Merge commit 'ef1b23ad21e3f12fc4ff2a73a6d4d4cd9d630c4b'

* commit 'ef1b23ad21e3f12fc4ff2a73a6d4d4cd9d630c4b': (21 commits) jvdec: set channel layout iss: set channel layout ipmovie: set channel layout iff: set channel layout idroqdec: set channel layout gxfdec: set channel layout when applicable gsmdec: set channel layout flvdec: set channel layout dv: set channel layout dsicin: set channel layout daud: set channel layout cdxl: set channel layout bmv: set channel layout bink: set channel layout bfi: set channel layout bethsoftvid: set channel layout apc: set channel layout amr: set channel_layout ppc: replace pointer casting with AV_COPY32 ppc: fix some unused variable warnings ... Conflicts: libavformat/amr.c libavformat/iff.c Merged-by: Michael Niedermayer <[email protected]>
SaschaVasarrhelyi · Nov 13, 2012 · 7eb40d8 · 7eb40d8
2 parents a8f2420 + ef1b23a
commit 7eb40d8
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Showing 21 changed files with 92 additions and 21 deletions.
diff --git a/configure b/configure
@@ -638,11 +638,10 @@ print_config(){
 }
 
 print_enabled(){
-    test x"$1" = x-n && end=" " && shift || end="\n"
     suf=$1
     shift
     for v; do
-        enabled $v && printf "%s$end" ${v%$suf};
+        enabled $v && printf "%s\n" ${v%$suf};
     done
 }
 

diff --git a/libavcodec/ppc/h264_altivec.c b/libavcodec/ppc/h264_altivec.c
@@ -19,6 +19,7 @@
  */
 
 #include "libavutil/cpu.h"
+#include "libavutil/intreadwrite.h"
 #include "libavutil/ppc/types_altivec.h"
 #include "libavutil/ppc/util_altivec.h"
 #include "libavcodec/dsputil.h"
@@ -788,7 +789,7 @@ static inline vec_u8 h264_deblock_q1(register vec_u8 p0,
     alphavec = vec_splat(alphavec, 0x0);                                                     \
     mask = h264_deblock_mask(p0, p1, q0, q1, alphavec, betavec); /*if in block */            \
                                                                                              \
-    *((int *)temp) = *((int *)tc0);                                                          \
+    AV_COPY32(temp, tc0);                                                                    \
     tc0vec = vec_ld(0, (signed char*)temp);                                                  \
     tc0vec = vec_mergeh(tc0vec, tc0vec);                                                     \
     tc0vec = vec_mergeh(tc0vec, tc0vec);                                                     \

diff --git a/libavcodec/ppc/h264_altivec_template.c b/libavcodec/ppc/h264_altivec_template.c
@@ -322,7 +322,7 @@ static void PREFIX_h264_qpel16_h_lowpass_altivec(uint8_t * dst, uint8_t * src, i
               pp1A, pp1B, pp2A, pp2B, pp3A, pp3B,
               psumA, psumB, sumA, sumB;
 
-    vec_u8 sum, vdst, fsum;
+    vec_u8 sum, fsum;
 
     for (i = 0 ; i < 16 ; i ++) {
         vec_u8 srcR1 = vec_ld(-2, src);
@@ -423,9 +423,8 @@ static void PREFIX_h264_qpel16_h_lowpass_altivec(uint8_t * dst, uint8_t * src, i
         sum = vec_packsu(sumA, sumB);
 
         ASSERT_ALIGNED(dst);
-        vdst = vec_ld(0, dst);
 
-        OP_U8_ALTIVEC(fsum, sum, vdst);
+        OP_U8_ALTIVEC(fsum, sum, vec_ld(0, dst));
 
         vec_st(fsum, 0, dst);
 
@@ -486,7 +485,7 @@ static void PREFIX_h264_qpel16_v_lowpass_altivec(uint8_t * dst, uint8_t * src, i
               srcP3ssA, srcP3ssB,
               sum1A, sum1B, sum2A, sum2B, sum3A, sum3B;
 
-    vec_u8 sum, vdst, fsum, srcP3a, srcP3b, srcP3;
+    vec_u8 sum, fsum, srcP3a, srcP3b, srcP3;
 
     for (i = 0 ; i < 16 ; i++) {
         srcP3a = vec_ld(0, srcbis += srcStride);
@@ -532,9 +531,8 @@ static void PREFIX_h264_qpel16_v_lowpass_altivec(uint8_t * dst, uint8_t * src, i
         sum = vec_packsu(sumA, sumB);
 
         ASSERT_ALIGNED(dst);
-        vdst = vec_ld(0, dst);
 
-        OP_U8_ALTIVEC(fsum, sum, vdst);
+        OP_U8_ALTIVEC(fsum, sum, vec_ld(0, dst));
 
         vec_st(fsum, 0, dst);
 
