice.h
 get_version LIBAVFILTER libavfilter/version.h
 get_version LIBAVFORMAT libavformat/version.h
+get_version LIBAVRESAMPLE libavresample/version.h
 get_version LIBAVUTIL   libavutil/avutil.h
 get_version LIBPOSTPROC libpostproc/postprocess.h
 get_version LIBSWRESAMPLE libswresample/swresample.h
@@ -3869,5 +3873,6 @@ pkgconfig_generate libavformat "FFmpeg container format library" "$LIBAVFORMAT_V
 pkgconfig_generate libavdevice "FFmpeg device handling library" "$LIBAVDEVICE_VERSION" "$extralibs" "$libavdevice_pc_deps"
 pkgconfig_generate libavfilter "FFmpeg video filtering library" "$LIBAVFILTER_VERSION" "$extralibs" "$libavfilter_pc_deps"
 pkgconfig_generate libpostproc "FFmpeg postprocessing library" "$LIBPOSTPROC_VERSION" "" "libavutil = $LIBAVUTIL_VERSION"
+pkgconfig_generate libavresample "Libav audio resampling library" "$LIBAVRESAMPLE_VERSION" "$extralibs"
 pkgconfig_generate libswscale "FFmpeg image rescaling library" "$LIBSWSCALE_VERSION" "$LIBM" "libavutil = $LIBAVUTIL_VERSION"
 pkgconfig_generate libswresample "FFmpeg audio rescaling library" "$LIBSWRESAMPLE_VERSION" "$LIBM" "libavutil = $LIBAVUTIL_VERSION"

doc/APIchanges

@@ -6,6 +6,7 @@ libavcodec:  2012-01-27
 libavdevice: 2011-04-18
 libavfilter: 2011-04-18
 libavformat: 2012-01-27
+libavresample: 2012-xx-xx
 libpostproc: 2011-04-18
 libswscale:  2011-06-20
 libavutil:   2011-04-18
@@ -22,6 +23,9 @@ API changes, most recent first:
 2012-03-26 - a67d9cf - lavfi 2.66.100
   Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.
 
+2012-xx-xx - xxxxxxx - lavr 0.0.0
+  Add libavresample audio conversion library
+
 2012-xx-xx - xxxxxxx - lavu 51.28.0 - audio_fifo.h
   Add audio FIFO functions:
     av_audio_fifo_free()

ffmpeg.c

@@ -36,7 +36,6 @@
 #include "libavdevice/avdevice.h"
 #include "libswscale/swscale.h"
 #include "libavutil/opt.h"
-#include "libavcodec/audioconvert.h"
 #include "libavutil/audioconvert.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/samplefmt.h"
@@ -300,6 +299,7 @@ typedef struct OutputStream {
     int audio_channels_mapped;           ///< number of channels in audio_channels_map
     int resample_sample_fmt;
     int resample_channels;
+    uint64_t resample_channel_layout;
     int resample_sample_rate;
     float rematrix_volume;
     AVFifoBuffer *fifo;     /* for compression: one audio fifo per codec */
@@ -1525,7 +1525,7 @@ static int encode_audio_frame(AVFormatContext *s, OutputStream *ost,
 }
 
 static int alloc_audio_output_buf(AVCodecContext *dec, AVCodecContext *enc,
-                                  int nb_samples)
+                                  int nb_samples, int *buf_linesize)
 {
     int64_t audio_buf_samples;
     int audio_buf_size;
@@ -1538,7 +1538,7 @@ static int alloc_audio_output_buf(AVCodecContext *dec, AVCodecContext *enc,
     if (audio_buf_samples > INT_MAX)
         return AVERROR(EINVAL);
 
-    audio_buf_size = av_samples_get_buffer_size(NULL, enc->channels,
+    audio_buf_size = av_samples_get_buffer_size(buf_linesize, enc->channels,
                                                 audio_buf_samples,
                                                 enc->sample_fmt, 0);
     if (audio_buf_size < 0)
@@ -1557,7 +1557,7 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
     uint8_t *buftmp;
     int64_t size_out;
 
-    int frame_bytes, resample_changed;
+    int frame_bytes, resample_changed, ret;
     AVCodecContext *enc = ost->st->codec;
     AVCodecContext *dec = ist->st->codec;
     int osize = av_get_bytes_per_sample(enc->sample_fmt);
@@ -1566,37 +1566,46 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
     int size     = decoded_frame->nb_samples * dec->channels * isize;
     int planes   = av_sample_fmt_is_planar(dec->sample_fmt) ? dec->channels : 1;
     int i;
+    int out_linesize = 0;
+    int buf_linesize = decoded_frame->linesize[0];
 
     av_assert0(planes <= AV_NUM_DATA_POINTERS);
 
     for(i=0; i<planes; i++)
         buf[i]= decoded_frame->data[i];
 
+
     get_default_channel_layouts(ost, ist);
 
-    if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples) < 0) {
+    if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples, &out_linesize) < 0) {
         av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n");
         exit_program(1);
     }
 
