Merge commit '074a00d192c0e749d677b008b337da42597e780f'

* commit '074a00d192c0e749d677b008b337da42597e780f':
  lavr: add a public function for setting a custom channel map
  lavr: typedef internal structs in internal.h
  doc: Extend commit message section

Conflicts:
	doc/APIchanges
	doc/developer.texi

Merged-by: Michael Niedermayer <michaelni@gmx.at>

15 files changed, 328 insertions, 41 deletions

diff --git a/doc/APIchanges b/doc/APIchanges
index a79dd2d..fc5c3f2 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -132,6 +132,10 @@ API changes, most recent first:
 2012-03-26 - a67d9cf - lavfi 2.66.100
   Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.
 
+2013-xx-xx - xxxxxxx - lavr 1.1.0
+  Add avresample_set_channel_mapping() for input channel reordering,
+  duplication, and silencing.
+
 2012-xx-xx - xxxxxxx - lavu 52.2.1 - avstring.h
   Add av_basename() and av_dirname().
 
diff --git a/doc/developer.texi b/doc/developer.texi
index fce842a..7e4b12c 100644
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -229,6 +229,13 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
    (#ifdef etc) by default so it does not interfere with other developers'
    work.
 @item
+   The commit message should have a short first line in the form of
+   a @samp{topic: short description} as a header, separated by a newline
+   from the body consisting of an explanation of why the change is necessary.
+   If the commit fixes a known bug on the bug tracker, the commit message
+   should include its bug ID. Referring to the issue on the bug tracker does
+   not exempt you from writing an excerpt of the bug in the commit message.
+@item
    You do not have to over-test things. If it works for you, and you think it
    should work for others, then commit. If your code has problems
    (portability, triggers compiler bugs, unusual environment etc) they will be
diff --git a/libavresample/audio_convert.c b/libavresample/audio_convert.c
index eb3bc1f..b57d2fa 100644
--- a/libavresample/audio_convert.c
+++ b/libavresample/audio_convert.c
@@ -30,7 +30,6 @@
 #include "audio_convert.h"
 #include "audio_data.h"
 #include "dither.h"
-#include "internal.h"
 
 enum ConvFuncType {
     CONV_FUNC_TYPE_FLAT,
@@ -51,6 +50,7 @@ struct AudioConvert {
     DitherContext *dc;
     enum AVSampleFormat in_fmt;
     enum AVSampleFormat out_fmt;
+    int apply_map;
     int channels;
     int planes;
     int ptr_align;
@@ -260,7 +260,8 @@ void ff_audio_convert_free(AudioConvert **ac)
 AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
                                      enum AVSampleFormat out_fmt,
                                      enum AVSampleFormat in_fmt,
-                                     int channels, int sample_rate)
+                                     int channels, int sample_rate,
+                                     int apply_map)
 {
     AudioConvert *ac;
     int in_planar, out_planar;
@@ -273,11 +274,13 @@ AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
     ac->out_fmt  = out_fmt;
     ac->in_fmt   = in_fmt;
     ac->channels = channels;
+    ac->apply_map = apply_map;
 
     if (avr->dither_method != AV_RESAMPLE_DITHER_NONE          &&
         av_get_packed_sample_fmt(out_fmt) == AV_SAMPLE_FMT_S16 &&
         av_get_bytes_per_sample(in_fmt) > 2) {
-        ac->dc = ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate);
+        ac->dc = ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate,
+                                 apply_map);
         if (!ac->dc) {
             av_free(ac);
             return NULL;
@@ -310,6 +313,7 @@ int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
 {
     int use_generic = 1;
     int len         = in->nb_samples;
+    int p;
 
     if (ac->dc) {
         /* dithered conversion */
@@ -336,9 +340,46 @@ int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
             av_get_sample_fmt_name(ac->out_fmt),
             use_generic ? ac->func_descr_generic : ac->func_descr);
 