@@ -582,7 +580,7 @@ static void PREFIX_h264_qpel16_hv_lowpass_altivec(uint8_t * dst, int16_t * tmp,
               pp3Ae, pp3Ao, pp3Be, pp3Bo, pp1cAe, pp1cAo, pp1cBe, pp1cBo,
               pp32Ae, pp32Ao, pp32Be, pp32Bo, sumAe, sumAo, sumBe, sumBo,
               ssumAe, ssumAo, ssumBe, ssumBo;
-    vec_u8 fsum, sumv, sum, vdst;
+    vec_u8 fsum, sumv, sum;
     vec_s16 ssume, ssumo;
 
     src -= (2 * srcStride);
@@ -766,9 +764,8 @@ static void PREFIX_h264_qpel16_hv_lowpass_altivec(uint8_t * dst, int16_t * tmp,
         sum = vec_perm(sumv, sumv, mperm);
 
         ASSERT_ALIGNED(dst);
-        vdst = vec_ld(0, dst);
 
-        OP_U8_ALTIVEC(fsum, sum, vdst);
+        OP_U8_ALTIVEC(fsum, sum, vec_ld(0, dst));
 
         vec_st(fsum, 0, dst);
 

diff --git a/libavformat/amr.c b/libavformat/amr.c
@@ -25,7 +25,9 @@ Write and read amr data according to RFC3267, http://www.ietf.org/rfc/rfc3267.tx
 Only mono files are supported.
 
 */
+
 #include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
 #include "avformat.h"
 #include "internal.h"
 
@@ -98,6 +100,7 @@ static int amr_read_header(AVFormatContext *s)
         st->codec->sample_rate = 8000;
     }
     st->codec->channels   = 1;
+    st->codec->channel_layout = AV_CH_LAYOUT_MONO;
     st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
     avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
 

diff --git a/libavformat/apc.c b/libavformat/apc.c
@@ -20,6 +20,8 @@
  */
 
 #include <string.h>
+
+#include "libavutil/channel_layout.h"
 #include "avformat.h"
 
 static int apc_probe(AVProbeData *p)
@@ -58,9 +60,13 @@ static int apc_read_header(AVFormatContext *s)
     /* initial predictor values for adpcm decoder */
     avio_read(pb, st->codec->extradata, 2 * 4);
 
-    st->codec->channels = 1;
-    if (avio_rl32(pb))
-        st->codec->channels = 2;
+    if (avio_rl32(pb)) {
+        st->codec->channels       = 2;
+        st->codec->channel_layout = AV_CH_LAYOUT_STEREO;
+    } else {
+        st->codec->channels       = 1;
+        st->codec->channel_layout = AV_CH_LAYOUT_MONO;
+    }
 
     st->codec->bits_per_coded_sample = 4;
     st->codec->bit_rate = st->codec->bits_per_coded_sample * st->codec->channels

diff --git a/libavformat/bethsoftvid.c b/libavformat/bethsoftvid.c
@@ -27,6 +27,7 @@
  * @see http://www.svatopluk.com/andux/docs/dfvid.html
  */
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/intreadwrite.h"
 #include "avformat.h"
 #include "internal.h"
@@ -240,6 +241,7 @@ static int vid_read_packet(AVFormatContext *s,
                 st->codec->codec_type            = AVMEDIA_TYPE_AUDIO;
                 st->codec->codec_id              = AV_CODEC_ID_PCM_U8;
                 st->codec->channels              = 1;
+                st->codec->channel_layout        = AV_CH_LAYOUT_MONO;
                 st->codec->bits_per_coded_sample = 8;
                 st->codec->sample_rate           = vid->sample_rate;
                 st->codec->bit_rate              = 8 * st->codec->sample_rate;

diff --git a/libavformat/bfi.c b/libavformat/bfi.c
@@ -26,6 +26,7 @@
  * @see http://wiki.multimedia.cx/index.php?title=BFI
  */
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/intreadwrite.h"
 #include "avformat.h"
 #include "internal.h"
@@ -96,6 +97,7 @@ static int bfi_read_header(AVFormatContext * s)
     astream->codec->codec_type      = AVMEDIA_TYPE_AUDIO;
     astream->codec->codec_id        = AV_CODEC_ID_PCM_U8;
     astream->codec->channels        = 1;
+    astream->codec->channel_layout  = AV_CH_LAYOUT_MONO;
     astream->codec->bits_per_coded_sample = 8;
     astream->codec->bit_rate        =
         astream->codec->sample_rate * astream->codec->bits_per_coded_sample;