-    if (enc->channels != dec->channels
-     || enc->sample_fmt != dec->sample_fmt
-     || enc->sample_rate!= dec->sample_rate
-    )
+    if (audio_sync_method > 1                      ||
+        enc->channels       != dec->channels       ||
+        enc->channel_layout != dec->channel_layout ||
+        enc->sample_rate    != dec->sample_rate    ||
+        dec->sample_fmt     != enc->sample_fmt)
         ost->audio_resample = 1;
 
     resample_changed = ost->resample_sample_fmt  != dec->sample_fmt ||
                        ost->resample_channels    != dec->channels   ||
+                       ost->resample_channel_layout != dec->channel_layout ||
                        ost->resample_sample_rate != dec->sample_rate;
 
     if ((ost->audio_resample && !ost->swr) || resample_changed || ost->audio_channels_mapped) {
+
         if (resample_changed) {
-            av_log(NULL, AV_LOG_INFO, "Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d to rate:%d fmt:%s ch:%d\n",
+            av_log(NULL, AV_LOG_INFO, "Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d chl:0x%"PRIx64" to rate:%d fmt:%s ch:%d chl:0x%"PRIx64"\n",
                    ist->file_index, ist->st->index,
-                   ost->resample_sample_rate, av_get_sample_fmt_name(ost->resample_sample_fmt), ost->resample_channels,
-                   dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt), dec->channels);
+                   ost->resample_sample_rate, av_get_sample_fmt_name(ost->resample_sample_fmt),
+                   ost->resample_channels, ost->resample_channel_layout,
+                   dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt),
+                   dec->channels, dec->channel_layout);
             ost->resample_sample_fmt  = dec->sample_fmt;
             ost->resample_channels    = dec->channels;
+            ost->resample_channel_layout = dec->channel_layout;
             ost->resample_sample_rate = dec->sample_rate;
             swr_free(&ost->swr);
         }
@@ -1604,6 +1613,7 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
         if (audio_sync_method <= 1 && !ost->audio_channels_mapped &&
             ost->resample_sample_fmt  == enc->sample_fmt &&
             ost->resample_channels    == enc->channels   &&
+            ost->resample_channel_layout == enc->channel_layout &&
             ost->resample_sample_rate == enc->sample_rate) {
             //ost->swr = NULL;
             ost->audio_resample = 0;
@@ -1673,7 +1683,7 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
                         exit_program(1);
                     }
 
-                    if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples + idelta) < 0) {
+                    if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples + idelta, &out_linesize) < 0) {
                         av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n");
                         exit_program(1);
                     }
@@ -1686,11 +1696,11 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
                         buf[i] = t;
                     }
                     size += byte_delta;
+                    buf_linesize = allocated_async_buf_size;
                     av_log(NULL, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", idelta);
                 }
             } else if (audio_sync_method > 1) {
                 int comp = av_clip(delta, -audio_sync_method, audio_sync_method);
-                av_assert0(ost->audio_resample);
                 av_log(NULL, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n",
                        delta, comp, enc->sample_rate);
 //                fprintf(stderr, "drift:%f len:%d opts:%"PRId64" ipts:%"PRId64" fifo:%d\n", delta, -1, ost->sync_opts, (int64_t)(get_sync_ipts(ost) * enc->sample_rate), av_fifo_size(ost->fifo)/(ost->st->codec->channels * 2));
@@ -1703,8 +1713,10 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
 
     if (ost->audio_resample || ost->audio_channels_mapped) {
         buftmp = audio_buf;
-        size_out = swr_convert(ost->swr, (      uint8_t*[]){buftmp}, allocated_audio_buf_size / (enc->channels * osize),
-                                         buf, size / (dec->channels * isize));
+        size_out = swr_convert(ost->swr, (      uint8_t*[]){buftmp},
+                                      allocated_audio_buf_size / (enc->channels * osize),
+                                      buf,
+                                      size / (dec->channels * isize));
         if (size_out < 0) {
             av_log(NULL, AV_LOG_FATAL, "swr_convert failed\n");
             exit_program(1);
@@ -3078,6 +3090,7 @@ static int transcode_init(void)
                 if (!ost->fifo) {
                     return AVERROR(ENOMEM);
                 }
+
                 if (!codec->sample_rate)
                     codec->sample_rate = icodec->sample_rate;
                 choose_sample_rate(ost->st, ost->enc);
@@ -3110,13 +3123,15 @@ static int transcode_init(void)
                 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;
-                ost->audio_resample      |=    codec->sample_fmt     != icodec->sample_fmt
-                                            || codec->channel_layout != icodec->channel_layout;
-                icodec->request_channels  = codec->channels;
+
+//                 ost->audio_resample       = codec->sample_rate != icodec->sample_rate || audio_sync_method > 1;
+//                 ost->audio_resample      |=    codec->sample_fmt     != icodec->sample_fmt
+//                                             || codec->channel_layout != icodec->channel_layout;
+                icodec->request_channels  = codec-> channels;
                 ost->resample_sample_fmt  = icodec->sample_fmt;
                 ost->resample_sample_rate = icodec->sample_rate;
                 ost->resample_channels    = icodec->channels;
+                ost->resample_channel_layout = icodec->channel_layout;
                 break;
             case AVMEDIA_TYPE_VIDEO:
                 if (!ost->filter) {

