Bump Major version, this commit is almost just renaming bits_per_sample to

bits_per_coded_sample but that cannot be done seperately.
Patch by Luca Abeni
Also reset the minor version and fix the forgotton change to libfaad.
Note: The API/ABI should not be considered stable yet, there still may
be a change done here or there if some developer has some cleanup ideas and
patches!

Originally committed as revision 15262 to svn://svn.ffmpeg.org/ffmpeg/trunk

47 files changed, 137 insertions, 137 deletions

diff --git a/libavformat/4xm.c b/libavformat/4xm.c
index 25c2080d7f..f8537165b5 100644
--- a/libavformat/4xm.c
+++ b/libavformat/4xm.c
@@ -193,13 +193,13 @@ static int fourxm_read_header(AVFormatContext *s,
             st->codec->codec_tag = 0;
             st->codec->channels = fourxm->tracks[current_track].channels;
             st->codec->sample_rate = fourxm->tracks[current_track].sample_rate;
-            st->codec->bits_per_sample = fourxm->tracks[current_track].bits;
+            st->codec->bits_per_coded_sample = fourxm->tracks[current_track].bits;
             st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-                st->codec->bits_per_sample;
-            st->codec->block_align = st->codec->channels * st->codec->bits_per_sample;
+                st->codec->bits_per_coded_sample;
+            st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
             if (fourxm->tracks[current_track].adpcm)
                 st->codec->codec_id = CODEC_ID_ADPCM_4XM;
-            else if (st->codec->bits_per_sample == 8)
+            else if (st->codec->bits_per_coded_sample == 8)
                 st->codec->codec_id = CODEC_ID_PCM_U8;
             else
                 st->codec->codec_id = CODEC_ID_PCM_S16LE;
diff --git a/libavformat/aiff.c b/libavformat/aiff.c
index 0eedfba88c..911814bf92 100644
--- a/libavformat/aiff.c
+++ b/libavformat/aiff.c
@@ -112,7 +112,7 @@ static unsigned int get_aiff_header(ByteIOContext *pb, AVCodecContext *codec,
     codec->codec_type = CODEC_TYPE_AUDIO;
     codec->channels = get_be16(pb);
     num_frames = get_be32(pb);
-    codec->bits_per_sample = get_be16(pb);
+    codec->bits_per_coded_sample = get_be16(pb);
 
     get_buffer(pb, (uint8_t*)&ext, sizeof(ext));/* Sample rate is in */
     sample_rate = av_ext2dbl(ext);          /* 80 bits BE IEEE extended float */
@@ -126,8 +126,8 @@ static unsigned int get_aiff_header(ByteIOContext *pb, AVCodecContext *codec,
 
         switch (codec->codec_id) {
         case CODEC_ID_PCM_S16BE:
-            codec->codec_id = aiff_codec_get_id(codec->bits_per_sample);
-            codec->bits_per_sample = av_get_bits_per_sample(codec->codec_id);
+            codec->codec_id = aiff_codec_get_id(codec->bits_per_coded_sample);
+            codec->bits_per_coded_sample = av_get_bits_per_sample(codec->codec_id);
             break;
         case CODEC_ID_ADPCM_IMA_QT:
             codec->block_align = 34*codec->channels;
@@ -151,14 +151,14 @@ static unsigned int get_aiff_header(ByteIOContext *pb, AVCodecContext *codec,
         size -= 4;
     } else {
         /* Need the codec type */
-        codec->codec_id = aiff_codec_get_id(codec->bits_per_sample);
-        codec->bits_per_sample = av_get_bits_per_sample(codec->codec_id);
+        codec->codec_id = aiff_codec_get_id(codec->bits_per_coded_sample);
+        codec->bits_per_coded_sample = av_get_bits_per_sample(codec->codec_id);
     }
 
     /* Block align needs to be computed in all cases, as the definition
      * is specific to applications -> here we use the WAVE format definition */
     if (!codec->block_align)
-        codec->block_align = (codec->bits_per_sample * codec->channels) >> 3;
+        codec->block_align = (codec->bits_per_coded_sample * codec->channels) >> 3;
 
     codec->bit_rate = (codec->frame_size ? codec->sample_rate/codec->frame_size :
                        codec->sample_rate) * (codec->block_align << 3);
@@ -198,7 +198,7 @@ static int aiff_write_header(AVFormatContext *s)
     put_tag(pb, aifc ? "AIFC" : "AIFF");
 
     if (aifc) { // compressed audio
-        enc->bits_per_sample = 16;
+        enc->bits_per_coded_sample = 16;
         if (!enc->block_align) {
             av_log(s, AV_LOG_ERROR, "block align not set\n");
             return -1;
@@ -217,16 +217,16 @@ static int aiff_write_header(AVFormatContext *s)
     aiff->frames = url_ftell(pb);
     put_be32(pb, 0);              /* Number of frames */
 
-    if (!enc->bits_per_sample)
-        enc->bits_per_sample = av_get_bits_per_sample(enc->codec_id);
-    if (!enc->bits_per_sample) {
+    if (!enc->bits_per_coded_sample)
+        enc->bits_per_coded_sample = av_get_bits_per_sample(enc->codec_id);
+    if (!enc->bits_per_coded_sample) {
         av_log(s, AV_LOG_ERROR, "could not compute bits per sample\n");
         return -1;
     }
     if (!enc->block_align)
-        enc->block_align = (enc->bits_per_sample * enc->channels) >> 3;
+        enc->block_align = (enc->bits_per_coded_sample * enc->channels) >> 3;
 
-    put_be16(pb, enc->bits_per_sample); /* Sample size */
+    put_be16(pb, enc->bits_per_coded_sample); /* Sample size */
 
     sample_rate = av_dbl2ext((double)enc->sample_rate);
     put_buffer(pb, (uint8_t*)&sample_rate, sizeof(sample_rate));
diff --git a/libavformat/apc.c b/libavformat/apc.c
index 20de1c7e58..14701d9229 100644
--- a/libavformat/apc.c
+++ b/libavformat/apc.c
@@ -62,8 +62,8 @@ static int apc_read_header(AVFormatContext *s, AVFormatParameters *ap)
     if (get_le32(pb))
         st->codec->channels = 2;
 
-    st->codec->bits_per_sample = 4;
-    st->codec->bit_rate = st->codec->bits_per_sample * st->codec->channels
+    st->codec->bits_per_coded_sample = 4;
+    st->codec->bit_rate = st->codec->bits_per_coded_sample * st->codec->channels
                           * st->codec->sample_rate;
     st->codec->block_align = 1;
 
diff --git a/libavformat/ape.c b/libavformat/ape.c
index 4c5d0461e2..5ab92b10e1 100644
--- a/libavformat/ape.c
+++ b/libavformat/ape.c
@@ -424,7 +424,7 @@ static int ape_read_header(AVFormatContext * s, AVFormatParameters * ap)
     st->codec->codec_tag       = MKTAG('A', 'P', 'E', ' ');
     st->codec->channels        = ape->channels;
     st->codec->sample_rate     = ape->samplerate;
-    st->codec->bits_per_sample = ape->bps;
+    st->codec->bits_per_coded_sample = ape->bps;
     st->codec->frame_size      = MAC_SUBFRAME_SIZE;
 