diff --git a/libavformat/bink.c b/libavformat/bink.c
@@ -28,6 +28,7 @@
  *  http://wiki.multimedia.cx/index.php?title=Bink_Container
  */
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/intreadwrite.h"
 #include "avformat.h"
 #include "internal.h"
@@ -143,7 +144,13 @@ static int read_header(AVFormatContext *s)
             flags = avio_rl16(pb);
             ast->codec->codec_id = flags & BINK_AUD_USEDCT ?
                                    AV_CODEC_ID_BINKAUDIO_DCT : AV_CODEC_ID_BINKAUDIO_RDFT;
-            ast->codec->channels = flags & BINK_AUD_STEREO ? 2 : 1;
+            if (flags & BINK_AUD_STEREO) {
+                ast->codec->channels       = 2;
+                ast->codec->channel_layout = AV_CH_LAYOUT_STEREO;
+            } else {
+                ast->codec->channels       = 1;
+                ast->codec->channel_layout = AV_CH_LAYOUT_MONO;
+            }
             ast->codec->extradata = av_mallocz(4 + FF_INPUT_BUFFER_PADDING_SIZE);
             if (!ast->codec->extradata)
                 return AVERROR(ENOMEM);

diff --git a/libavformat/bmv.c b/libavformat/bmv.c
@@ -19,6 +19,7 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/channel_layout.h"
 #include "avformat.h"
 #include "internal.h"
 
@@ -58,6 +59,7 @@ static int bmv_read_header(AVFormatContext *s)
     ast->codec->codec_type      = AVMEDIA_TYPE_AUDIO;
     ast->codec->codec_id        = AV_CODEC_ID_BMV_AUDIO;
     ast->codec->channels        = 2;
+    ast->codec->channel_layout  = AV_CH_LAYOUT_STEREO;
     ast->codec->sample_rate     = 22050;
     avpriv_set_pts_info(ast, 16, 1, 22050);
 

diff --git a/libavformat/cdxl.c b/libavformat/cdxl.c
@@ -19,6 +19,7 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/opt.h"
@@ -143,7 +144,13 @@ static int cdxl_read_packet(AVFormatContext *s, AVPacket *pkt)
             st->codec->codec_type    = AVMEDIA_TYPE_AUDIO;
             st->codec->codec_tag     = 0;
             st->codec->codec_id      = AV_CODEC_ID_PCM_S8;
-            st->codec->channels      = cdxl->header[1] & 0x10 ? 2 : 1;
+            if (cdxl->header[1] & 0x10) {
+                st->codec->channels       = 2;
+                st->codec->channel_layout = AV_CH_LAYOUT_STEREO;
+            } else {
+                st->codec->channels       = 1;
+                st->codec->channel_layout = AV_CH_LAYOUT_MONO;
+            }
             st->codec->sample_rate   = cdxl->sample_rate;
             st->start_time           = 0;
             cdxl->audio_stream_index = st->index;

diff --git a/libavformat/daud.c b/libavformat/daud.c
@@ -18,6 +18,8 @@
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
+
+#include "libavutil/channel_layout.h"
 #include "avformat.h"
 
 static int daud_header(AVFormatContext *s) {
@@ -28,6 +30,7 @@ static int daud_header(AVFormatContext *s) {
     st->codec->codec_id = AV_CODEC_ID_PCM_S24DAUD;
     st->codec->codec_tag = MKTAG('d', 'a', 'u', 'd');
     st->codec->channels = 6;
+    st->codec->channel_layout = AV_CH_LAYOUT_5POINT1;
     st->codec->sample_rate = 96000;
     st->codec->bit_rate = 3 * 6 * 96000 * 8;
     st->codec->block_align = 3 * 6;

diff --git a/libavformat/dsicin.c b/libavformat/dsicin.c
@@ -24,6 +24,7 @@
  * Delphine Software International CIN file demuxer
  */
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/intreadwrite.h"
 #include "avformat.h"
 #include "internal.h"
@@ -132,6 +133,7 @@ static int cin_read_header(AVFormatContext *s)
     st->codec->codec_id = AV_CODEC_ID_DSICINAUDIO;
     st->codec->codec_tag = 0;  /* no tag */
     st->codec->channels = 1;
+    st->codec->channel_layout = AV_CH_LAYOUT_MONO;
     st->codec->sample_rate = 22050;
     st->codec->bits_per_coded_sample = 8;
     st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_coded_sample * st->codec->channels;

diff --git a/libavformat/dv.c b/libavformat/dv.c
@@ -33,6 +33,7 @@
 #include "internal.h"
 #include "libavcodec/dv_profile.h"
 #include "libavcodec/dvdata.h"
+#include "libavutil/channel_layout.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/timecode.h"
@@ -252,6 +253,7 @@ static int dv_extract_audio_info(DVDemuxContext* c, uint8_t* frame)
         }
         c->ast[i]->codec->sample_rate = dv_audio_frequency[freq];
         c->ast[i]->codec->channels    = 2;
+        c->ast[i]->codec->channel_layout = AV_CH_LAYOUT_STEREO;
         c->ast[i]->codec->bit_rate    = 2 * dv_audio_frequency[freq] * 16;
         c->ast[i]->start_time         = 0;
     }