libavresample/Makefile

+NAME = avresample
+FFLIBS = avutil
+
+HEADERS = avresample.h                                                  \
+          version.h
+
+OBJS = audio_convert.o                                                  \
+       audio_data.o                                                     \
+       audio_mix.o                                                      \
+       audio_mix_matrix.o                                               \
+       options.o                                                        \
+       resample.o                                                       \
+       utils.o
+
+TESTPROGS = avresample

libavresample/audio_convert.h

+/*
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * 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
+ */
+
+#ifndef AVRESAMPLE_AUDIO_CONVERT_H
+#define AVRESAMPLE_AUDIO_CONVERT_H
+
+#include "libavutil/samplefmt.h"
+#include "avresample.h"
+#include "audio_data.h"
+
+typedef struct AudioConvert AudioConvert;
+
+/**
+ * Set conversion function if the parameters match.
+ *
+ * This compares the parameters of the conversion function to the parameters
+ * in the AudioConvert context. If the parameters do not match, no changes are
+ * made to the active functions. If the parameters do match and the alignment
+ * is not constrained, the function is set as the generic conversion function.
+ * If the parameters match and the alignment is constrained, the function is
+ * set as the optimized conversion function.
+ *
+ * @param ac             AudioConvert context
+ * @param out_fmt        output sample format
+ * @param in_fmt         input sample format
+ * @param channels       number of channels, or 0 for any number of channels
+ * @param ptr_align      buffer pointer alignment, in bytes
+ * @param sample_align   buffer size alignment, in samples
+ * @param descr          function type description (e.g. "C" or "SSE")
+ * @param conv           conversion function pointer
+ */
+void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
+                               enum AVSampleFormat in_fmt, int channels,
+                               int ptr_align, int samples_align,
+                               const char *descr, void *conv);
+
+/**
+ * Allocate and initialize AudioConvert context for sample format conversion.
+ *
+ * @param avr      AVAudioResampleContext
+ * @param out_fmt  output sample format
+ * @param in_fmt   input sample format
+ * @param channels number of channels
+ * @return         newly-allocated AudioConvert context
+ */
+AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
+                                     enum AVSampleFormat out_fmt,
+                                     enum AVSampleFormat in_fmt,
+                                     int channels);
+
+/**
+ * Convert audio data from one sample format to another.
+ *
+ * For each call, the alignment of the input and output AudioData buffers are
+ * examined to determine whether to use the generic or optimized conversion
+ * function (when available).
+ *
+ * @param ac     AudioConvert context
+ * @param out    output audio data
+ * @param in     input audio data
+ * @param len    number of samples to convert
+ * @return       0 on success, negative AVERROR code on failure
+ */
+int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in, int len);
+
+/* arch-specific initialization functions */
+
+void ff_audio_convert_init_x86(AudioConvert *ac);
+
+#endif /* AVRESAMPLE_AUDIO_CONVERT_H */

