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>

10 files changed, 89 insertions, 28 deletions

diff --git a/doc/APIchanges b/doc/APIchanges
index d0357516ca..991630969e 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -16,6 +16,9 @@ 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 - lavu 51.27.0 - samplefmt.h
+  Add av_get_packed_sample_fmt() and av_get_planar_sample_fmt()
+
 2012-03-21 - b75c67d - lavu 51.43.100
   Add bprint.h for bprint API.
 
diff --git a/ffmpeg.c b/ffmpeg.c
index 0f70118982..0a2375d38a 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -272,7 +272,6 @@ typedef struct OutputStream {
 
     AVFilterContext *output_video_filter;
     AVFilterContext *input_video_filter;
-    AVFilterBufferRef *picref;
     char *avfilter;
     AVFilterGraph *graph;
 
@@ -1130,7 +1129,7 @@ static int alloc_audio_output_buf(AVCodecContext *dec, AVCodecContext *enc,
 
     audio_buf_size = av_samples_get_buffer_size(NULL, enc->channels,
                                                 audio_buf_samples,
-                                                enc->sample_fmt, 32);
+                                                enc->sample_fmt, 0);
     if (audio_buf_size < 0)
         return audio_buf_size;
 
@@ -2116,6 +2115,9 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
 
     rate_emu_sleep(ist);
 
+    if (ist->st->sample_aspect_ratio.num)
+        decoded_frame->sample_aspect_ratio = ist->st->sample_aspect_ratio;
+
     for (i = 0; i < nb_output_streams; i++) {
         OutputStream *ost = &output_streams[i];
 
@@ -2125,8 +2127,9 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
         while (av_buffersink_poll_frame(ost->output_video_filter)) {
             AVRational ist_pts_tb = ost->output_video_filter->inputs[0]->time_base;
             AVFrame *filtered_frame;
+            AVFilterBufferRef *picref;
 
-            if (av_buffersink_get_buffer_ref(ost->output_video_filter, &ost->picref, 0) < 0){
+            if (av_buffersink_get_buffer_ref(ost->output_video_filter, &picref, 0) < 0){
                 av_log(NULL, AV_LOG_WARNING, "AV Filter told us it has a frame available but failed to output one\n");
                 goto cont;
             }
@@ -2136,13 +2139,13 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
             }
             filtered_frame = ist->filtered_frame;
             *filtered_frame= *decoded_frame; //for me_threshold
-            avfilter_fill_frame_from_video_buffer_ref(filtered_frame, ost->picref);
-            filtered_frame->pts = av_rescale_q(ost->picref->pts, ist_pts_tb, AV_TIME_BASE_Q);
+            avfilter_fill_frame_from_video_buffer_ref(filtered_frame, picref);
+            filtered_frame->pts = av_rescale_q(picref->pts, ist_pts_tb, AV_TIME_BASE_Q);
             if (!ost->frame_aspect_ratio)
-                ost->st->codec->sample_aspect_ratio = ost->picref->video->sample_aspect_ratio;
+                ost->st->codec->sample_aspect_ratio = picref->video->sample_aspect_ratio;
             do_video_out(output_files[ost->file_index].ctx, ost, ist, filtered_frame);
             cont:
-            avfilter_unref_buffer(ost->picref);
+            avfilter_unref_buffer(picref);
         }
     }
 
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index de9b245879..56fc15cad1 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -4018,7 +4018,7 @@ int avcodec_encode_audio2(AVCodecContext *avctx, AVPacket *avpkt,
  * @param sample_fmt  sample format
  * @param buf         buffer to use for frame data
  * @param buf_size    size of buffer
- * @param align       plane size sample alignment
+ * @param align       plane size sample alignment (0 = default)
  * @return            0 on success, negative error code on failure
  */
 int avcodec_fill_audio_frame(AVFrame *frame, int nb_channels,
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index ab34b40c62..f6c9466fbc 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -319,7 +319,7 @@ static int audio_get_buffer(AVCodecContext *avctx, AVFrame *frame)
 
     buf_size = av_samples_get_buffer_size(NULL, avctx->channels,
                                           frame->nb_samples, avctx->sample_fmt,
-                                          32);
+                                          0);
     if (buf_size < 0)
         return AVERROR(EINVAL);
 
@@ -361,7 +361,7 @@ static int audio_get_buffer(AVCodecContext *avctx, AVFrame *frame)
         }
         if ((ret = avcodec_fill_audio_frame(frame, avctx->channels,
                                             avctx->sample_fmt, buf->data[0],
-                                            buf->audio_data_size, 32)))
+                                            buf->audio_data_size, 0)))
             return ret;
 
         if (frame->extended_data == frame->data)
@@ -913,10 +913,18 @@ int attribute_align_arg avcodec_open2(AVCodecContext *avctx, AVCodec *codec, AVD
         }
     }
 
-    if (av_codec_is_decoder(avctx->codec) && !avctx->bit_rate)
-        avctx->bit_rate = get_bit_rate(avctx);
-
     ret=0;
+
+    if (av_codec_is_decoder(avctx->codec)) {
+        if (!avctx->bit_rate)
+            avctx->bit_rate = get_bit_rate(avctx);
+        /* validate channel layout from the decoder */
+        if (avctx->channel_layout &&
+            av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels) {
+            av_log(avctx, AV_LOG_WARNING, "channel layout does not match number of channels\n");
+            avctx->channel_layout = 0;
+        }
+    }
 end:
     entangled_thread_counter--;
 
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 47591fde76..6c631d8045 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -22,7 +22,7 @@
 
 #define LIBAVCODEC_VERSION_MAJOR 54
 #define LIBAVCODEC_VERSION_MINOR  14
-#define LIBAVCODEC_VERSION_MICRO 100
+#define LIBAVCODEC_VERSION_MICRO 101
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
                                                LIBAVCODEC_VERSION_MINOR, \
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()
  */