     st->nb_frames = ape->totalframes;
diff --git a/libavformat/asf.c b/libavformat/asf.c
index 6d35b8f56d..b837e76891 100644
--- a/libavformat/asf.c
+++ b/libavformat/asf.c
@@ -315,7 +315,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
                 st->codec->height = get_le32(pb);
                 /* not available for asf */
                 get_le16(pb); /* panes */
-                st->codec->bits_per_sample = get_le16(pb); /* depth */
+                st->codec->bits_per_coded_sample = get_le16(pb); /* depth */
                 tag1 = get_le32(pb);
                 url_fskip(pb, 20);
 //                av_log(NULL, AV_LOG_DEBUG, "size:%d tsize:%d sizeX:%d\n", size, total_size, sizeX);
@@ -329,7 +329,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
         /* Extract palette from extradata if bpp <= 8 */
         /* This code assumes that extradata contains only palette */
         /* This is true for all paletted codecs implemented in ffmpeg */
-        if (st->codec->extradata_size && (st->codec->bits_per_sample <= 8)) {
+        if (st->codec->extradata_size && (st->codec->bits_per_coded_sample <= 8)) {
             st->codec->palctrl = av_mallocz(sizeof(AVPaletteControl));
 #ifdef WORDS_BIGENDIAN
             for (i = 0; i < FFMIN(st->codec->extradata_size, AVPALETTE_SIZE)/4; i++)
diff --git a/libavformat/avidec.c b/libavformat/avidec.c
index ae19e53033..b4494f7fbd 100644
--- a/libavformat/avidec.c
+++ b/libavformat/avidec.c
@@ -445,7 +445,7 @@ static int avi_read_header(AVFormatContext *s, AVFormatParameters *ap)
                     st->codec->width = get_le32(pb);
                     st->codec->height = get_le32(pb);
                     get_le16(pb); /* panes */
-                    st->codec->bits_per_sample= get_le16(pb); /* depth */
+                    st->codec->bits_per_coded_sample= get_le16(pb); /* depth */
                     tag1 = get_le32(pb);
                     get_le32(pb); /* ImageSize */
                     get_le32(pb); /* XPelsPerMeter */
@@ -472,7 +472,7 @@ static int avi_read_header(AVFormatContext *s, AVFormatParameters *ap)
                     /* Extract palette from extradata if bpp <= 8. */
                     /* This code assumes that extradata contains only palette. */
                     /* This is true for all paletted codecs implemented in FFmpeg. */
-                    if (st->codec->extradata_size && (st->codec->bits_per_sample <= 8)) {
+                    if (st->codec->extradata_size && (st->codec->bits_per_coded_sample <= 8)) {
                         st->codec->palctrl = av_mallocz(sizeof(AVPaletteControl));
 #ifdef WORDS_BIGENDIAN
                         for (i = 0; i < FFMIN(st->codec->extradata_size, AVPALETTE_SIZE)/4; i++)
diff --git a/libavformat/avs.c b/libavformat/avs.c
index 16be4dc617..6fcb230100 100644
--- a/libavformat/avs.c
+++ b/libavformat/avs.c
@@ -182,7 +182,7 @@ static int avs_read_packet(AVFormatContext * s, AVPacket * pkt)
                     avs->st_video->codec->codec_id = CODEC_ID_AVS;
                     avs->st_video->codec->width = avs->width;
                     avs->st_video->codec->height = avs->height;
-                    avs->st_video->codec->bits_per_sample=avs->bits_per_sample;
+                    avs->st_video->codec->bits_per_coded_sample=avs->bits_per_sample;
                     avs->st_video->nb_frames = avs->nb_frames;
                     avs->st_video->codec->time_base = (AVRational) {
                     1, avs->fps};
diff --git a/libavformat/bethsoftvid.c b/libavformat/bethsoftvid.c
index e3024c22b8..a80b105e2c 100644
--- a/libavformat/bethsoftvid.c
+++ b/libavformat/bethsoftvid.c
@@ -89,8 +89,8 @@ static int vid_read_header(AVFormatContext *s,
     stream->codec->codec_id = CODEC_ID_PCM_U8;
     stream->codec->channels = 1;
     stream->codec->sample_rate = 11025;
-    stream->codec->bits_per_sample = 8;
-    stream->codec->bit_rate = stream->codec->channels * stream->codec->sample_rate * stream->codec->bits_per_sample;
+    stream->codec->bits_per_coded_sample = 8;
+    stream->codec->bit_rate = stream->codec->channels * stream->codec->sample_rate * stream->codec->bits_per_coded_sample;
 
     return 0;
 }
@@ -197,7 +197,7 @@ static int vid_read_packet(AVFormatContext *s,
             get_le16(pb);
             // soundblaster DAC used for sample rate, as on specification page (link above)
             s->streams[1]->codec->sample_rate = 1000000 / (256 - get_byte(pb));
-            s->streams[1]->codec->bit_rate = s->streams[1]->codec->channels * s->streams[1]->codec->sample_rate * s->streams[1]->codec->bits_per_sample;
+            s->streams[1]->codec->bit_rate = s->streams[1]->codec->channels * s->streams[1]->codec->sample_rate * s->streams[1]->codec->bits_per_coded_sample;
         case AUDIO_BLOCK:
             audio_length = get_le16(pb);
             ret_value = av_get_packet(pb, pkt, audio_length);
diff --git a/libavformat/bfi.c b/libavformat/bfi.c
index 5a8d56919d..c6efdd47e0 100644
--- a/libavformat/bfi.c
+++ b/libavformat/bfi.c
@@ -94,9 +94,9 @@ static int bfi_read_header(AVFormatContext * s, AVFormatParameters * ap)
     astream->codec->codec_type      = CODEC_TYPE_AUDIO;
     astream->codec->codec_id        = CODEC_ID_PCM_U8;
     astream->codec->channels        = 1;
-    astream->codec->bits_per_sample = 8;
+    astream->codec->bits_per_coded_sample = 8;
     astream->codec->bit_rate        =
-        astream->codec->sample_rate * astream->codec->bits_per_sample;
+        astream->codec->sample_rate * astream->codec->bits_per_coded_sample;
     url_fseek(pb, chunk_header - 3, SEEK_SET);
     av_set_pts_info(astream, 64, 1, astream->codec->sample_rate);
     return 0;
diff --git a/libavformat/daud.c b/libavformat/daud.c
index c0626dfb24..2bca9478ea 100644
--- a/libavformat/daud.c
+++ b/libavformat/daud.c
@@ -31,7 +31,7 @@ static int daud_header(AVFormatContext *s, AVFormatParameters *ap) {
     st->codec->sample_rate = 96000;
     st->codec->bit_rate = 3 * 6 * 96000 * 8;
     st->codec->block_align = 3 * 6;
-    st->codec->bits_per_sample = 24;
+    st->codec->bits_per_coded_sample = 24;
     return 0;
 }
 
diff --git a/libavformat/dsicin.c b/libavformat/dsicin.c
index eef205cf55..945bacbb47 100644
--- a/libavformat/dsicin.c
+++ b/libavformat/dsicin.c
@@ -130,9 +130,9 @@ static int cin_read_header(AVFormatContext *s, AVFormatParameters *ap)
     st->codec->codec_tag = 0;  /* no tag */
     st->codec->channels = 1;
     st->codec->sample_rate = 22050;
-    st->codec->bits_per_sample = 16;
-    st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_sample * st->codec->channels;
-    st->codec->block_align = st->codec->channels * st->codec->bits_per_sample;
+    st->codec->bits_per_coded_sample = 16;
+    st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_coded_sample * st->codec->channels;
+    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
 