libavresample/audio_data.h

+/*
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * 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
+ */
+
+#ifndef AVRESAMPLE_AUDIO_DATA_H
+#define AVRESAMPLE_AUDIO_DATA_H
+
+#include <stdint.h>
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/log.h"
+#include "libavutil/samplefmt.h"
+#include "avresample.h"
+
+/**
+ * Audio buffer used for intermediate storage between conversion phases.
+ */
+typedef struct AudioData {
+    const AVClass *class;               /**< AVClass for logging            */
+    uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers        */
+    uint8_t *buffer;                    /**< data buffer                    */
+    unsigned int buffer_size;           /**< allocated buffer size          */
+    int allocated_samples;              /**< number of samples the buffer can hold */
+    int nb_samples;                     /**< current number of samples      */
+    enum AVSampleFormat sample_fmt;     /**< sample format                  */
+    int channels;                       /**< channel count                  */
+    int allocated_channels;             /**< allocated channel count        */
+    int is_planar;                      /**< sample format is planar        */
+    int planes;                         /**< number of data planes          */
+    int sample_size;                    /**< bytes per sample               */
+    int stride;                         /**< sample byte offset within a plane */
+    int read_only;                      /**< data is read-only              */
+    int allow_realloc;                  /**< realloc is allowed             */
+    int ptr_align;                      /**< minimum data pointer alignment */
+    int samples_align;                  /**< allocated samples alignment    */
+    const char *name;                   /**< name for debug logging         */
+} AudioData;
+
+int ff_audio_data_set_channels(AudioData *a, int channels);
+
+/**
+ * Initialize AudioData using a given source.
+ *
+ * This does not allocate an internal buffer. It only sets the data pointers
+ * and audio parameters.
+ *
+ * @param a               AudioData struct
+ * @param src             source data pointers
+ * @param plane_size      plane size, in bytes.
+ *                        This can be 0 if unknown, but that will lead to
+ *                        optimized functions not being used in many cases,
+ *                        which could slow down some conversions.
+ * @param channels        channel count
+ * @param nb_samples      number of samples in the source data
+ * @param sample_fmt      sample format
+ * @param read_only       indicates if buffer is read only or read/write
+ * @param name            name for debug logging (can be NULL)
+ * @return                0 on success, negative AVERROR value on error
+ */
+int ff_audio_data_init(AudioData *a, void **src, int plane_size, int channels,
+                       int nb_samples, enum AVSampleFormat sample_fmt,
+                       int read_only, const char *name);
+
+/**
+ * Allocate AudioData.
+ *
+ * This allocates an internal buffer and sets audio parameters.
+ *
+ * @param channels        channel count
+ * @param nb_samples      number of samples to allocate space for
+ * @param sample_fmt      sample format
+ * @param name            name for debug logging (can be NULL)
+ * @return                newly allocated AudioData struct, or NULL on error
+ */
+AudioData *ff_audio_data_alloc(int channels, int nb_samples,
+                               enum AVSampleFormat sample_fmt,
+                               const char *name);
+
+/**
+ * Reallocate AudioData.
+ *
+ * The AudioData must have been previously allocated with ff_audio_data_alloc().
+ *
+ * @param a           AudioData struct
+ * @param nb_samples  number of samples to allocate space for
+ * @return            0 on success, negative AVERROR value on error
+ */
+int ff_audio_data_realloc(AudioData *a, int nb_samples);
+
+/**
+ * Free AudioData.
+ *
+ * The AudioData must have been previously allocated with ff_audio_data_alloc().
+ *
+ * @param a  AudioData struct
+ */
+void ff_audio_data_free(AudioData **a);
+
+/**
+ * Copy data from one AudioData to another.
+ *
+ * @param out  output AudioData
+ * @param in   input AudioData
+ * @return     0 on success, negative AVERROR value on error
+ */
+int ff_audio_data_copy(AudioData *out, AudioData *in);
+
+/**
+ * Append data from one AudioData to the end of another.
+ *
+ * @param dst         destination AudioData
+ * @param dst_offset  offset, in samples, to start writing, relative to the
+ *                    start of dst
+ * @param src         source AudioData
+ * @param src_offset  offset, in samples, to start copying, relative to the
+ *                    start of the src
+ * @param nb_samples  number of samples to copy
+ * @return            0 on success, negative AVERROR value on error
+ */
+int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
+                          int src_offset, int nb_samples);
+
+/**
+ * Drain samples from the start of the AudioData.
+ *
+ * Remaining samples are shifted to the start of the AudioData.
+ *
+ * @param a           AudioData struct
+ * @param nb_samples  number of samples to drain
+ */
+void ff_audio_data_drain(AudioData *a, int nb_samples);
+
+/**
+ * Add samples in AudioData to an AVAudioFifo.
+ *
+ * @param af          Audio FIFO Buffer
+ * @param a           AudioData struct
+ * @param offset      number of samples to skip from the start of the data
+ * @param nb_samples  number of samples to add to the FIFO
+ * @return            number of samples actually added to the FIFO, or
+ *                    negative AVERROR code on error
+ */
+int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
+                              int nb_samples);
+
+/**
+ * Read samples from an AVAudioFifo to AudioData.
+ *
+ * @param af          Audio FIFO Buffer
+ * @param a           AudioData struct
+ * @param nb_samples  number of samples to read from the FIFO
+ * @return            number of samples actually read from the FIFO, or
+ *                    negative AVERROR code on error
+ */
+int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples);
+
+#endif /* AVRESAMPLE_AUDIO_DATA_H */

libavresample/audio_mix.h

+/*
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * 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
+ */
+
+#ifndef AVRESAMPLE_AUDIO_MIX_H
+#define AVRESAMPLE_AUDIO_MIX_H
+
+#include <stdint.h>
+
+#include "libavutil/samplefmt.h"
+#include "avresample.h"
+#include "audio_data.h"
+
+typedef void (mix_func)(uint8_t **src, void **matrix, int len, int out_ch,
+                        int in_ch);
+
+typedef struct AudioMix {
+    AVAudioResampleContext *avr;
+    enum AVSampleFormat fmt;
+    enum AVMixCoeffType coeff_type;
+    uint64_t in_layout;
+    uint64_t out_layout;
+    int in_channels;
+    int out_channels;
+
+    int ptr_align;
+    int samples_align;
+    int has_optimized_func;
+    const char *func_descr;
+    const char *func_descr_generic;
+    mix_func *mix;
+    mix_func *mix_generic;
+
+    int16_t *matrix_q6[AVRESAMPLE_MAX_CHANNELS];
+    int32_t *matrix_q15[AVRESAMPLE_MAX_CHANNELS];
+    float   *matrix_flt[AVRESAMPLE_MAX_CHANNELS];
+    void   **matrix;
+} AudioMix;
+
+/**
+ * Set mixing function if the parameters match.
+ *
+ * This compares the parameters of the mixing function to the parameters in the
+ * AudioMix context. If the parameters do not match, no changes are made to the
+ * active functions. If the parameters do match and the alignment is not
+ * constrained, the function is set as the generic mixing function. If the
+ * parameters match and the alignment is constrained, the function is set as
+ * the optimized mixing function.
+ *
+ * @param am             AudioMix context
+ * @param fmt            input/output sample format
+ * @param coeff_type     mixing coefficient type
+ * @param in_channels    number of input channels, or 0 for any number of channels
+ * @param out_channels   number of output channels, or 0 for any number of channels
+ * @param ptr_align      buffer pointer alignment, in bytes
+ * @param sample_align   buffer size alignment, in samples
+ * @param descr          function type description (e.g. "C" or "SSE")
+ * @param mix_func       mixing function pointer
+ */
+void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt,
+                           enum AVMixCoeffType coeff_type, int in_channels,
+                           int out_channels, int ptr_align, int samples_align,
+                           const char *descr, void *mix_func);
+
+/**
+ * Initialize the AudioMix context in the AVAudioResampleContext.
+ *
+ * The parameters in the AVAudioResampleContext are used to initialize the
+ * AudioMix context and set the mixing matrix.
+ *
+ * @param avr  AVAudioResampleContext
+ * @return     0 on success, negative AVERROR code on failure
+ */
+int ff_audio_mix_init(AVAudioResampleContext *avr);
+
+/**
+ * Close an AudioMix context.
+ *
+ * This clears and frees the mixing matrix arrays.
+ */
+void ff_audio_mix_close(AudioMix *am);
+
+/**
+ * Apply channel mixing to audio data using the current mixing matrix.
+ */
+int ff_audio_mix(AudioMix *am, AudioData *src);
+
+/* arch-specific initialization functions */
+
+void ff_audio_mix_init_x86(AudioMix *am);
+
+#endif /* AVRESAMPLE_AUDIO_MIX_H */

