Merge remote-tracking branch 'qatar/master'

* qatar/master:
  avconv: use default alignment for audio buffer
  avcodec: use align == 0 for default alignment in avcodec_fill_audio_frame()
  avutil: use align == 0 for default alignment in audio sample buffer functions
  avutil: allow NULL linesize in av_samples_fill_arrays() and av_samples_alloc()
  avconv: remove OutputStream.picref.
  avconv: only set SAR once on the decoded frame.
  avcodec: validate the channel layout vs. channel count for decoders
  audioconvert: make av_get_channel_layout accept composite names.
  avutil: add av_get_packed_sample_fmt() and av_get_planar_sample_fmt()

Conflicts:
	doc/APIchanges
	ffmpeg.c
	libavcodec/utils.c
	libavcodec/version.h
	libavutil/audioconvert.c
	libavutil/audioconvert.h
	libavutil/avutil.h
	libavutil/samplefmt.c
	libavutil/samplefmt.h

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

5 files changed, 61 insertions, 14 deletions

diff --git a/libavutil/audioconvert.c b/libavutil/audioconvert.c
index aa652a66b1..877027f1f7 100644
--- a/libavutil/audioconvert.c
+++ b/libavutil/audioconvert.c
@@ -80,19 +80,16 @@ static const struct {
     { "4.1",         5,  AV_CH_LAYOUT_4POINT1 },
     { "5.1",         6,  AV_CH_LAYOUT_5POINT1_BACK },
     { "5.1(side)",   6,  AV_CH_LAYOUT_5POINT1 },
-//     { "5.1+downmix", 8,  AV_CH_LAYOUT_5POINT1|AV_CH_LAYOUT_STEREO_DOWNMIX, },
     { "6.0",         6,  AV_CH_LAYOUT_6POINT0 },
     { "6.0(front)",  6,  AV_CH_LAYOUT_6POINT0_FRONT },
     { "hexagonal",   6,  AV_CH_LAYOUT_HEXAGONAL },
     { "6.1",         7,  AV_CH_LAYOUT_6POINT1 },
     { "6.1",         7,  AV_CH_LAYOUT_6POINT1_BACK },
     { "6.1(front)",  7,  AV_CH_LAYOUT_6POINT1_FRONT },
-//     { "6.1+downmix", 9,  AV_CH_LAYOUT_6POINT1|AV_CH_LAYOUT_STEREO_DOWNMIX, },
     { "7.0",         7,  AV_CH_LAYOUT_7POINT0 },
     { "7.0(front)",  7,  AV_CH_LAYOUT_7POINT0_FRONT },
     { "7.1",         8,  AV_CH_LAYOUT_7POINT1 },
     { "7.1(wide)",   8,  AV_CH_LAYOUT_7POINT1_WIDE },
-//     { "7.1+downmix", 10, AV_CH_LAYOUT_7POINT1|AV_CH_LAYOUT_STEREO_DOWNMIX, },
     { "octagonal",   8,  AV_CH_LAYOUT_OCTAGONAL },
     { "downmix",     2,  AV_CH_LAYOUT_STEREO_DOWNMIX, },
 };
diff --git a/libavutil/audioconvert.h b/libavutil/audioconvert.h
index 6414d3c3a0..f402b62f63 100644
--- a/libavutil/audioconvert.h
+++ b/libavutil/audioconvert.h
@@ -106,19 +106,21 @@
  */
 
 /**
- * Return a channel layout id that matches name, 0 if no match.
+ * Return a channel layout id that matches name, or 0 if no match is found.
+ *
  * name can be one or several of the following notations,
  * separated by '+' or '|':
  * - the name of an usual channel layout (mono, stereo, 4.0, quad, 5.0,
  *   5.0(side), 5.1, 5.1(side), 7.1, 7.1(wide), downmix);
  * - the name of a single channel (FL, FR, FC, LFE, BL, BR, FLC, FRC, BC,
  *   SL, SR, TC, TFL, TFC, TFR, TBL, TBC, TBR, DL, DR);
- * - a number of channels, in decimal, optionnally followed by 'c', yielding
+ * - a number of channels, in decimal, optionally followed by 'c', yielding
  *   the default channel layout for that number of channels (@see
  *   av_get_default_channel_layout);
  * - a channel layout mask, in hexadecimal starting with "0x" (see the
  *   AV_CH_* macros).
- + Example: "stereo+FC" = "2+FC" = "2c+1c" = "0x7"
+ *
+ * Example: "stereo+FC" = "2+FC" = "2c+1c" = "0x7"
  */
 uint64_t av_get_channel_layout(const char *name);
 
diff --git a/libavutil/avutil.h b/libavutil/avutil.h
index 150b788bfb..6464167473 100644
--- a/libavutil/avutil.h
+++ b/libavutil/avutil.h
@@ -153,7 +153,7 @@
  */
 
 #define LIBAVUTIL_VERSION_MAJOR 51
-#define LIBAVUTIL_VERSION_MINOR 45
+#define LIBAVUTIL_VERSION_MINOR 46
 #define LIBAVUTIL_VERSION_MICRO 100
 