     return 0;
 }
diff --git a/libavformat/electronicarts.c b/libavformat/electronicarts.c
index 2e627792c3..09da524ffd 100644
--- a/libavformat/electronicarts.c
+++ b/libavformat/electronicarts.c
@@ -411,10 +411,10 @@ static int ea_read_header(AVFormatContext *s,
         st->codec->codec_tag = 0;  /* no tag */
         st->codec->channels = ea->num_channels;
         st->codec->sample_rate = ea->sample_rate;
-        st->codec->bits_per_sample = ea->bytes * 8;
+        st->codec->bits_per_coded_sample = ea->bytes * 8;
         st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-            st->codec->bits_per_sample / 4;
-        st->codec->block_align = st->codec->channels*st->codec->bits_per_sample;
+            st->codec->bits_per_coded_sample / 4;
+        st->codec->block_align = st->codec->channels*st->codec->bits_per_coded_sample;
         ea->audio_stream_index = st->index;
         ea->audio_frame_counter = 0;
     }
diff --git a/libavformat/flvdec.c b/libavformat/flvdec.c
index ff6f98106d..e548fb4a85 100644
--- a/libavformat/flvdec.c
+++ b/libavformat/flvdec.c
@@ -42,7 +42,7 @@ static void flv_set_audio_codec(AVFormatContext *s, AVStream *astream, int flv_c
     switch(flv_codecid) {
         //no distinction between S16 and S8 PCM codec flags
         case FLV_CODECID_PCM:
-            acodec->codec_id = acodec->bits_per_sample == 8 ? CODEC_ID_PCM_S8 :
+            acodec->codec_id = acodec->bits_per_coded_sample == 8 ? CODEC_ID_PCM_S8 :
 #ifdef WORDS_BIGENDIAN
                                 CODEC_ID_PCM_S16BE;
 #else
@@ -50,7 +50,7 @@ static void flv_set_audio_codec(AVFormatContext *s, AVStream *astream, int flv_c
 #endif
             break;
         case FLV_CODECID_PCM_LE:
-            acodec->codec_id = acodec->bits_per_sample == 8 ? CODEC_ID_PCM_S8 : CODEC_ID_PCM_S16LE; break;
+            acodec->codec_id = acodec->bits_per_coded_sample == 8 ? CODEC_ID_PCM_S8 : CODEC_ID_PCM_S16LE; break;
         case FLV_CODECID_AAC  : acodec->codec_id = CODEC_ID_AAC;                                    break;
         case FLV_CODECID_ADPCM: acodec->codec_id = CODEC_ID_ADPCM_SWF;                              break;
         case FLV_CODECID_SPEEX: acodec->codec_id = CODEC_ID_SPEEX;                                  break;
@@ -185,7 +185,7 @@ static int amf_parse_object(AVFormatContext *s, AVStream *astream, AVStream *vst
             else if(!strcmp(key, "videocodecid") && vcodec && 0 <= (int)num_val)
                 flv_set_video_codec(s, vstream, (int)num_val);
             else if(!strcmp(key, "audiosamplesize") && acodec && 0 < (int)num_val) {
-                acodec->bits_per_sample = num_val;
+                acodec->bits_per_coded_sample = num_val;
                 //we may have to rewrite a previously read codecid because FLV only marks PCM endianness.
                 if(num_val == 8 && (acodec->codec_id == CODEC_ID_PCM_S16BE || acodec->codec_id == CODEC_ID_PCM_S16LE))
                     acodec->codec_id = CODEC_ID_PCM_S8;
@@ -382,13 +382,13 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
     }
 
     if(is_audio){
-        if(!st->codec->sample_rate || !st->codec->bits_per_sample || (!st->codec->codec_id && !st->codec->codec_tag)) {
+        if(!st->codec->sample_rate || !st->codec->bits_per_coded_sample || (!st->codec->codec_id && !st->codec->codec_tag)) {
             st->codec->channels = (flags & FLV_AUDIO_CHANNEL_MASK) == FLV_STEREO ? 2 : 1;
             if((flags & FLV_AUDIO_CODECID_MASK) == FLV_CODECID_NELLYMOSER_8HZ_MONO)
                 st->codec->sample_rate= 8000;
             else
                 st->codec->sample_rate = (44100 << ((flags & FLV_AUDIO_SAMPLERATE_MASK) >> FLV_AUDIO_SAMPLERATE_OFFSET) >> 3);
-            st->codec->bits_per_sample = (flags & FLV_AUDIO_SAMPLESIZE_MASK) ? 16 : 8;
+            st->codec->bits_per_coded_sample = (flags & FLV_AUDIO_SAMPLESIZE_MASK) ? 16 : 8;
             flv_set_audio_codec(s, st, flags & FLV_AUDIO_CODECID_MASK);
         }
     }else{
diff --git a/libavformat/flvenc.c b/libavformat/flvenc.c
index 81c43bf4c9..16c18d9c51 100644
--- a/libavformat/flvenc.c
+++ b/libavformat/flvenc.c
@@ -55,7 +55,7 @@ typedef struct FLVContext {
 } FLVContext;
 
 static int get_audio_flags(AVCodecContext *enc){
-    int flags = (enc->bits_per_sample == 16) ? FLV_SAMPLESSIZE_16BIT : FLV_SAMPLESSIZE_8BIT;
+    int flags = (enc->bits_per_coded_sample == 16) ? FLV_SAMPLESSIZE_16BIT : FLV_SAMPLESSIZE_8BIT;
 
     if (enc->codec_id == CODEC_ID_AAC) // specs force these parameters
         return FLV_CODECID_AAC | FLV_SAMPLERATE_44100HZ | FLV_SAMPLESSIZE_16BIT | FLV_STEREO;
diff --git a/libavformat/gxf.c b/libavformat/gxf.c
index 43193beb72..dbc069c193 100644
--- a/libavformat/gxf.c
+++ b/libavformat/gxf.c
@@ -120,7 +120,7 @@ static int get_sindex(AVFormatContext *s, int id, int format) {
             st->codec->sample_rate = 48000;
             st->codec->bit_rate = 3 * 1 * 48000 * 8;
             st->codec->block_align = 3 * 1;
-            st->codec->bits_per_sample = 24;
+            st->codec->bits_per_coded_sample = 24;
             break;
         case 10:
             st->codec->codec_type = CODEC_TYPE_AUDIO;
@@ -129,7 +129,7 @@ static int get_sindex(AVFormatContext *s, int id, int format) {
             st->codec->sample_rate = 48000;
             st->codec->bit_rate = 2 * 1 * 48000 * 8;
             st->codec->block_align = 2 * 1;
-            st->codec->bits_per_sample = 16;
+            st->codec->bits_per_coded_sample = 16;
             break;
         case 17:
             st->codec->codec_type = CODEC_TYPE_AUDIO;
diff --git a/libavformat/idcin.c b/libavformat/idcin.c
index 1a8e558e5d..55ad28b080 100644
--- a/libavformat/idcin.c
+++ b/libavformat/idcin.c
@@ -181,7 +181,7 @@ static int idcin_read_header(AVFormatContext *s,
         st->codec->codec_tag = 1;
         st->codec->channels = channels;
         st->codec->sample_rate = sample_rate;
-        st->codec->bits_per_sample = bytes_per_sample * 8;
+        st->codec->bits_per_coded_sample = bytes_per_sample * 8;
         st->codec->bit_rate = sample_rate * bytes_per_sample * 8 * channels;
         st->codec->block_align = bytes_per_sample * channels;
         if (bytes_per_sample == 1)
diff --git a/libavformat/idroq.c b/libavformat/idroq.c
index 801f3dc450..b5dae21b04 100644
--- a/libavformat/idroq.c
+++ b/libavformat/idroq.c
@@ -161,10 +161,10 @@ static int roq_read_header(AVFormatContext *s,
         st->codec->codec_tag = 0;  /* no tag */
         st->codec->channels = roq->audio_channels;
         st->codec->sample_rate = RoQ_AUDIO_SAMPLE_RATE;
-        st->codec->bits_per_sample = 16;
+        st->codec->bits_per_coded_sample = 16;
         st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-            st->codec->bits_per_sample;
-        st->codec->block_align = st->codec->channels * st->codec->bits_per_sample;
+            st->codec->bits_per_coded_sample;
+        st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
     }
 
     return 0;
diff --git a/libavformat/iff.c b/libavformat/iff.c
index a996a7fbc2..1fb94c057a 100644
--- a/libavformat/iff.c
+++ b/libavformat/iff.c
@@ -149,9 +149,9 @@ static int iff_read_header(AVFormatContext *s,
         return -1;
     }
 