libavresample/internal.h

+/*
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * 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
+ */
+
+#ifndef AVRESAMPLE_INTERNAL_H
+#define AVRESAMPLE_INTERNAL_H
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/log.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+#include "avresample.h"
+#include "audio_convert.h"
+#include "audio_data.h"
+#include "audio_mix.h"
+#include "resample.h"
+
+struct AVAudioResampleContext {
+    const AVClass *av_class;        /**< AVClass for logging and AVOptions  */
+
+    uint64_t in_channel_layout;                 /**< input channel layout   */
+    enum AVSampleFormat in_sample_fmt;          /**< input sample format    */
+    int in_sample_rate;                         /**< input sample rate      */
+    uint64_t out_channel_layout;                /**< output channel layout  */
+    enum AVSampleFormat out_sample_fmt;         /**< output sample format   */
+    int out_sample_rate;                        /**< output sample rate     */
+    enum AVSampleFormat internal_sample_fmt;    /**< internal sample format */
+    enum AVMixCoeffType mix_coeff_type;         /**< mixing coefficient type */
+    double center_mix_level;                    /**< center mix level       */
+    double surround_mix_level;                  /**< surround mix level     */
+    double lfe_mix_level;                       /**< lfe mix level          */
+    int force_resampling;                       /**< force resampling       */
+    int filter_size;                            /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
+    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 */
+
+    int in_channels;        /**< number of input channels                   */
+    int out_channels;       /**< number of output channels                  */
+    int resample_channels;  /**< number of channels used for resampling     */
+    int downmix_needed;     /**< downmixing is needed                       */
+    int upmix_needed;       /**< upmixing is needed                         */
+    int mixing_needed;      /**< either upmixing or downmixing is needed    */
+    int resample_needed;    /**< resampling is needed                       */
+    int in_convert_needed;  /**< input sample format conversion is needed   */
+    int out_convert_needed; /**< output sample format conversion is needed  */
+
+    AudioData *in_buffer;           /**< buffer for converted input         */
+    AudioData *resample_out_buffer; /**< buffer for output from resampler   */
+    AudioData *out_buffer;          /**< buffer for converted output        */
+    AVAudioFifo *out_fifo;          /**< FIFO for output samples            */
+
+    AudioConvert *ac_in;        /**< input sample format conversion context  */
+    AudioConvert *ac_out;       /**< output sample format conversion context */
+    ResampleContext *resample;  /**< resampling context                      */
+    AudioMix *am;               /**< channel mixing context                  */
+};
+
+#endif /* AVRESAMPLE_INTERNAL_H */

libavresample/libavresample.v

+LIBAVRESAMPLE_$MAJOR {
+        global: av*;
+        local:  *;
+};