diff --git a/libavformat/flvdec.c b/libavformat/flvdec.c
@@ -25,6 +25,7 @@
  */
 
 #include "libavutil/avstring.h"
+#include "libavutil/channel_layout.h"
 #include "libavutil/dict.h"
 #include "libavutil/opt.h"
 #include "libavutil/intfloat.h"
@@ -778,6 +779,8 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
         bits_per_coded_sample = (flags & FLV_AUDIO_SAMPLESIZE_MASK) ? 16 : 8;
         if(!st->codec->channels || !st->codec->sample_rate || !st->codec->bits_per_coded_sample) {
             st->codec->channels              = channels;
+            st->codec->channel_layout        = channels == 1 ? AV_CH_LAYOUT_MONO :
+                                                               AV_CH_LAYOUT_STEREO;
             st->codec->sample_rate           = sample_rate;
             st->codec->bits_per_coded_sample = bits_per_coded_sample;
         }
@@ -824,6 +827,7 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
                 if (avpriv_mpeg4audio_get_config(&cfg, st->codec->extradata,
                                              st->codec->extradata_size * 8, 1) >= 0) {
                 st->codec->channels = cfg.channels;
+                st->codec->channel_layout = 0;
                 if (cfg.ext_sample_rate)
                     st->codec->sample_rate = cfg.ext_sample_rate;
                 else

diff --git a/libavformat/gsmdec.c b/libavformat/gsmdec.c
@@ -19,6 +19,7 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/opt.h"
 #include "avformat.h"
@@ -63,6 +64,7 @@ static int gsm_read_header(AVFormatContext *s)
     st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
     st->codec->codec_id    = s->iformat->raw_codec_id;
     st->codec->channels    = 1;
+    st->codec->channel_layout = AV_CH_LAYOUT_MONO;
     st->codec->sample_rate = c->sample_rate;
     st->codec->bit_rate    = GSM_BLOCK_SIZE * 8 * c->sample_rate / GSM_BLOCK_SAMPLES;
 

diff --git a/libavformat/gxf.c b/libavformat/gxf.c
@@ -19,6 +19,7 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/common.h"
 #include "avformat.h"
 #include "internal.h"
@@ -141,6 +142,7 @@ static int get_sindex(AVFormatContext *s, int id, int format) {
             st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
             st->codec->codec_id = AV_CODEC_ID_PCM_S24LE;
             st->codec->channels = 1;
+            st->codec->channel_layout = AV_CH_LAYOUT_MONO;
             st->codec->sample_rate = 48000;
             st->codec->bit_rate = 3 * 1 * 48000 * 8;
             st->codec->block_align = 3 * 1;
@@ -150,6 +152,7 @@ static int get_sindex(AVFormatContext *s, int id, int format) {
             st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
             st->codec->codec_id = AV_CODEC_ID_PCM_S16LE;
             st->codec->channels = 1;
+            st->codec->channel_layout = AV_CH_LAYOUT_MONO;
             st->codec->sample_rate = 48000;
             st->codec->bit_rate = 2 * 1 * 48000 * 8;
             st->codec->block_align = 2 * 1;
@@ -159,6 +162,7 @@ static int get_sindex(AVFormatContext *s, int id, int format) {
             st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
             st->codec->codec_id = AV_CODEC_ID_AC3;
             st->codec->channels = 2;
+            st->codec->channel_layout = AV_CH_LAYOUT_STEREO;
             st->codec->sample_rate = 48000;
             break;
         // timecode tracks:

diff --git a/libavformat/idroqdec.c b/libavformat/idroqdec.c
@@ -27,6 +27,7 @@
  *   http://www.csse.monash.edu.au/~timf/
  */
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/intreadwrite.h"
 #include "avformat.h"
 #include "internal.h"
@@ -178,7 +179,14 @@ static int roq_read_packet(AVFormatContext *s,
                 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
                 st->codec->codec_id = AV_CODEC_ID_ROQ_DPCM;
                 st->codec->codec_tag = 0;  /* no tag */
-                st->codec->channels = roq->audio_channels = chunk_type == RoQ_SOUND_STEREO ? 2 : 1;
+                if (chunk_type == RoQ_SOUND_STEREO) {
+                    st->codec->channels       = 2;
+                    st->codec->channel_layout = AV_CH_LAYOUT_STEREO;
+                } else {
+                    st->codec->channels       = 1;
+                    st->codec->channel_layout = AV_CH_LAYOUT_MONO;
+                }
+                roq->audio_channels    = st->codec->channels;
                 st->codec->sample_rate = RoQ_AUDIO_SAMPLE_RATE;
                 st->codec->bits_per_coded_sample = 16;
                 st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *