diff mbox series

[FFmpeg-devel,2/2] avformat/mvdec: handle audio sample size

Message ID 4824174c-eab2-91e9-0fd1-f83ea67a91c0@personalprojects.net
State New
Headers show
Series Untitled series #5337
Related show

Commit Message

John-Paul Stewart Nov. 26, 2021, 9:52 p.m. UTC
Adds support for reading audio sample size from the data instead of
assuming all audio is 16 bits per sample.

Signed-off-by: John-Paul Stewart <jpstewart@personalprojects.net>
---
 libavformat/mvdec.c | 18 +++++++++++++++---
 1 file changed, 15 insertions(+), 3 deletions(-)

         }
@@ -369,7 +381,7 @@ static int mv_read_header(AVFormatContext *avctx)
             avio_skip(pb, 8);
             av_add_index_entry(ast, pos, timestamp, asize, 0, AVINDEX_KEYFRAME);
             av_add_index_entry(vst, pos + asize, i, vsize, 0, AVINDEX_KEYFRAME);
-            timestamp += asize / (ast->codecpar->channels * 2LL);
+            timestamp += asize / (ast->codecpar->channels * bytes_per_sample);
         }
     } else if (!version && avio_rb16(pb) == 3) {
         avio_skip(pb, 4);

Comments

Michael Niedermayer Nov. 27, 2021, 3:35 p.m. UTC | #1
On Fri, Nov 26, 2021 at 04:52:14PM -0500, John-Paul Stewart wrote:
> Adds support for reading audio sample size from the data instead of
> assuming all audio is 16 bits per sample.
> 
> Signed-off-by: John-Paul Stewart <jpstewart@personalprojects.net>
> ---
>  libavformat/mvdec.c | 18 +++++++++++++++---
>  1 file changed, 15 insertions(+), 3 deletions(-)
> 
> diff --git a/libavformat/mvdec.c b/libavformat/mvdec.c
> index 8492928820..8b54a9ab04 100644
> --- a/libavformat/mvdec.c
> +++ b/libavformat/mvdec.c
> @@ -299,6 +299,8 @@ static int mv_read_header(AVFormatContext *avctx)
>      if (version == 2) {
>          uint64_t timestamp;
>          int v;
> +        uint32_t bytes_per_sample;
> +
>          avio_skip(pb, 22);
> 
>          /* allocate audio track first to prevent unnecessary seeking
> @@ -341,11 +343,21 @@ static int mv_read_header(AVFormatContext *avctx)
>          }
>          avpriv_set_pts_info(ast, 33, 1, ast->codecpar->sample_rate);
> 
> -        avio_skip(pb, 4);
> +        bytes_per_sample = avio_rb32(pb);
> 
>          v = avio_rb32(pb);
>          if (v == AUDIO_FORMAT_SIGNED) {
> -            ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE;
> +            switch (bytes_per_sample) {
> +            case 1:
> +                ast->codecpar->codec_id = AV_CODEC_ID_PCM_S8;
> +                break;
> +            case 2:
> +                ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE;
> +                break;
> +            default:
> +                avpriv_request_sample(avctx, "Audio sample size %i bytes", bytes_per_sample);
> +                break;
> +            }
>          } else {
>              avpriv_request_sample(avctx, "Audio compression (format
> %i)", v);
>          }

your MUA added a linebreak here

Applying: avformat/mvdec: handle audio sample size
error: corrupt patch at line 41


[...]
John-Paul Stewart Nov. 27, 2021, 9:39 p.m. UTC | #2
On 2021-11-27 10:35 a.m., Michael Niedermayer wrote:
> On Fri, Nov 26, 2021 at 04:52:14PM -0500, John-Paul Stewart wrote:
>> Adds support for reading audio sample size from the data instead of
>> assuming all audio is 16 bits per sample.
>>
>> Signed-off-by: John-Paul Stewart <jpstewart@personalprojects.net>
>> ---
>>  libavformat/mvdec.c | 18 +++++++++++++++---
>>  1 file changed, 15 insertions(+), 3 deletions(-)
>>
>> diff --git a/libavformat/mvdec.c b/libavformat/mvdec.c
>> index 8492928820..8b54a9ab04 100644
>> --- a/libavformat/mvdec.c
>> +++ b/libavformat/mvdec.c
>> @@ -299,6 +299,8 @@ static int mv_read_header(AVFormatContext *avctx)
>>      if (version == 2) {
>>          uint64_t timestamp;
>>          int v;
>> +        uint32_t bytes_per_sample;
>> +
>>          avio_skip(pb, 22);
>>
>>          /* allocate audio track first to prevent unnecessary seeking
>> @@ -341,11 +343,21 @@ static int mv_read_header(AVFormatContext *avctx)
>>          }
>>          avpriv_set_pts_info(ast, 33, 1, ast->codecpar->sample_rate);
>>
>> -        avio_skip(pb, 4);
>> +        bytes_per_sample = avio_rb32(pb);
>>
>>          v = avio_rb32(pb);
>>          if (v == AUDIO_FORMAT_SIGNED) {
>> -            ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE;
>> +            switch (bytes_per_sample) {
>> +            case 1:
>> +                ast->codecpar->codec_id = AV_CODEC_ID_PCM_S8;
>> +                break;
>> +            case 2:
>> +                ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE;
>> +                break;
>> +            default:
>> +                avpriv_request_sample(avctx, "Audio sample size %i bytes", bytes_per_sample);
>> +                break;
>> +            }
>>          } else {
>>              avpriv_request_sample(avctx, "Audio compression (format
>> %i)", v);
>>          }
> 
> your MUA added a linebreak here
> 
> Applying: avformat/mvdec: handle audio sample size
> error: corrupt patch at line 41

Oops!  Sorry about that.  I thought I'd turned off automatic
line-wrapping in the MUA.

I'll try to re-send the patch directly from 'git send-email' instead.
diff mbox series

Patch

diff --git a/libavformat/mvdec.c b/libavformat/mvdec.c
index 8492928820..8b54a9ab04 100644
--- a/libavformat/mvdec.c
+++ b/libavformat/mvdec.c
@@ -299,6 +299,8 @@  static int mv_read_header(AVFormatContext *avctx)
     if (version == 2) {
         uint64_t timestamp;
         int v;
+        uint32_t bytes_per_sample;
+
         avio_skip(pb, 22);

         /* allocate audio track first to prevent unnecessary seeking
@@ -341,11 +343,21 @@  static int mv_read_header(AVFormatContext *avctx)
         }
         avpriv_set_pts_info(ast, 33, 1, ast->codecpar->sample_rate);

-        avio_skip(pb, 4);
+        bytes_per_sample = avio_rb32(pb);

         v = avio_rb32(pb);
         if (v == AUDIO_FORMAT_SIGNED) {
-            ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE;
+            switch (bytes_per_sample) {
+            case 1:
+                ast->codecpar->codec_id = AV_CODEC_ID_PCM_S8;
+                break;
+            case 2:
+                ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE;
+                break;
+            default:
+                avpriv_request_sample(avctx, "Audio sample size %i bytes", bytes_per_sample);
+                break;
+            }
         } else {
             avpriv_request_sample(avctx, "Audio compression (format
%i)", v);