libavresample/options.c

+/*
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * 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
+ */
+
+#include "libavutil/mathematics.h"
+#include "libavutil/opt.h"
+#include "avresample.h"
+#include "internal.h"
+#include "audio_mix.h"
+
+/**
+ * @file
+ * Options definition for AVAudioResampleContext.
+ */
+
+#define OFFSET(x) offsetof(AVAudioResampleContext, x)
+#define PARAM AV_OPT_FLAG_AUDIO_PARAM
+
+static const AVOption options[] = {
+    { "in_channel_layout",      "Input Channel Layout",     OFFSET(in_channel_layout),      AV_OPT_TYPE_INT64,  { 0                     }, INT64_MIN,            INT64_MAX,              PARAM },
+    { "in_sample_fmt",          "Input Sample Format",      OFFSET(in_sample_fmt),          AV_OPT_TYPE_INT,    { AV_SAMPLE_FMT_S16     }, AV_SAMPLE_FMT_U8,     AV_SAMPLE_FMT_NB-1,     PARAM },
+    { "in_sample_rate",         "Input Sample Rate",        OFFSET(in_sample_rate),         AV_OPT_TYPE_INT,    { 48000                 }, 1,                    INT_MAX,                PARAM },
+    { "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 },
+    { "mix_coeff_type",         "Mixing Coefficient Type",  OFFSET(mix_coeff_type),         AV_OPT_TYPE_INT,    { AV_MIX_COEFF_TYPE_FLT }, AV_MIX_COEFF_TYPE_Q6, AV_MIX_COEFF_TYPE_NB-1, PARAM, "mix_coeff_type" },
+        { "q6",  "16-bit 10.6 Fixed-Point",  0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q6  }, 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" },
+        { "flt", "Floating-Point",           0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_FLT }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
+    { "center_mix_level",       "Center Mix Level",         OFFSET(center_mix_level),       AV_OPT_TYPE_DOUBLE, { M_SQRT1_2             }, -32.0,                32.0,                   PARAM },
+    { "surround_mix_level",     "Surround Mix Level",       OFFSET(surround_mix_level),     AV_OPT_TYPE_DOUBLE, { M_SQRT1_2             }, -32.0,                32.0,                   PARAM },
+    { "lfe_mix_level",          "LFE Mix Level",            OFFSET(lfe_mix_level),          AV_OPT_TYPE_DOUBLE, { 0.0                   }, -32.0,                32.0,                   PARAM },
+    { "force_resampling",       "Force Resampling",         OFFSET(force_resampling),       AV_OPT_TYPE_INT,    { 0                     }, 0,                    1,                      PARAM },
+    { "filter_size",            "Resampling Filter Size",   OFFSET(filter_size),            AV_OPT_TYPE_INT,    { 16                    }, 0,                    32, /* ??? */           PARAM },
+    { "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 },
+    { NULL },
+};
+
+static const AVClass av_resample_context_class = {
+    .class_name = "AVAudioResampleContext",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+AVAudioResampleContext *avresample_alloc_context(void)
+{
+    AVAudioResampleContext *avr;
+
+    avr = av_mallocz(sizeof(*avr));
+    if (!avr)
+        return NULL;
+
+    avr->av_class = &av_resample_context_class;
+    av_opt_set_defaults(avr);
+
+    avr->am = av_mallocz(sizeof(*avr->am));
+    if (!avr->am) {
+        av_free(avr);
+        return NULL;
+    }
+    avr->am->avr = avr;
+
+    return avr;
+}
+
+const AVClass *avresample_get_class(void)
+{
+    return &av_resample_context_class;
+}

libavresample/resample.h

+/*
+ * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * 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
+ */
+
+#ifndef AVRESAMPLE_RESAMPLE_H
+#define AVRESAMPLE_RESAMPLE_H
+
+#include "avresample.h"
+#include "audio_data.h"
+
+typedef struct ResampleContext ResampleContext;
+
+/**
+ * Allocate and initialize a ResampleContext.
+ *
+ * The parameters in the AVAudioResampleContext are used to initialize the
+ * ResampleContext.
+ *
+ * @param avr  AVAudioResampleContext
+ * @return     newly-allocated ResampleContext
+ */
+ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr);
+
+/**
+ * Free a ResampleContext.
+ *
+ * @param c  ResampleContext
+ */
+void ff_audio_resample_free(ResampleContext **c);
+
+/**
+ * Resample audio data.
+ *
+ * Changes the sample rate.
+ *
+ * @par
+ * All samples in the source data may not be consumed depending on the
+ * resampling parameters and the size of the output buffer. The unconsumed
+ * samples are automatically added to the start of the source in the next call.
+ * If the destination data can be reallocated, that may be done in this function
+ * in order to fit all available output. If it cannot be reallocated, fewer
+ * input samples will be consumed in order to have the output fit in the
+ * destination data buffers.
+ *
+ * @param c         ResampleContext
+ * @param dst       destination audio data
+ * @param src       source audio data
+ * @param consumed  number of samples consumed from the source
+ * @return          number of samples written to the destination
+ */
+int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
+                      int *consumed);
+
+#endif /* AVRESAMPLE_RESAMPLE_H */

libavresample/version.h

+/*
+ * 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
+ */
+
+#ifndef AVRESAMPLE_VERSION_H
+#define AVRESAMPLE_VERSION_H
+
+#define LIBAVRESAMPLE_VERSION_MAJOR  0
+#define LIBAVRESAMPLE_VERSION_MINOR  0
+#define LIBAVRESAMPLE_VERSION_MICRO  0
+
+#define LIBAVRESAMPLE_VERSION_INT  AV_VERSION_INT(LIBAVRESAMPLE_VERSION_MAJOR, \
+                                                  LIBAVRESAMPLE_VERSION_MINOR, \
+                                                  LIBAVRESAMPLE_VERSION_MICRO)
+#define LIBAVRESAMPLE_VERSION          AV_VERSION(LIBAVRESAMPLE_VERSION_MAJOR, \
+                                                  LIBAVRESAMPLE_VERSION_MINOR, \
+                                                  LIBAVRESAMPLE_VERSION_MICRO)
+#define LIBAVRESAMPLE_BUILD        LIBAVRESAMPLE_VERSION_INT
+
+#define LIBAVRESAMPLE_IDENT        "Lavr" AV_STRINGIFY(LIBAVRESAMPLE_VERSION)
+
+/**
+ * These FF_API_* defines are not part of public API.
+ * They may change, break or disappear at any time.
+ */
+
+#endif /* AVRESAMPLE_VERSION_H */

libavresample/x86/Makefile

+OBJS      += x86/audio_convert_init.o                                   \
+             x86/audio_mix_init.o
+
+YASM-OBJS += x86/audio_convert.o                                        \
+             x86/audio_mix.o

libavresample/x86/audio_convert.asm

+;******************************************************************************
+;* x86 optimized Format Conversion Utils
+;* Copyright (c) 2008 Loren Merritt
+;*
+;* 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
+;******************************************************************************
+
+%include "x86inc.asm"
+%include "x86util.asm"
+
+SECTION_TEXT
+
+;-----------------------------------------------------------------------------
+; void ff_conv_fltp_to_flt_6ch(float *dst, float *const *src, int len,
+;                              int channels);
+;-----------------------------------------------------------------------------
+
+%macro CONV_FLTP_TO_FLT_6CH 0
+cglobal conv_fltp_to_flt_6ch, 2,8,7, dst, src, src1, src2, src3, src4, src5, len
+%if ARCH_X86_64
+    mov     lend, r2d
+%else
+    %define lend dword r2m
+%endif
+    mov    src1q, [srcq+1*gprsize]
+    mov    src2q, [srcq+2*gprsize]
+    mov    src3q, [srcq+3*gprsize]
+    mov    src4q, [srcq+4*gprsize]
+    mov    src5q, [srcq+5*gprsize]
+    mov     srcq, [srcq]
+    sub    src1q, srcq
+    sub    src2q, srcq
+    sub    src3q, srcq
+    sub    src4q, srcq
+    sub    src5q, srcq
+.loop:
+    mova      m0, [srcq      ]
+    mova      m1, [srcq+src1q]
+    mova      m2, [srcq+src2q]
+    mova      m3, [srcq+src3q]
+    mova      m4, [srcq+src4q]
+    mova      m5, [srcq+src5q]
+%if cpuflag(sse)
+    SBUTTERFLYPS 0, 1, 6
+    SBUTTERFLYPS 2, 3, 6
+    SBUTTERFLYPS 4, 5, 6
+
+    movaps    m6, m4
+    shufps    m4, m0, q3210
+    movlhps   m0, m2
+    movhlps   m6, m2
+    movaps [dstq   ], m0
+    movaps [dstq+16], m4
+    movaps [dstq+32], m6
+
+    movaps    m6, m5
+    shufps    m5, m1, q3210
+    movlhps   m1, m3
+    movhlps   m6, m3
+    movaps [dstq+48], m1
+    movaps [dstq+64], m5
+    movaps [dstq+80], m6
+%else ; mmx
+    SBUTTERFLY dq, 0, 1, 6
+    SBUTTERFLY dq, 2, 3, 6
+    SBUTTERFLY dq, 4, 5, 6
+
+    movq   [dstq   ], m0
+    movq   [dstq+ 8], m2
+    movq   [dstq+16], m4
+    movq   [dstq+24], m1
+    movq   [dstq+32], m3
+    movq   [dstq+40], m5
+%endif
+    add      srcq, mmsize
+    add      dstq, mmsize*6
+    sub      lend, mmsize/4
+    jg .loop
+%if mmsize == 8
+    emms
+    RET
+%else
+    REP_RET
+%endif
+%endmacro
+
+INIT_MMX mmx
+CONV_FLTP_TO_FLT_6CH
+INIT_XMM sse
+CONV_FLTP_TO_FLT_6CH

libavresample/x86/audio_convert_init.c

+/*
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * 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
+ */
+
+#include "config.h"
+#include "libavutil/cpu.h"
+#include "libavresample/audio_convert.h"
+
+extern void ff_conv_fltp_to_flt_6ch_mmx(float *dst, float *const *src, int len);
+extern void ff_conv_fltp_to_flt_6ch_sse(float *dst, float *const *src, int len);
+
+av_cold void ff_audio_convert_init_x86(AudioConvert *ac)
+{
+#if HAVE_YASM
+    int mm_flags = av_get_cpu_flags();
+
+    if (mm_flags & AV_CPU_FLAG_MMX && HAVE_MMX) {
+        ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
+                                  6, 1, 4, "MMX", ff_conv_fltp_to_flt_6ch_mmx);
+    }
+    if (mm_flags & AV_CPU_FLAG_SSE && HAVE_SSE) {
+        ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
+                                  6, 16, 4, "SSE", ff_conv_fltp_to_flt_6ch_sse);
+    }
+#endif
+}

