avcodec: Add bits_per_raw_sample to AVCodecParameters

The bits_per_raw_sample represents the number of bits of precision per sample.

The field is added at the logical place, not at the end as the code was just
recently added

This fixes the regression about losing the audio sample precision information

The change in the fate test checksum un-does the change from the merge

Previous version reviewed by: wm4 <nfxjfg@googlemail.com>
Previous version reviewed by: Dominik 'Rathann' Mierzejewski <dominik@greysector.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

1 files changed, 24 insertions, 0 deletions

diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index b3655c5..9e6169f 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -3832,9 +3832,33 @@ typedef struct AVCodecParameters {
      */
     int64_t bit_rate;
 
+    /**
+     * The number of bits per sample in the codedwords.
+     *
+     * This is basically the bitrate per sample. It is mandatory for a bunch of
+     * formats to actually decode them. It's the number of bits for one sample in
+     * the actual coded bitstream.
+     *
+     * This could be for example 4 for ADPCM
+     * For PCM formats this matches bits_per_raw_sample
+     * Can be 0
+     */
     int bits_per_coded_sample;
 
     /**
+     * This is the number of valid bits in each output sample. If the
+     * sample format has more bits, the least significant bits are additional
+     * padding bits, which are always 0. Use right shifts to reduce the sample
+     * to its actual size. For example, audio formats with 24 bit samples will
+     * have bits_per_raw_sample set to 24, and format set to AV_SAMPLEFMT_S32.
+     * To get the original sample use "(uint32_t)sample >> 8"."
+     *
+     * For ADPCM this might be 12 or 16 or similar
+     * Can be 0
+     */
+    int bits_per_raw_sample;
+
+    /**
      * Codec-specific bitstream restrictions that the stream conforms to.
      */
     int profile;