-    st->codec->bits_per_sample = 8;
-    st->codec->bit_rate = st->codec->channels * st->codec->sample_rate * st->codec->bits_per_sample;
-    st->codec->block_align = st->codec->channels * st->codec->bits_per_sample;
+    st->codec->bits_per_coded_sample = 8;
+    st->codec->bit_rate = st->codec->channels * st->codec->sample_rate * st->codec->bits_per_coded_sample;
+    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
 
     return 0;
 }
diff --git a/libavformat/ipmovie.c b/libavformat/ipmovie.c
index c2c5585daa..839e099a5e 100644
--- a/libavformat/ipmovie.c
+++ b/libavformat/ipmovie.c
@@ -574,12 +574,12 @@ static int ipmovie_read_header(AVFormatContext *s,
         st->codec->codec_tag = 0;  /* no tag */
         st->codec->channels = ipmovie->audio_channels;
         st->codec->sample_rate = ipmovie->audio_sample_rate;
-        st->codec->bits_per_sample = ipmovie->audio_bits;
+        st->codec->bits_per_coded_sample = ipmovie->audio_bits;
         st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-            st->codec->bits_per_sample;
+            st->codec->bits_per_coded_sample;
         if (st->codec->codec_id == CODEC_ID_INTERPLAY_DPCM)
             st->codec->bit_rate /= 2;
-        st->codec->block_align = st->codec->channels * st->codec->bits_per_sample;
+        st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
     }
 
     return 0;
diff --git a/libavformat/mmf.c b/libavformat/mmf.c
index d23df63454..3941bee14a 100644
--- a/libavformat/mmf.c
+++ b/libavformat/mmf.c
@@ -248,8 +248,8 @@ static int mmf_read_header(AVFormatContext *s,
     st->codec->codec_id = CODEC_ID_ADPCM_YAMAHA;
     st->codec->sample_rate = rate;
     st->codec->channels = 1;
-    st->codec->bits_per_sample = 4;
-    st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_sample;
+    st->codec->bits_per_coded_sample = 4;
+    st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_coded_sample;
 
     av_set_pts_info(st, 64, 1, st->codec->sample_rate);
 
diff --git a/libavformat/mov.c b/libavformat/mov.c
index 2cd73d702f..73ed11ddad 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -796,13 +796,13 @@ static int mov_read_stsd(MOVContext *c, ByteIOContext *pb, MOV_atom_t atom)
                 st->codec->codec_name[codec_name[0]] = 0;
             }
 
-            st->codec->bits_per_sample = get_be16(pb); /* depth */
+            st->codec->bits_per_coded_sample = get_be16(pb); /* depth */
             st->codec->color_table_id = get_be16(pb); /* colortable id */
             dprintf(c->fc, "depth %d, ctab id %d\n",
-                   st->codec->bits_per_sample, st->codec->color_table_id);
+                   st->codec->bits_per_coded_sample, st->codec->color_table_id);
             /* figure out the palette situation */
-            color_depth = st->codec->bits_per_sample & 0x1F;
-            color_greyscale = st->codec->bits_per_sample & 0x20;
+            color_depth = st->codec->bits_per_coded_sample & 0x1F;
+            color_greyscale = st->codec->bits_per_coded_sample & 0x20;
 
             /* if the depth is 2, 4, or 8 bpp, file is palettized */
             if ((color_depth == 2) || (color_depth == 4) ||
@@ -814,7 +814,7 @@ static int mov_read_stsd(MOVContext *c, ByteIOContext *pb, MOV_atom_t atom)
                 if (color_greyscale) {
                     int color_index, color_dec;
                     /* compute the greyscale palette */
-                    st->codec->bits_per_sample = color_depth;
+                    st->codec->bits_per_coded_sample = color_depth;
                     color_count = 1 << color_depth;
                     color_index = 255;
                     color_dec = 256 / (color_count - 1);
@@ -882,7 +882,7 @@ static int mov_read_stsd(MOVContext *c, ByteIOContext *pb, MOV_atom_t atom)
 
             st->codec->channels = get_be16(pb);             /* channel count */
             dprintf(c->fc, "audio channels %d\n", st->codec->channels);
-            st->codec->bits_per_sample = get_be16(pb);      /* sample size */
+            st->codec->bits_per_coded_sample = get_be16(pb);      /* sample size */
 
             sc->audio_cid = get_be16(pb);
             get_be16(pb); /* packet size = 0 */
@@ -902,26 +902,26 @@ static int mov_read_stsd(MOVContext *c, ByteIOContext *pb, MOV_atom_t atom)
                     st->codec->sample_rate = av_int2dbl(get_be64(pb)); /* float 64 */
                     st->codec->channels = get_be32(pb);
                     get_be32(pb); /* always 0x7F000000 */
-                    st->codec->bits_per_sample = get_be32(pb); /* bits per channel if sound is uncompressed */
+                    st->codec->bits_per_coded_sample = get_be32(pb); /* bits per channel if sound is uncompressed */
                     flags = get_be32(pb); /* lcpm format specific flag */
                     sc->bytes_per_frame = get_be32(pb); /* bytes per audio packet if constant */
                     sc->samples_per_frame = get_be32(pb); /* lpcm frames per audio packet if constant */
                     if (format == MKTAG('l','p','c','m'))
-                        st->codec->codec_id = mov_get_lpcm_codec_id(st->codec->bits_per_sample, flags);
+                        st->codec->codec_id = mov_get_lpcm_codec_id(st->codec->bits_per_coded_sample, flags);
                 }
             }
 
             switch (st->codec->codec_id) {
             case CODEC_ID_PCM_S8:
             case CODEC_ID_PCM_U8:
-                if (st->codec->bits_per_sample == 16)
+                if (st->codec->bits_per_coded_sample == 16)
                     st->codec->codec_id = CODEC_ID_PCM_S16BE;
                 break;
             case CODEC_ID_PCM_S16LE:
             case CODEC_ID_PCM_S16BE:
-                if (st->codec->bits_per_sample == 8)
+                if (st->codec->bits_per_coded_sample == 8)
                     st->codec->codec_id = CODEC_ID_PCM_S8;
-                else if (st->codec->bits_per_sample == 24)
+                else if (st->codec->bits_per_coded_sample == 24)
                     st->codec->codec_id =
                         st->codec->codec_id == CODEC_ID_PCM_S16BE ?
                         CODEC_ID_PCM_S24BE : CODEC_ID_PCM_S24LE;
@@ -949,7 +949,7 @@ static int mov_read_stsd(MOVContext *c, ByteIOContext *pb, MOV_atom_t atom)
 
             bits_per_sample = av_get_bits_per_sample(st->codec->codec_id);
             if (bits_per_sample) {
-                st->codec->bits_per_sample = bits_per_sample;
+                st->codec->bits_per_coded_sample = bits_per_sample;
                 sc->sample_size = (bits_per_sample >> 3) * st->codec->channels;
             }
         } else if(st->codec->codec_type==CODEC_TYPE_SUBTITLE){
diff --git a/libavformat/movenc.c b/libavformat/movenc.c
index f1afa82a5c..8ddc8e9e24 100644
--- a/libavformat/movenc.c
+++ b/libavformat/movenc.c
@@ -684,8 +684,8 @@ static int mov_write_video_tag(ByteIOContext *pb, MOVTrack *track)
     put_byte(pb, strlen(compressor_name));
     put_buffer(pb, compressor_name, 31);
 
-    if (track->mode == MODE_MOV && track->enc->bits_per_sample)
-        put_be16(pb, track->enc->bits_per_sample);
+    if (track->mode == MODE_MOV && track->enc->bits_per_coded_sample)
+        put_be16(pb, track->enc->bits_per_coded_sample);
     else
         put_be16(pb, 0x18); /* Reserved */
     put_be16(pb, 0xffff); /* Reserved */
diff --git a/libavformat/mpc.c b/libavformat/mpc.c
index 7ed574d866..1a22c113b0 100644
--- a/libavformat/mpc.c
+++ b/libavformat/mpc.c
@@ -97,7 +97,7 @@ static int mpc_read_header(AVFormatContext *s, AVFormatParameters *ap)
     st->codec->codec_type = CODEC_TYPE_AUDIO;
     st->codec->codec_id = CODEC_ID_MUSEPACK7;
     st->codec->channels = 2;
-    st->codec->bits_per_sample = 16;
+    st->codec->bits_per_coded_sample = 16;
 
     st->codec->extradata_size = 16;
     st->codec->extradata = av_mallocz(st->codec->extradata_size+FF_INPUT_BUFFER_PADDING_SIZE);
diff --git a/libavformat/mpc8.c b/libavformat/mpc8.c
index 6ede777b5f..9487b99496 100644
--- a/libavformat/mpc8.c
+++ b/libavformat/mpc8.c
@@ -182,7 +182,7 @@ static int mpc8_read_header(AVFormatContext *s, AVFormatParameters *ap)
         return AVERROR(ENOMEM);
     st->codec->codec_type = CODEC_TYPE_AUDIO;
     st->codec->codec_id = CODEC_ID_MUSEPACK8;
-    st->codec->bits_per_sample = 16;
+    st->codec->bits_per_coded_sample = 16;
 
     st->codec->extradata_size = 2;
     st->codec->extradata = av_mallocz(st->codec->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
diff --git a/libavformat/mpeg.c b/libavformat/mpeg.c
index 0d9b9fde7f..289be8caf6 100644
--- a/libavformat/mpeg.c
+++ b/libavformat/mpeg.c
@@ -520,13 +520,13 @@ static int mpegps_read_packet(AVFormatContext *s,
         freq = (b1 >> 4) & 3;
         st->codec->sample_rate = lpcm_freq_tab[freq];
         st->codec->channels = 1 + (b1 & 7);
-        st->codec->bits_per_sample = 16 + ((b1 >> 6) & 3) * 4;
+        st->codec->bits_per_coded_sample = 16 + ((b1 >> 6) & 3) * 4;
         st->codec->bit_rate = st->codec->channels *
                               st->codec->sample_rate *
-                              st->codec->bits_per_sample;
-        if (st->codec->bits_per_sample == 16)
+                              st->codec->bits_per_coded_sample;
+        if (st->codec->bits_per_coded_sample == 16)
             st->codec->codec_id = CODEC_ID_PCM_S16BE;
-        else if (st->codec->bits_per_sample == 28)
+        else if (st->codec->bits_per_coded_sample == 28)
             return AVERROR(EINVAL);
     }
     av_new_packet(pkt, len);
diff --git a/libavformat/mtv.c b/libavformat/mtv.c
index d97b9cdd85..75da0e76e3 100644
--- a/libavformat/mtv.c
+++ b/libavformat/mtv.c
@@ -102,7 +102,7 @@ static int mtv_read_header(AVFormatContext *s, AVFormatParameters *ap)
     st->codec->codec_tag       = MKTAG('R', 'G', 'B', mtv->img_bpp);
     st->codec->width           = mtv->img_width;
     st->codec->height          = mtv->img_height;
-    st->codec->bits_per_sample = mtv->img_bpp;
+    st->codec->bits_per_coded_sample = mtv->img_bpp;
     st->codec->sample_rate     = mtv->video_fps;
 
     /* audio - mp3 */
diff --git a/libavformat/mvi.c b/libavformat/mvi.c
index 03d4a5d6b5..2dc4667a68 100644
--- a/libavformat/mvi.c
+++ b/libavformat/mvi.c
@@ -80,7 +80,7 @@ static int read_header(AVFormatContext *s, AVFormatParameters *ap)
     ast->codec->codec_type      = CODEC_TYPE_AUDIO;
     ast->codec->codec_id        = CODEC_ID_PCM_U8;
     ast->codec->channels        = 1;
-    ast->codec->bits_per_sample = 8;
+    ast->codec->bits_per_coded_sample = 8;
     ast->codec->bit_rate        = ast->codec->sample_rate * 8;
 
     av_set_pts_info(vst, 64, msecs_per_frame, 1000000);
diff --git a/libavformat/mxfdec.c b/libavformat/mxfdec.c
index 2bf0cbc224..913848fa8f 100644
--- a/libavformat/mxfdec.c
+++ b/libavformat/mxfdec.c
@@ -222,7 +222,7 @@ static int mxf_get_d10_aes3_packet(ByteIOContext *pb, AVStream *st, AVPacket *pk
     for (; buf_ptr < end_ptr; ) {
         for (i = 0; i < st->codec->channels; i++) {
             uint32_t sample = bytestream_get_le32(&buf_ptr);
-            if (st->codec->bits_per_sample == 24)
+            if (st->codec->bits_per_coded_sample == 24)
                 bytestream_put_le24(&data_ptr, (sample >> 4) & 0xffffff);
             else
                 bytestream_put_le16(&data_ptr, (sample >> 12) & 0xffff);
@@ -788,7 +788,7 @@ static int mxf_parse_structural_metadata(MXFContext *mxf)
                 st->codec->codec_id = container_ul->id;
             st->codec->width = descriptor->width;
             st->codec->height = descriptor->height;
-            st->codec->bits_per_sample = descriptor->bits_per_sample; /* Uncompressed */
+            st->codec->bits_per_coded_sample = descriptor->bits_per_sample; /* Uncompressed */
             st->sample_aspect_ratio = descriptor->aspect_ratio;
             st->need_parsing = AVSTREAM_PARSE_HEADERS;
         } else if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
@@ -796,7 +796,7 @@ static int mxf_parse_structural_metadata(MXFContext *mxf)
             if (st->codec->codec_id == CODEC_ID_NONE)
                 st->codec->codec_id = container_ul->id;
             st->codec->channels = descriptor->channels;
-            st->codec->bits_per_sample = descriptor->bits_per_sample;
+            st->codec->bits_per_coded_sample = descriptor->bits_per_sample;
             st->codec->sample_rate = descriptor->sample_rate.num / descriptor->sample_rate.den;
             /* TODO: implement CODEC_ID_RAWAUDIO */
             if (st->codec->codec_id == CODEC_ID_PCM_S16LE) {
diff --git a/libavformat/mxfenc.c b/libavformat/mxfenc.c
index 1d36f44757..c649870aac 100644
--- a/libavformat/mxfenc.c
+++ b/libavformat/mxfenc.c
@@ -588,7 +588,7 @@ static void mxf_write_wav_desc(AVFormatContext *s, AVStream *st)
     put_be32(pb, st->codec->channels);
 
     mxf_write_local_tag(pb, 4, 0x3D01);
-    put_be32(pb, st->codec->bits_per_sample);
+    put_be32(pb, st->codec->bits_per_coded_sample);
 }
 
 static void mxf_write_package(AVFormatContext *s, enum MXFMetadataSetType type)
diff --git a/libavformat/nsvdec.c b/libavformat/nsvdec.c
index 9fdb1a8796..dbc2d00a24 100644
--- a/libavformat/nsvdec.c
+++ b/libavformat/nsvdec.c
@@ -456,7 +456,7 @@ static int nsv_parse_NSVs_header(AVFormatContext *s, AVFormatParameters *ap)
             st->codec->codec_id = codec_get_id(nsv_codec_video_tags, vtag);
             st->codec->width = vwidth;
             st->codec->height = vheight;
-            st->codec->bits_per_sample = 24; /* depth XXX */
+            st->codec->bits_per_coded_sample = 24; /* depth XXX */
 
             av_set_pts_info(st, 64, framerate.den, framerate.num);
             st->start_time = 0;
diff --git a/libavformat/nuv.c b/libavformat/nuv.c
index c3439c12bc..bd988398ff 100644
--- a/libavformat/nuv.c
+++ b/libavformat/nuv.c
@@ -93,11 +93,11 @@ static int get_codec_data(ByteIOContext *pb, AVStream *vst,
                 if (ast) {
                     ast->codec->codec_tag = get_le32(pb);
                     ast->codec->sample_rate = get_le32(pb);
-                    ast->codec->bits_per_sample = get_le32(pb);
+                    ast->codec->bits_per_coded_sample = get_le32(pb);
                     ast->codec->channels = get_le32(pb);
                     ast->codec->codec_id =
                         wav_codec_get_id(ast->codec->codec_tag,
-                                         ast->codec->bits_per_sample);
+                                         ast->codec->bits_per_coded_sample);
                     ast->need_parsing = AVSTREAM_PARSE_FULL;
                 } else
                     url_fskip(pb, 4 * 4);
@@ -157,7 +157,7 @@ static int nuv_header(AVFormatContext *s, AVFormatParameters *ap) {
         vst->codec->codec_id = CODEC_ID_NUV;
         vst->codec->width = width;
         vst->codec->height = height;
-        vst->codec->bits_per_sample = 10;
+        vst->codec->bits_per_coded_sample = 10;
         vst->sample_aspect_ratio = av_d2q(aspect * height / width, 10000);
         vst->r_frame_rate = av_d2q(fps, 60000);
         av_set_pts_info(vst, 32, 1, 1000);
@@ -175,7 +175,7 @@ static int nuv_header(AVFormatContext *s, AVFormatParameters *ap) {
         ast->codec->sample_rate = 44100;
         ast->codec->bit_rate = 2 * 2 * 44100 * 8;
         ast->codec->block_align = 2 * 2;
-        ast->codec->bits_per_sample = 16;
+        ast->codec->bits_per_coded_sample = 16;
         av_set_pts_info(ast, 32, 1, 1000);
     } else
         ctx->a_id = -1;
diff --git a/libavformat/riff.c b/libavformat/riff.c
index 95e6248193..0cab2650ce 100644
--- a/libavformat/riff.c
+++ b/libavformat/riff.c
@@ -251,8 +251,8 @@ int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
         if (!(bps = av_get_bits_per_sample(enc->codec_id)))
             bps = 16; // default to 16
     }
-    if(bps != enc->bits_per_sample && enc->bits_per_sample){
-        av_log(enc, AV_LOG_WARNING, "requested bits_per_sample (%d) and actually stored (%d) differ\n", enc->bits_per_sample, bps);
+    if(bps != enc->bits_per_coded_sample && enc->bits_per_coded_sample){
+        av_log(enc, AV_LOG_WARNING, "requested bits_per_coded_sample (%d) and actually stored (%d) differ\n", enc->bits_per_coded_sample, bps);
     }
 
     if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3 || enc->codec_id == CODEC_ID_AC3) {
@@ -327,7 +327,7 @@ void put_bmp_header(ByteIOContext *pb, AVCodecContext *enc, const AVCodecTag *ta
     put_le32(pb, enc->height);
     put_le16(pb, 1); /* planes */
 
-    put_le16(pb, enc->bits_per_sample ? enc->bits_per_sample : 24); /* depth */
+    put_le16(pb, enc->bits_per_coded_sample ? enc->bits_per_coded_sample : 24); /* depth */
     /* compression type */
     put_le32(pb, enc->codec_tag);
     put_le32(pb, enc->width * enc->height * 3);
@@ -363,15 +363,15 @@ void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size)
     codec->bit_rate = get_le32(pb) * 8;
     codec->block_align = get_le16(pb);
     if (size == 14) {  /* We're dealing with plain vanilla WAVEFORMAT */
-        codec->bits_per_sample = 8;
+        codec->bits_per_coded_sample = 8;
     }else
-        codec->bits_per_sample = get_le16(pb);
+        codec->bits_per_coded_sample = get_le16(pb);
     if (size >= 18) {  /* We're obviously dealing with WAVEFORMATEX */
         int cbSize = get_le16(pb); /* cbSize */
         size -= 18;
         cbSize = FFMIN(size, cbSize);
         if (cbSize >= 22 && id == 0xfffe) { /* WAVEFORMATEXTENSIBLE */
-            codec->bits_per_sample = get_le16(pb);
+            codec->bits_per_coded_sample = get_le16(pb);
             get_le32(pb); /* dwChannelMask */
             id = get_le32(pb); /* 4 first bytes of GUID */
             url_fskip(pb, 12); /* skip end of GUID */
@@ -389,7 +389,7 @@ void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size)
         if (size > 0)
             url_fskip(pb, size);
     }
-    codec->codec_id = wav_codec_get_id(id, codec->bits_per_sample);
+    codec->codec_id = wav_codec_get_id(id, codec->bits_per_coded_sample);
 }
 
 
diff --git a/libavformat/rl2.c b/libavformat/rl2.c
index 407867fb5b..5bab85b331 100644
--- a/libavformat/rl2.c
+++ b/libavformat/rl2.c
@@ -148,12 +148,12 @@ static av_cold int rl2_read_header(AVFormatContext *s,
         st->codec->codec_id = CODEC_ID_PCM_U8;
         st->codec->codec_tag = 1;
         st->codec->channels = channels;
-        st->codec->bits_per_sample = 8;
+        st->codec->bits_per_coded_sample = 8;
         st->codec->sample_rate = rate;
         st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-            st->codec->bits_per_sample;
+            st->codec->bits_per_coded_sample;
         st->codec->block_align = st->codec->channels *
-            st->codec->bits_per_sample / 8;
+            st->codec->bits_per_coded_sample / 8;
         av_set_pts_info(st,32,1,rate);
     }
 
diff --git a/libavformat/rpl.c b/libavformat/rpl.c
index 7340e18e63..dd5dedf817 100644
--- a/libavformat/rpl.c
+++ b/libavformat/rpl.c
@@ -142,7 +142,7 @@ static int rpl_read_header(AVFormatContext *s, AVFormatParameters *ap)
     vst->codec->codec_tag       = read_line_and_int(pb, &error);  // video format
     vst->codec->width           = read_line_and_int(pb, &error);  // video width
     vst->codec->height          = read_line_and_int(pb, &error);  // video height
-    vst->codec->bits_per_sample = read_line_and_int(pb, &error);  // video bits per sample
+    vst->codec->bits_per_coded_sample = read_line_and_int(pb, &error);  // video bits per sample
     error |= read_line(pb, line, sizeof(line));                   // video frames per second
     fps = read_fps(line, &error);
     av_set_pts_info(vst, 32, fps.den, fps.num);
@@ -157,7 +157,7 @@ static int rpl_read_header(AVFormatContext *s, AVFormatParameters *ap)
         case 124:
             vst->codec->codec_id = CODEC_ID_ESCAPE124;
             // The header is wrong here, at least sometimes
-            vst->codec->bits_per_sample = 16;
+            vst->codec->bits_per_coded_sample = 16;
             break;
 #if 0
         case 130:
@@ -184,20 +184,20 @@ static int rpl_read_header(AVFormatContext *s, AVFormatParameters *ap)
         ast->codec->codec_tag       = audio_format;
         ast->codec->sample_rate     = read_line_and_int(pb, &error);  // audio bitrate
         ast->codec->channels        = read_line_and_int(pb, &error);  // number of audio channels
-        ast->codec->bits_per_sample = read_line_and_int(pb, &error);  // audio bits per sample
+        ast->codec->bits_per_coded_sample = read_line_and_int(pb, &error);  // audio bits per sample
         // At least one sample uses 0 for ADPCM, which is really 4 bits
         // per sample.
-        if (ast->codec->bits_per_sample == 0)
-            ast->codec->bits_per_sample = 4;
+        if (ast->codec->bits_per_coded_sample == 0)
+            ast->codec->bits_per_coded_sample = 4;
 
         ast->codec->bit_rate = ast->codec->sample_rate *
-                               ast->codec->bits_per_sample *
+                               ast->codec->bits_per_coded_sample *
                                ast->codec->channels;
 
         ast->codec->codec_id = CODEC_ID_NONE;
         switch (audio_format) {
             case 1:
-                if (ast->codec->bits_per_sample == 16) {
+                if (ast->codec->bits_per_coded_sample == 16) {
                     // 16-bit audio is always signed
                     ast->codec->codec_id = CODEC_ID_PCM_S16LE;
                     break;
@@ -206,12 +206,12 @@ static int rpl_read_header(AVFormatContext *s, AVFormatParameters *ap)
                 // samples needed.
                 break;
             case 101:
-                if (ast->codec->bits_per_sample == 8) {
+                if (ast->codec->bits_per_coded_sample == 8) {
                     // The samples with this kind of audio that I have
                     // are all unsigned.
                     ast->codec->codec_id = CODEC_ID_PCM_U8;
                     break;
-                } else if (ast->codec->bits_per_sample == 4) {
+                } else if (ast->codec->bits_per_coded_sample == 4) {
                     ast->codec->codec_id = CODEC_ID_ADPCM_IMA_EA_SEAD;
                     break;
                 }
diff --git a/libavformat/segafilm.c b/libavformat/segafilm.c
index 4c6d138d6b..4bc46c2773 100644
--- a/libavformat/segafilm.c
+++ b/libavformat/segafilm.c
@@ -148,12 +148,12 @@ static int film_read_header(AVFormatContext *s,
         st->codec->codec_id = film->audio_type;
         st->codec->codec_tag = 1;
         st->codec->channels = film->audio_channels;
-        st->codec->bits_per_sample = film->audio_bits;
+        st->codec->bits_per_coded_sample = film->audio_bits;
         st->codec->sample_rate = film->audio_samplerate;
         st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-            st->codec->bits_per_sample;
+            st->codec->bits_per_coded_sample;
         st->codec->block_align = st->codec->channels *
-            st->codec->bits_per_sample / 8;
+            st->codec->bits_per_coded_sample / 8;
     }
 
     /* load the sample table */
diff --git a/libavformat/sierravmd.c b/libavformat/sierravmd.c
index 7b0e700718..20c4f58022 100644
--- a/libavformat/sierravmd.c
+++ b/libavformat/sierravmd.c
@@ -120,13 +120,13 @@ static int vmd_read_header(AVFormatContext *s,
         st->codec->sample_rate = vmd->sample_rate;
         st->codec->block_align = AV_RL16(&vmd->vmd_header[806]);
         if (st->codec->block_align & 0x8000) {
-            st->codec->bits_per_sample = 16;
+            st->codec->bits_per_coded_sample = 16;
             st->codec->block_align = -(st->codec->block_align - 0x10000);
         } else {
-            st->codec->bits_per_sample = 8;
+            st->codec->bits_per_coded_sample = 8;
         }
         st->codec->bit_rate = st->codec->sample_rate *
-            st->codec->bits_per_sample * st->codec->channels;
+            st->codec->bits_per_coded_sample * st->codec->channels;
 
         /* calculate pts */
         num = st->codec->block_align;
diff --git a/libavformat/siff.c b/libavformat/siff.c
index 30bd307c97..63fe80dadb 100644
--- a/libavformat/siff.c
+++ b/libavformat/siff.c
@@ -75,7 +75,7 @@ static int create_audio_stream(AVFormatContext *s, SIFFContext *c)
     ast->codec->codec_type      = CODEC_TYPE_AUDIO;
     ast->codec->codec_id        = CODEC_ID_PCM_U8;
     ast->codec->channels        = 1;
-    ast->codec->bits_per_sample = c->bits;
+    ast->codec->bits_per_coded_sample = c->bits;
     ast->codec->sample_rate     = c->rate;
     ast->codec->frame_size      = c->block_align;
     av_set_pts_info(ast, 16, 1, c->rate);
diff --git a/libavformat/smacker.c b/libavformat/smacker.c
index 8d0b7428ba..d11bddc145 100644
--- a/libavformat/smacker.c
+++ b/libavformat/smacker.c
@@ -179,11 +179,11 @@ static int smacker_read_header(AVFormatContext *s, AVFormatParameters *ap)
             ast[i]->codec->codec_tag = MKTAG('S', 'M', 'K', 'A');
             ast[i]->codec->channels = (smk->rates[i] & SMK_AUD_STEREO) ? 2 : 1;
             ast[i]->codec->sample_rate = smk->rates[i] & 0xFFFFFF;
-            ast[i]->codec->bits_per_sample = (smk->rates[i] & SMK_AUD_16BITS) ? 16 : 8;
-            if(ast[i]->codec->bits_per_sample == 16 && ast[i]->codec->codec_id == CODEC_ID_PCM_U8)
+            ast[i]->codec->bits_per_coded_sample = (smk->rates[i] & SMK_AUD_16BITS) ? 16 : 8;
+            if(ast[i]->codec->bits_per_coded_sample == 16 && ast[i]->codec->codec_id == CODEC_ID_PCM_U8)
                 ast[i]->codec->codec_id = CODEC_ID_PCM_S16LE;
             av_set_pts_info(ast[i], 64, 1, ast[i]->codec->sample_rate
-                    * ast[i]->codec->channels * ast[i]->codec->bits_per_sample / 8);
+                    * ast[i]->codec->channels * ast[i]->codec->bits_per_coded_sample / 8);
         }
     }
 
diff --git a/libavformat/tiertexseq.c b/libavformat/tiertexseq.c
index 00dcb5e232..7588ff037d 100644
--- a/libavformat/tiertexseq.c
+++ b/libavformat/tiertexseq.c
@@ -230,9 +230,9 @@ static int seq_read_header(AVFormatContext *s, AVFormatParameters *ap)
     st->codec->codec_tag = 0;  /* no tag */
     st->codec->channels = 1;
     st->codec->sample_rate = SEQ_SAMPLE_RATE;
-    st->codec->bits_per_sample = 16;
-    st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_sample * st->codec->channels;
-    st->codec->block_align = st->codec->channels * st->codec->bits_per_sample;
+    st->codec->bits_per_coded_sample = 16;
+    st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_coded_sample * st->codec->channels;
+    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
 
     return 0;
 }
diff --git a/libavformat/tta.c b/libavformat/tta.c
index 9a30e1fc87..884664c98d 100644
--- a/libavformat/tta.c
+++ b/libavformat/tta.c
@@ -91,7 +91,7 @@ static int tta_read_header(AVFormatContext *s, AVFormatParameters *ap)
     st->codec->codec_id = CODEC_ID_TTA;
     st->codec->channels = channels;
     st->codec->sample_rate = samplerate;
-    st->codec->bits_per_sample = bps;
+    st->codec->bits_per_coded_sample = bps;
 
     st->codec->extradata_size = url_ftell(s->pb);
     if(st->codec->extradata_size+FF_INPUT_BUFFER_PADDING_SIZE <= (unsigned)st->codec->extradata_size){
diff --git a/libavformat/utils.c b/libavformat/utils.c
index 87376485dd..e82324f323 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -2166,7 +2166,7 @@ int av_find_stream_info(AVFormatContext *ic)
     for(i=0;i<ic->nb_streams;i++) {
         st = ic->streams[i];
         if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
-            if(st->codec->codec_id == CODEC_ID_RAWVIDEO && !st->codec->codec_tag && !st->codec->bits_per_sample)
+            if(st->codec->codec_id == CODEC_ID_RAWVIDEO && !st->codec->codec_tag && !st->codec->bits_per_coded_sample)
                 st->codec->codec_tag= avcodec_pix_fmt_to_codec_tag(st->codec->pix_fmt);
 
             if(duration_count[i]
@@ -2198,8 +2198,8 @@ int av_find_stream_info(AVFormatContext *ic)
                 }
             }
         }else if(st->codec->codec_type == CODEC_TYPE_AUDIO) {
-            if(!st->codec->bits_per_sample)
-                st->codec->bits_per_sample= av_get_bits_per_sample(st->codec->codec_id);
+            if(!st->codec->bits_per_coded_sample)
+                st->codec->bits_per_coded_sample= av_get_bits_per_sample(st->codec->codec_id);
         }
     }
 
diff --git a/libavformat/vocdec.c b/libavformat/vocdec.c
index 178ecb5ac6..7aec2ab6df 100644
--- a/libavformat/vocdec.c
+++ b/libavformat/vocdec.c
@@ -84,7 +84,7 @@ voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
                 dec->sample_rate = sample_rate;
             dec->channels = channels;
             dec->codec_id = codec_get_id(ff_voc_codec_tags, get_byte(pb));
-            dec->bits_per_sample = av_get_bits_per_sample(dec->codec_id);
+            dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id);
             voc->remaining_size -= 2;
             max_size -= 2;
             channels = 1;
@@ -104,7 +104,7 @@ voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
 
         case VOC_TYPE_NEW_VOICE_DATA:
             dec->sample_rate = get_le32(pb);
-            dec->bits_per_sample = get_byte(pb);
+            dec->bits_per_coded_sample = get_byte(pb);
             dec->channels = get_byte(pb);
             dec->codec_id = codec_get_id(ff_voc_codec_tags, get_le16(pb));
             url_fskip(pb, 4);
@@ -120,7 +120,7 @@ voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
         }
     }
 
-    dec->bit_rate = dec->sample_rate * dec->bits_per_sample;
+    dec->bit_rate = dec->sample_rate * dec->bits_per_coded_sample;
 
     if (max_size <= 0)
         max_size = 2048;
diff --git a/libavformat/vocenc.c b/libavformat/vocenc.c
index 0802f50e62..4badb1d533 100644
--- a/libavformat/vocenc.c
+++ b/libavformat/vocenc.c
@@ -55,7 +55,7 @@ static int voc_write_packet(AVFormatContext *s, AVPacket *pkt)
             put_byte(pb, VOC_TYPE_NEW_VOICE_DATA);
             put_le24(pb, pkt->size + 12);
             put_le32(pb, enc->sample_rate);
-            put_byte(pb, enc->bits_per_sample);
+            put_byte(pb, enc->bits_per_coded_sample);
             put_byte(pb, enc->channels);
             put_le16(pb, enc->codec_tag);
             put_le32(pb, 0);
diff --git a/libavformat/wc3movie.c b/libavformat/wc3movie.c
index bb9a6e275d..df207ef9d9 100644
--- a/libavformat/wc3movie.c
+++ b/libavformat/wc3movie.c
@@ -259,10 +259,10 @@ static int wc3_read_header(AVFormatContext *s,
     st->codec->codec_id = CODEC_ID_PCM_S16LE;
     st->codec->codec_tag = 1;
     st->codec->channels = WC3_AUDIO_CHANNELS;
-    st->codec->bits_per_sample = WC3_AUDIO_BITS;
+    st->codec->bits_per_coded_sample = WC3_AUDIO_BITS;
     st->codec->sample_rate = WC3_SAMPLE_RATE;
     st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-        st->codec->bits_per_sample;
+        st->codec->bits_per_coded_sample;
     st->codec->block_align = WC3_AUDIO_BITS * WC3_AUDIO_CHANNELS;
 
     return 0;
diff --git a/libavformat/westwood.c b/libavformat/westwood.c
index d1ac63c762..600863e837 100644
--- a/libavformat/westwood.c
+++ b/libavformat/westwood.c
@@ -154,10 +154,10 @@ static int wsaud_read_header(AVFormatContext *s,
     st->codec->codec_tag = 0;  /* no tag */
     st->codec->channels = wsaud->audio_channels;
     st->codec->sample_rate = wsaud->audio_samplerate;
-    st->codec->bits_per_sample = wsaud->audio_bits;
+    st->codec->bits_per_coded_sample = wsaud->audio_bits;
     st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-        st->codec->bits_per_sample / 4;
-    st->codec->block_align = st->codec->channels * st->codec->bits_per_sample;
+        st->codec->bits_per_coded_sample / 4;
+    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
 
     wsaud->audio_stream_index = st->index;
     wsaud->audio_frame_counter = 0;
@@ -264,10 +264,10 @@ static int wsvqa_read_header(AVFormatContext *s,
         st->codec->channels = header[26];
         if (!st->codec->channels)
             st->codec->channels = 1;
-        st->codec->bits_per_sample = 16;
+        st->codec->bits_per_coded_sample = 16;
         st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-            st->codec->bits_per_sample / 4;
-        st->codec->block_align = st->codec->channels * st->codec->bits_per_sample;
+            st->codec->bits_per_coded_sample / 4;
+        st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
 
         wsvqa->audio_stream_index = st->index;
         wsvqa->audio_samplerate = st->codec->sample_rate;
diff --git a/libavformat/wv.c b/libavformat/wv.c
index eddd32148a..c9f71f95bd 100644
--- a/libavformat/wv.c
+++ b/libavformat/wv.c
@@ -151,7 +151,7 @@ static int wv_read_header(AVFormatContext *s,
     st->codec->codec_id = CODEC_ID_WAVPACK;
     st->codec->channels = wc->chan;
     st->codec->sample_rate = wc->rate;
-    st->codec->bits_per_sample = wc->bpp;
+    st->codec->bits_per_coded_sample = wc->bpp;
     av_set_pts_info(st, 64, 1, wc->rate);
     s->start_time = 0;
     s->duration = (int64_t)wc->samples * AV_TIME_BASE / st->codec->sample_rate;
diff --git a/libavformat/xa.c b/libavformat/xa.c
index c79df78a48..3ea2aaecc9 100644
--- a/libavformat/xa.c
+++ b/libavformat/xa.c
@@ -72,7 +72,7 @@ static int xa_read_header(AVFormatContext *s,
     /* Value in file is average byte rate*/
     st->codec->bit_rate     = get_le32(pb) * 8;
     st->codec->block_align  = get_le16(pb);
-    st->codec->bits_per_sample = get_le16(pb);
+    st->codec->bits_per_coded_sample = get_le16(pb);
 
     av_set_pts_info(st, 64, 1, st->codec->sample_rate);