libavresample/x86/audio_mix.asm

+;******************************************************************************
+;* x86 optimized channel mixing
+;* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+;*
+;* 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
+;******************************************************************************
+
+%include "x86inc.asm"
+%include "x86util.asm"
+
+SECTION_TEXT
+
+;-----------------------------------------------------------------------------
+; void ff_mix_2_to_1_fltp_flt(float **src, float **matrix, int len,
+;                             int out_ch, int in_ch);
+;-----------------------------------------------------------------------------
+
+%macro MIX_2_TO_1_FLTP_FLT 0
+cglobal mix_2_to_1_fltp_flt, 3,4,6, src, matrix, len, src1
+    mov       src1q, [srcq+gprsize]
+    mov        srcq, [srcq        ]
+    sub       src1q, srcq
+    mov     matrixq, [matrixq  ]
+    VBROADCASTSS m4, [matrixq  ]
+    VBROADCASTSS m5, [matrixq+4]
+    ALIGN 16
+.loop:
+    mulps        m0, m4, [srcq             ]
+    mulps        m1, m5, [srcq+src1q       ]
+    mulps        m2, m4, [srcq+      mmsize]
+    mulps        m3, m5, [srcq+src1q+mmsize]
+    addps        m0, m0, m1
+    addps        m2, m2, m3
+    mova  [srcq       ], m0
+    mova  [srcq+mmsize], m2
+    add        srcq, mmsize*2
+    sub        lend, mmsize*2/4
+    jg .loop
+%if mmsize == 32
+    vzeroupper
+    RET
+%else
+    REP_RET
+%endif
+%endmacro
+
+INIT_XMM sse
+MIX_2_TO_1_FLTP_FLT
+INIT_YMM avx
+MIX_2_TO_1_FLTP_FLT

libavresample/x86/audio_mix_init.c

+/*
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * 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
+ */
+
+#include "config.h"
+#include "libavutil/cpu.h"
+#include "libavresample/audio_mix.h"
+
+extern void ff_mix_2_to_1_fltp_flt_sse(float **src, float **matrix, int len,
+                                       int out_ch, int in_ch);
+extern void ff_mix_2_to_1_fltp_flt_avx(float **src, float **matrix, int len,
+                                       int out_ch, int in_ch);
+
+av_cold void ff_audio_mix_init_x86(AudioMix *am)
+{
+#if HAVE_YASM
+    int mm_flags = av_get_cpu_flags();
+
+    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);
+    }
+    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);
+    }
+#endif
+}

libavutil/x86/x86util.asm

@@ -585,3 +585,12 @@
     pminsd  %1, %3
     pmaxsd  %1, %2
 %endmacro
+
+%macro VBROADCASTSS 2 ; dst xmm/ymm, src m32
+%if cpuflag(avx)
+    vbroadcastss %1, %2
+%else ; sse
+    movss        %1, %2
+    shufps       %1, %1, 0
+%endif
+%endmacro

tests/fate/aac.mak

@@ -55,8 +55,8 @@ fate-aac-ap05_48: CMD = pcm -i $(SAMPLES)/aac/ap05_48.mp4
 fate-aac-ap05_48: REF = $(SAMPLES)/aac/ap05_48.s16
 
 FATE_AAC += fate-aac-latm_stereo_to_51
-fate-aac-latm_stereo_to_51: CMD = pcm -i $(SAMPLES)/aac/latm_stereo_to_51.ts -ac 6
-fate-aac-latm_stereo_to_51: REF = $(SAMPLES)/aac/latm_stereo_to_51.s16
+fate-aac-latm_stereo_to_51: CMD = pcm -i $(SAMPLES)/aac/latm_stereo_to_51.ts -channel_layout 5.1
+fate-aac-latm_stereo_to_51: REF = $(SAMPLES)/aac/latm_stereo_to_51_ref.s16
 
 fate-aac-ct%: CMD = pcm -i $(SAMPLES)/aac/CT_DecoderCheck/$(@:fate-aac-ct-%=%)
 fate-aac-ct%: REF = $(SAMPLES)/aac/CT_DecoderCheck/aacPlusv2.wav

tests/lavf-regression.sh

@@ -118,7 +118,7 @@ fi
 if [ -n "$do_dv_fmt" ] ; then
 do_lavf_timecode_nodrop dv "-ar 48000 -r 25 -s pal -ac 2"
 do_lavf_timecode_drop   dv "-ar 48000 -pix_fmt yuv411p -s ntsc -ac 2"
-do_lavf dv "-ar 48000" "-r 25 -s pal -ac 2"
+do_lavf dv "-ar 48000 -channel_layout stereo" "-r 25 -s pal"
 fi
 
 if [ -n "$do_gxf" ] ; then

tests/ref/lavf/dv_fmt

@@ -4,6 +4,6 @@
 cc33ae4f9e6828914dea0f09d1241b7e *./tests/data/lavf/lavf.dv
 3480000 ./tests/data/lavf/lavf.dv
 ./tests/data/lavf/lavf.dv CRC=0x8d5e9e8f
-b36c83cd0ba0ebe719f09f885c4bbcd3 *./tests/data/lavf/lavf.dv
+87d3b20f656235671383a7eaa2f66330 *./tests/data/lavf/lavf.dv
 3600000 ./tests/data/lavf/lavf.dv
-./tests/data/lavf/lavf.dv CRC=0x2bc2ae3a
+./tests/data/lavf/lavf.dv CRC=0x0e868a82