+    if (ac->apply_map) {
+        ChannelMapInfo *map = &ac->avr->ch_map_info;
+
+        if (!av_sample_fmt_is_planar(ac->out_fmt)) {
+            av_log(ac->avr, AV_LOG_ERROR, "cannot remap packed format during conversion\n");
+            return AVERROR(EINVAL);
+        }
+
+        if (map->do_remap) {
+            if (av_sample_fmt_is_planar(ac->in_fmt)) {
+                conv_func_flat *convert = use_generic ? ac->conv_flat_generic :
+                                                        ac->conv_flat;
+
+                for (p = 0; p < ac->planes; p++)
+                    if (map->channel_map[p] >= 0)
+                        convert(out->data[p], in->data[map->channel_map[p]], len);
+            } else {
+                uint8_t *data[AVRESAMPLE_MAX_CHANNELS];
+                conv_func_deinterleave *convert = use_generic ?
+                                                  ac->conv_deinterleave_generic :
+                                                  ac->conv_deinterleave;
+
+                for (p = 0; p < ac->channels; p++)
+                    data[map->input_map[p]] = out->data[p];
+
+                convert(data, in->data[0], len, ac->channels);
+            }
+        }
+        if (map->do_copy || map->do_zero) {
+            for (p = 0; p < ac->planes; p++) {
+                if (map->channel_copy[p])
+                    memcpy(out->data[p], out->data[map->channel_copy[p]],
+                           len * out->stride);
+                else if (map->channel_zero[p])
+                    av_samples_set_silence(&out->data[p], 0, len, 1, ac->out_fmt);
+            }
+        }
+    } else {
     switch (ac->func_type) {
     case CONV_FUNC_TYPE_FLAT: {
-        int p;
         if (!in->is_planar)
             len *= in->channels;
         if (use_generic) {
@@ -363,6 +404,7 @@ int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
             ac->conv_deinterleave(out->data, in->data[0], len, ac->channels);
         break;
     }
+    }
 
     out->nb_samples = in->nb_samples;
     return 0;
diff --git a/libavresample/audio_convert.h b/libavresample/audio_convert.h
index b8808f1..6a3089d 100644
--- a/libavresample/audio_convert.h
+++ b/libavresample/audio_convert.h
@@ -23,10 +23,9 @@
 
 #include "libavutil/samplefmt.h"
 #include "avresample.h"
+#include "internal.h"
 #include "audio_data.h"
 
-typedef struct AudioConvert AudioConvert;
-
 /**
  * Set conversion function if the parameters match.
  *
@@ -59,12 +58,14 @@ void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
  * @param in_fmt      input sample format
  * @param channels    number of channels
  * @param sample_rate sample rate (used for dithering)
+ * @param apply_map   apply channel map during conversion
  * @return            newly-allocated AudioConvert context
  */
 AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
                                      enum AVSampleFormat out_fmt,
                                      enum AVSampleFormat in_fmt,
-                                     int channels, int sample_rate);
+                                     int channels, int sample_rate,
+                                     int apply_map);
 
 /**
  * Free AudioConvert.
diff --git a/libavresample/audio_data.c b/libavresample/audio_data.c
index 199a68c..c52f518 100644
--- a/libavresample/audio_data.c
+++ b/libavresample/audio_data.c
@@ -213,7 +213,7 @@ void ff_audio_data_free(AudioData **a)
     av_freep(a);
 }
 
-int ff_audio_data_copy(AudioData *dst, AudioData *src)
+int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map)
 {
     int ret, p;
 
@@ -221,6 +221,11 @@ int ff_audio_data_copy(AudioData *dst, AudioData *src)
     if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels)
         return AVERROR(EINVAL);
 
+    if (map && !src->is_planar) {
+        av_log(src, AV_LOG_ERROR, "cannot remap packed format during copy\n");
+        return AVERROR(EINVAL);
+    }
+
     /* if the input is empty, just empty the output */
     if (!src->nb_samples) {
         dst->nb_samples = 0;
@@ -233,8 +238,29 @@ int ff_audio_data_copy(AudioData *dst, AudioData *src)
         return ret;
 
     /* copy data */
-    for (p = 0; p < src->planes; p++)
-        memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride);
+    if (map) {
+        if (map->do_remap) {
+            for (p = 0; p < src->planes; p++) {
+                if (map->channel_map[p] >= 0)
+                    memcpy(dst->data[p], src->data[map->channel_map[p]],
+                           src->nb_samples * src->stride);
+            }
+        }
+        if (map->do_copy || map->do_zero) {
+            for (p = 0; p < src->planes; p++) {
+                if (map->channel_copy[p])
+                    memcpy(dst->data[p], dst->data[map->channel_copy[p]],
+                           src->nb_samples * src->stride);
+                else if (map->channel_zero[p])
+                    av_samples_set_silence(&dst->data[p], 0, src->nb_samples,
+                                           1, dst->sample_fmt);
+            }
+        }
+    } else {
+        for (p = 0; p < src->planes; p++)
+            memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride);
+    }
+
     dst->nb_samples = src->nb_samples;
 
     return 0;
diff --git a/libavresample/audio_data.h b/libavresample/audio_data.h
index 558e7e6..97236bb 100644
--- a/libavresample/audio_data.h
+++ b/libavresample/audio_data.h
@@ -27,11 +27,12 @@
 #include "libavutil/log.h"
 #include "libavutil/samplefmt.h"
 #include "avresample.h"
+#include "internal.h"
 
 /**
  * Audio buffer used for intermediate storage between conversion phases.
  */
-typedef struct AudioData {
+struct AudioData {
     const AVClass *class;               /**< AVClass for logging            */
     uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers        */
     uint8_t *buffer;                    /**< data buffer                    */
@@ -50,7 +51,7 @@ typedef struct AudioData {
     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);
 
@@ -117,9 +118,10 @@ void ff_audio_data_free(AudioData **a);
  *
  * @param out  output AudioData
  * @param in   input AudioData
+ * @param map  channel map, NULL if not remapping
  * @return     0 on success, negative AVERROR value on error
  */
-int ff_audio_data_copy(AudioData *out, AudioData *in);
+int ff_audio_data_copy(AudioData *out, AudioData *in, ChannelMapInfo *map);
 
 /**
  * Append data from one AudioData to the end of another.
diff --git a/libavresample/audio_mix.h b/libavresample/audio_mix.h
index 4ccfc1f..5bae5ab 100644
--- a/libavresample/audio_mix.h
+++ b/libavresample/audio_mix.h
@@ -25,13 +25,12 @@
 
 #include "libavutil/samplefmt.h"
 #include "avresample.h"
+#include "internal.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 AudioMix;
-
 /**
  * Set mixing function if the parameters match.
  *
diff --git a/libavresample/avresample.h b/libavresample/avresample.h
index 0012787..d26f2ca 100644
--- a/libavresample/avresample.h
+++ b/libavresample/avresample.h
@@ -259,6 +259,36 @@ int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
                           int stride);
 
 /**
+ * Set a customized input channel mapping.
+ *
+ * This function can only be called when the allocated context is not open.
+ * Also, the input channel layout must have already been set.
+ *
+ * Calling avresample_close() on the context will clear the channel mapping.
+ *
+ * The map for each input channel specifies the channel index in the source to
+ * use for that particular channel, or -1 to mute the channel. Source channels
+ * can be duplicated by using the same index for multiple input channels.
+ *
+ * Examples:
+ *
+ * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to Libav order (L,R,C,LFE,Ls,Rs):
+ * { 1, 2, 0, 5, 3, 4 }
+ *
+ * Muting the 3rd channel in 4-channel input:
+ * { 0, 1, -1, 3 }
+ *
+ * Duplicating the left channel of stereo input:
+ * { 0, 0 }
+ *
+ * @param avr         audio resample context
+ * @param channel_map customized input channel mapping
+ * @return            0 on success, negative AVERROR code on failure
+ */
+int avresample_set_channel_mapping(AVAudioResampleContext *avr,
+                                   const int *channel_map);
+
+/**
  * Set compensation for resampling.
  *
  * This can be called anytime after avresample_open(). If resampling is not
diff --git a/libavresample/dither.c b/libavresample/dither.c
index 9c1e1c1..dfff03e 100644
--- a/libavresample/dither.c
+++ b/libavresample/dither.c
@@ -53,6 +53,8 @@ typedef struct DitherState {
 struct DitherContext {
     DitherDSPContext  ddsp;
     enum AVResampleDitherMethod method;
+    int apply_map;
+    ChannelMapInfo *ch_map_info;
 
     int mute_dither_threshold;  // threshold for disabling dither
     int mute_reset_threshold;   // threshold for resetting noise shaping
@@ -251,17 +253,23 @@ int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src)
             return ret;
     }
 
-    if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) {
+    if (src->sample_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {
         /* make sure flt_data is large enough for the input */
         ret = ff_audio_data_realloc(c->flt_data, src->nb_samples);
         if (ret < 0)
             return ret;
         flt_data = c->flt_data;
+    }
 
+    if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) {
         /* convert input samples to fltp and scale to s16 range */
         ret = ff_audio_convert(c->ac_in, flt_data, src);
         if (ret < 0)
             return ret;
+    } else if (c->apply_map) {
+        ret = ff_audio_data_copy(flt_data, src, c->ch_map_info);
+        if (ret < 0)
+            return ret;
     } else {
         flt_data = src;
     }
@@ -333,7 +341,7 @@ static void dither_init(DitherDSPContext *ddsp,
 DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
                                enum AVSampleFormat out_fmt,
                                enum AVSampleFormat in_fmt,
-                               int channels, int sample_rate)
+                               int channels, int sample_rate, int apply_map)
 {
     AVLFG seed_gen;
     DitherContext *c;
@@ -350,6 +358,10 @@ DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
     if (!c)
         return NULL;
 
+    c->apply_map = apply_map;
+    if (apply_map)
+        c->ch_map_info = &avr->ch_map_info;
+
     if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS &&
         sample_rate != 48000 && sample_rate != 44100) {
         av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz "
@@ -379,19 +391,20 @@ DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
             goto fail;
 
         c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P,
-                                           channels, sample_rate);
+                                           channels, sample_rate, 0);
         if (!c->ac_out)
             goto fail;
     }
 
-    if (in_fmt != AV_SAMPLE_FMT_FLTP) {
+    if (in_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {
         c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP,
                                           "dither flt buffer");
         if (!c->flt_data)
             goto fail;
-
+    }
+    if (in_fmt != AV_SAMPLE_FMT_FLTP) {
         c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt,
-                                          channels, sample_rate);
+                                          channels, sample_rate, c->apply_map);
         if (!c->ac_in)
             goto fail;
     }
diff --git a/libavresample/dither.h b/libavresample/dither.h
index 8b30dd2..d6a7d3e 100644
--- a/libavresample/dither.h
+++ b/libavresample/dither.h
@@ -66,7 +66,7 @@ typedef struct DitherDSPContext {
 DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
                                enum AVSampleFormat out_fmt,
                                enum AVSampleFormat in_fmt,
-                               int channels, int sample_rate);
+                               int channels, int sample_rate, int apply_map);
 
 /**
  * Free a DitherContext.
diff --git a/libavresample/internal.h b/libavresample/internal.h
index 2e139ab..057f89a 100644
--- a/libavresample/internal.h
+++ b/libavresample/internal.h
@@ -26,10 +26,29 @@
 #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"
+
+typedef struct AudioData AudioData;
+typedef struct AudioConvert AudioConvert;
+typedef struct AudioMix AudioMix;
+typedef struct ResampleContext ResampleContext;
+
+enum RemapPoint {
+    REMAP_NONE,
+    REMAP_IN_COPY,
+    REMAP_IN_CONVERT,
+    REMAP_OUT_COPY,
+    REMAP_OUT_CONVERT,
+};
+
+typedef struct ChannelMapInfo {
+    int channel_map[AVRESAMPLE_MAX_CHANNELS];   /**< source index of each output channel, -1 if not remapped */
+    int do_remap;                               /**< remap needed */
+    int channel_copy[AVRESAMPLE_MAX_CHANNELS];  /**< dest index to copy from */
+    int do_copy;                                /**< copy needed */
+    int channel_zero[AVRESAMPLE_MAX_CHANNELS];  /**< dest index to zero */
+    int do_zero;                                /**< zeroing needed */
+    int input_map[AVRESAMPLE_MAX_CHANNELS];     /**< dest index of each input channel */
+} ChannelMapInfo;
 
 struct AVAudioResampleContext {
     const AVClass *av_class;        /**< AVClass for logging and AVOptions  */
@@ -64,6 +83,7 @@ struct AVAudioResampleContext {
     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  */
+    int in_copy_needed;     /**< input data copy is needed                  */
 