 #define LIBAVUTIL_VERSION_INT   AV_VERSION_INT(LIBAVUTIL_VERSION_MAJOR, \
diff --git a/libavutil/samplefmt.c b/libavutil/samplefmt.c
index f4300036c2..98dacb1f66 100644
--- a/libavutil/samplefmt.c
+++ b/libavutil/samplefmt.c
@@ -69,6 +69,24 @@ enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int pl
     return sample_fmt_info[sample_fmt].altform;
 }
 
+enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
+{
+    if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
+        return AV_SAMPLE_FMT_NONE;
+    if (sample_fmt_info[sample_fmt].planar)
+        return sample_fmt_info[sample_fmt].altform;
+    return sample_fmt;
+}
+
+enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
+{
+    if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
+        return AV_SAMPLE_FMT_NONE;
+    if (sample_fmt_info[sample_fmt].planar)
+        return sample_fmt;
+    return sample_fmt_info[sample_fmt].altform;
+}
+
 char *av_get_sample_fmt_string (char *buf, int buf_size, enum AVSampleFormat sample_fmt)
 {
     /* print header */
@@ -114,6 +132,10 @@ int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
     if (!sample_size || nb_samples <= 0 || nb_channels <= 0)
         return AVERROR(EINVAL);
 
+    /* auto-select alignment if not specified */
+    if (!align)
+        align = 32;
+
     /* check for integer overflow */
     if (nb_channels > INT_MAX / align ||
         (int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size)
@@ -131,17 +153,20 @@ int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
                            uint8_t *buf, int nb_channels, int nb_samples,
                            enum AVSampleFormat sample_fmt, int align)
 {
-    int ch, planar, buf_size;
+    int ch, planar, buf_size, line_size;
 
     planar   = av_sample_fmt_is_planar(sample_fmt);
-    buf_size = av_samples_get_buffer_size(linesize, nb_channels, nb_samples,
+    buf_size = av_samples_get_buffer_size(&line_size, nb_channels, nb_samples,
                                           sample_fmt, align);
     if (buf_size < 0)
         return buf_size;
 
     audio_data[0] = buf;
     for (ch = 1; planar && ch < nb_channels; ch++)
-        audio_data[ch] = audio_data[ch-1] + *linesize;
+        audio_data[ch] = audio_data[ch-1] + line_size;
+
+    if (linesize)
+        *linesize = line_size;
 
     return 0;
 }
diff --git a/libavutil/samplefmt.h b/libavutil/samplefmt.h
index 855cffd838..af015d0cc3 100644
--- a/libavutil/samplefmt.h
+++ b/libavutil/samplefmt.h
@@ -62,6 +62,28 @@ enum AVSampleFormat av_get_sample_fmt(const char *name);
 enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar);
 
 /**
+ * Get the packed alternative form of the given sample format.
+ *
+ * If the passed sample_fmt is already in packed format, the format returned is
+ * the same as the input.
+ *
+ * @return  the packed alternative form of the given sample format or
+            AV_SAMPLE_FMT_NONE on error.
+ */
+enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt);
+
+/**
+ * Get the planar alternative form of the given sample format.
+ *
+ * If the passed sample_fmt is already in planar format, the format returned is
+ * the same as the input.
+ *
+ * @return  the planar alternative form of the given sample format or
+            AV_SAMPLE_FMT_NONE on error.
+ */
+enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt);
+
+/**
  * Generate a string corresponding to the sample format with
  * sample_fmt, or a header if sample_fmt is negative.
  *
@@ -107,6 +129,7 @@ int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt);
  * @param nb_channels   the number of channels
  * @param nb_samples    the number of samples in a single channel
  * @param sample_fmt    the sample format
+ * @param align         buffer size alignment (0 = default, 1 = no alignment)
  * @return              required buffer size, or negative error code on failure
  */
 int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
@@ -125,12 +148,12 @@ int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
  * for packed layout.
  *
  * @param[out] audio_data  array to be filled with the pointer for each channel
- * @param[out] linesize    calculated linesize
+ * @param[out] linesize    calculated linesize, may be NULL
  * @param buf              the pointer to a buffer containing the samples
  * @param nb_channels      the number of channels
  * @param nb_samples       the number of samples in a single channel
  * @param sample_fmt       the sample format
- * @param align            buffer size alignment (1 = no alignment required)
+ * @param align            buffer size alignment (0 = default, 1 = no alignment)
  * @return                 0 on success or a negative error code on failure
  */
 int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, uint8_t *buf,
@@ -143,10 +166,10 @@ int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, uint8_t *buf,
  * The allocated samples buffer can be freed by using av_freep(&audio_data[0])
  *
  * @param[out] audio_data  array to be filled with the pointer for each channel
- * @param[out] linesize    aligned size for audio buffer(s)
+ * @param[out] linesize    aligned size for audio buffer(s), may be NULL
  * @param nb_channels      number of audio channels
  * @param nb_samples       number of samples per channel
- * @param align            buffer size alignment (1 = no alignment required)
+ * @param align            buffer size alignment (0 = default, 1 = no alignment)
  * @return                 0 on success or a negative error code on failure
  * @see av_samples_fill_arrays()
  */