     AudioData *in_buffer;           /**< buffer for converted input         */
     AudioData *resample_out_buffer; /**< buffer for output from resampler   */
@@ -81,6 +101,10 @@ struct AVAudioResampleContext {
      * only used if avresample_set_matrix() is called before avresample_open()
      */
     double *mix_matrix;
+
+    int use_channel_map;
+    enum RemapPoint remap_point;
+    ChannelMapInfo ch_map_info;
 };
 
 #endif /* AVRESAMPLE_INTERNAL_H */
diff --git a/libavresample/resample.c b/libavresample/resample.c
index dc121fe..69c9bab 100644
--- a/libavresample/resample.c
+++ b/libavresample/resample.c
@@ -23,6 +23,7 @@
 #include "libavutil/libm.h"
 #include "libavutil/log.h"
 #include "internal.h"
+#include "resample.h"
 #include "audio_data.h"
 
 struct ResampleContext {
diff --git a/libavresample/resample.h b/libavresample/resample.h
index 7534e26..4544dab 100644
--- a/libavresample/resample.h
+++ b/libavresample/resample.h
@@ -22,10 +22,9 @@
 #define AVRESAMPLE_RESAMPLE_H
 
 #include "avresample.h"
+#include "internal.h"
 #include "audio_data.h"
 
-typedef struct ResampleContext ResampleContext;
-
 /**
  * Allocate and initialize a ResampleContext.
  *
diff --git a/libavresample/utils.c b/libavresample/utils.c
index 8f6583d..c159e33 100644
--- a/libavresample/utils.c
+++ b/libavresample/utils.c
@@ -26,8 +26,11 @@
 #include "libavutil/opt.h"
 
 #include "avresample.h"
-#include "audio_data.h"
 #include "internal.h"
+#include "audio_data.h"
+#include "audio_convert.h"
+#include "audio_mix.h"
+#include "resample.h"
 
 int avresample_open(AVAudioResampleContext *avr)
 {
@@ -93,20 +96,84 @@ int avresample_open(AVAudioResampleContext *avr)
                av_get_sample_fmt_name(avr->internal_sample_fmt));
     }
 
-    /* set sample format conversion parameters */
+    /* treat all mono as planar for easier comparison */
     if (avr->in_channels == 1)
         avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
     if (avr->out_channels == 1)
         avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
-    avr->in_convert_needed = (avr->resample_needed || avr->mixing_needed) &&
-                              avr->in_sample_fmt != avr->internal_sample_fmt;
+
+    /* we may need to add an extra conversion in order to remap channels if
+       the output format is not planar */
+    if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed &&
+        !av_sample_fmt_is_planar(avr->out_sample_fmt)) {
+        avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
+    }
+
+    /* set sample format conversion parameters */
     if (avr->resample_needed || avr->mixing_needed)
+        avr->in_convert_needed = avr->in_sample_fmt != avr->internal_sample_fmt;
+    else
+        avr->in_convert_needed = avr->use_channel_map &&
+                                 !av_sample_fmt_is_planar(avr->out_sample_fmt);
+
+    if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed)
         avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
     else
         avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
 
+    avr->in_copy_needed = !avr->in_convert_needed && (avr->mixing_needed ||
+                          (avr->use_channel_map && avr->resample_needed));
+
+    if (avr->use_channel_map) {
+        if (avr->in_copy_needed) {
+            avr->remap_point = REMAP_IN_COPY;
+            av_dlog(avr, "remap channels during in_copy\n");
+        } else if (avr->in_convert_needed) {
+            avr->remap_point = REMAP_IN_CONVERT;
+            av_dlog(avr, "remap channels during in_convert\n");
+        } else if (avr->out_convert_needed) {
+            avr->remap_point = REMAP_OUT_CONVERT;
+            av_dlog(avr, "remap channels during out_convert\n");
+        } else {
+            avr->remap_point = REMAP_OUT_COPY;
+            av_dlog(avr, "remap channels during out_copy\n");
+        }
+
+#ifdef DEBUG
+        {
+            int ch;
+            av_dlog(avr, "output map: ");
+            if (avr->ch_map_info.do_remap)
+                for (ch = 0; ch < avr->in_channels; ch++)
+                    av_dlog(avr, " % 2d", avr->ch_map_info.channel_map[ch]);
+            else
+                av_dlog(avr, "n/a");
+            av_dlog(avr, "\n");
+            av_dlog(avr, "copy map:   ");
+            if (avr->ch_map_info.do_copy)
+                for (ch = 0; ch < avr->in_channels; ch++)
+                    av_dlog(avr, " % 2d", avr->ch_map_info.channel_copy[ch]);
+            else
+                av_dlog(avr, "n/a");
+            av_dlog(avr, "\n");
+            av_dlog(avr, "zero map:   ");
+            if (avr->ch_map_info.do_zero)
+                for (ch = 0; ch < avr->in_channels; ch++)
+                    av_dlog(avr, " % 2d", avr->ch_map_info.channel_zero[ch]);
+            else
+                av_dlog(avr, "n/a");
+            av_dlog(avr, "\n");
+            av_dlog(avr, "input map:  ");
+            for (ch = 0; ch < avr->in_channels; ch++)
+                av_dlog(avr, " % 2d", avr->ch_map_info.input_map[ch]);
+            av_dlog(avr, "\n");
+        }
+#endif
+    } else
+        avr->remap_point = REMAP_NONE;
+
     /* allocate buffers */
-    if (avr->mixing_needed || avr->in_convert_needed) {
+    if (avr->in_copy_needed || avr->in_convert_needed) {
         avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
                                              0, avr->internal_sample_fmt,
                                              "in_buffer");
@@ -143,7 +210,8 @@ int avresample_open(AVAudioResampleContext *avr)
     if (avr->in_convert_needed) {
         avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
                                             avr->in_sample_fmt, avr->in_channels,
-                                            avr->in_sample_rate);
+                                            avr->in_sample_rate,
+                                            avr->remap_point == REMAP_IN_CONVERT);
         if (!avr->ac_in) {
             ret = AVERROR(ENOMEM);
             goto error;
@@ -157,7 +225,8 @@ int avresample_open(AVAudioResampleContext *avr)
             src_fmt = avr->in_sample_fmt;
         avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
                                              avr->out_channels,
-                                             avr->out_sample_rate);
+                                             avr->out_sample_rate,
+                                             avr->remap_point == REMAP_OUT_CONVERT);
         if (!avr->ac_out) {
             ret = AVERROR(ENOMEM);
             goto error;
@@ -197,6 +266,8 @@ void avresample_close(AVAudioResampleContext *avr)
     ff_audio_resample_free(&avr->resample);
     ff_audio_mix_free(&avr->am);
     av_freep(&avr->mix_matrix);
+
+    avr->use_channel_map = 0;
 }
 
 void avresample_free(AVAudioResampleContext **avr)
@@ -239,7 +310,9 @@ static int handle_buffered_output(AVAudioResampleContext *avr,
            data in the output FIFO */
         av_dlog(avr, "[copy] %s to output\n", converted->name);
         output->nb_samples = 0;
-        ret = ff_audio_data_copy(output, converted);
+        ret = ff_audio_data_copy(output, converted,
+                                 avr->remap_point == REMAP_OUT_COPY ?
+                                 &avr->ch_map_info : NULL);
         if (ret < 0)
             return ret;
         av_dlog(avr, "[end conversion]\n");
@@ -303,11 +376,24 @@ int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
             /* in some rare cases we can copy input to output and upmix
                directly in the output buffer */
             av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
-            ret = ff_audio_data_copy(&output_buffer, current_buffer);
+            ret = ff_audio_data_copy(&output_buffer, current_buffer,
+                                     avr->remap_point == REMAP_OUT_COPY ?
+                                     &avr->ch_map_info : NULL);
             if (ret < 0)
                 return ret;
             current_buffer = &output_buffer;
-        } else if (avr->mixing_needed || avr->in_convert_needed) {
+        } else if (avr->remap_point == REMAP_OUT_COPY &&
+                   (!direct_output || out_samples < in_samples)) {
+            /* if remapping channels during output copy, we may need to
+             * use an intermediate buffer in order to remap before adding
+             * samples to the output fifo */
+            av_dlog(avr, "[copy] %s to out_buffer\n", current_buffer->name);
+            ret = ff_audio_data_copy(avr->out_buffer, current_buffer,
+                                     &avr->ch_map_info);
+            if (ret < 0)
+                return ret;
+            current_buffer = avr->out_buffer;
+        } else if (avr->in_copy_needed || avr->in_convert_needed) {
             /* if needed, copy or convert input to in_buffer, and downmix if
                applicable */
             if (avr->in_convert_needed) {
@@ -322,7 +408,9 @@ int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
                     return ret;
             } else {
                 av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
-                ret = ff_audio_data_copy(avr->in_buffer, current_buffer);
+                ret = ff_audio_data_copy(avr->in_buffer, current_buffer,
+                                         avr->remap_point == REMAP_IN_COPY ?
+                                         &avr->ch_map_info : NULL);
                 if (ret < 0)
                     return ret;
             }
@@ -467,6 +555,57 @@ int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
     return 0;
 }
 
+int avresample_set_channel_mapping(AVAudioResampleContext *avr,
+                                   const int *channel_map)
+{
+    ChannelMapInfo *info = &avr->ch_map_info;
+    int in_channels, ch, i;
+
+    in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
+    if (in_channels <= 0 ||  in_channels > AVRESAMPLE_MAX_CHANNELS) {
+        av_log(avr, AV_LOG_ERROR, "Invalid input channel layout\n");
+        return AVERROR(EINVAL);
+    }
+
+    memset(info, 0, sizeof(*info));
+    memset(info->input_map, -1, sizeof(info->input_map));
+
+    for (ch = 0; ch < in_channels; ch++) {
+        if (channel_map[ch] >= in_channels) {
+            av_log(avr, AV_LOG_ERROR, "Invalid channel map\n");
+            return AVERROR(EINVAL);
+        }
+        if (channel_map[ch] < 0) {
+            info->channel_zero[ch] =  1;
+            info->channel_map[ch]  = -1;
+            info->do_zero          =  1;
+        } else if (info->input_map[channel_map[ch]] >= 0) {
+            info->channel_copy[ch] = info->input_map[channel_map[ch]];
+            info->channel_map[ch]  = -1;
+            info->do_copy          =  1;
+        } else {
+            info->channel_map[ch]            = channel_map[ch];
+            info->input_map[channel_map[ch]] = ch;
+            info->do_remap                   =  1;
+        }
+    }
+    /* Fill-in unmapped input channels with unmapped output channels.
+       This is used when remapping during conversion from interleaved to
+       planar format. */
+    for (ch = 0, i = 0; ch < in_channels && i < in_channels; ch++, i++) {
+        while (ch < in_channels && info->input_map[ch] >= 0)
+            ch++;
+        while (i < in_channels && info->channel_map[i] >= 0)
+            i++;
+        if (ch >= in_channels || i >= in_channels)
+            break;
+        info->input_map[ch] = i;
+    }
+
+    avr->use_channel_map = 1;
+    return 0;
+}
+
 int avresample_available(AVAudioResampleContext *avr)
 {
     return av_audio_fifo_size(avr->out_fifo);
diff --git a/libavresample/version.h b/libavresample/version.h
index ebcd07f..387d097 100644
--- a/libavresample/version.h
+++ b/libavresample/version.h
@@ -20,8 +20,8 @@
 #define AVRESAMPLE_VERSION_H
 
 #define LIBAVRESAMPLE_VERSION_MAJOR  1
-#define LIBAVRESAMPLE_VERSION_MINOR  0
-#define LIBAVRESAMPLE_VERSION_MICRO  1
+#define LIBAVRESAMPLE_VERSION_MINOR  1
+#define LIBAVRESAMPLE_VERSION_MICRO  0
 
 #define LIBAVRESAMPLE_VERSION_INT  AV_VERSION_INT(LIBAVRESAMPLE_VERSION_MAJOR, \
                                                   LIBAVRESAMPLE_VERSION_MINOR, \