[issue1569] flvenc.c audio_codec_ids error.

[Daniel Verkamp]

Daniel Verkamp <dan...@drv.nu> added the comment:

Attached patch fixes the issue in the FLV muxer and the equivalent 
problem in the demuxer.

Tested with the Windows Flash browser plugin - unsigned sounds correct, 
signed does not.

_____________________________________________________
FFmpeg issue tracker <iss...@roundup.ffmpeg.org>
<https://roundup.ffmpeg.org/roundup/ffmpeg/issue1569>
_____________________________________________________
>From dc1fc04a56b5c9baa770f481a6f26de2151002db Mon Sep 17 00:00:00 2001
From: Daniel Verkamp <dan...@drv.nu>
Date: Sun, 13 Dec 2009 04:45:57 -0500
Subject: [PATCH] FLV 8-bit PCM is unsigned, not signed

Fixes issue #1569
---
 libavformat/flvdec.c |    4 ++--
 libavformat/flvenc.c |    6 +++---
 2 files changed, 5 insertions(+), 5 deletions(-)

diff --git a/libavformat/flvdec.c b/libavformat/flvdec.c
index 2be5e2d..9c43149 100644
--- a/libavformat/flvdec.c
+++ b/libavformat/flvdec.c
@@ -50,7 +50,7 @@ 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_coded_sample == 8 ? 
CODEC_ID_PCM_S8 :
+            acodec->codec_id = acodec->bits_per_coded_sample == 8 ? 
CODEC_ID_PCM_U8 :
 #if HAVE_BIGENDIAN
                                 CODEC_ID_PCM_S16BE;
 #else
@@ -58,7 +58,7 @@ 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_coded_sample == 8 ? 
CODEC_ID_PCM_S8 : CODEC_ID_PCM_S16LE; break;
+            acodec->codec_id = acodec->bits_per_coded_sample == 8 ? 
CODEC_ID_PCM_U8 : 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:
diff --git a/libavformat/flvenc.c b/libavformat/flvenc.c
index ce8f565..c255ff0 100644
--- a/libavformat/flvenc.c
+++ b/libavformat/flvenc.c
@@ -38,7 +38,7 @@ static const AVCodecTag flv_video_codec_ids[] = {
 
 static const AVCodecTag flv_audio_codec_ids[] = {
     {CODEC_ID_MP3,       FLV_CODECID_MP3    >> FLV_AUDIO_CODECID_OFFSET},
-    {CODEC_ID_PCM_S8,    FLV_CODECID_PCM    >> FLV_AUDIO_CODECID_OFFSET},
+    {CODEC_ID_PCM_U8,    FLV_CODECID_PCM    >> FLV_AUDIO_CODECID_OFFSET},
     {CODEC_ID_PCM_S16BE, FLV_CODECID_PCM    >> FLV_AUDIO_CODECID_OFFSET},
     {CODEC_ID_PCM_S16LE, FLV_CODECID_PCM_LE >> FLV_AUDIO_CODECID_OFFSET},
     {CODEC_ID_ADPCM_SWF, FLV_CODECID_ADPCM  >> FLV_AUDIO_CODECID_OFFSET},
@@ -107,7 +107,7 @@ static int get_audio_flags(AVCodecContext *enc){
     case CODEC_ID_MP3:
         flags |= FLV_CODECID_MP3    | FLV_SAMPLESSIZE_16BIT;
         break;
-    case CODEC_ID_PCM_S8:
+    case CODEC_ID_PCM_U8:
         flags |= FLV_CODECID_PCM    | FLV_SAMPLESSIZE_8BIT;
         break;
     case CODEC_ID_PCM_S16BE:
@@ -248,7 +248,7 @@ static int flv_write_header(AVFormatContext *s)
         put_amf_double(pb, audio_enc->sample_rate);
 
         put_amf_string(pb, "audiosamplesize");
-        put_amf_double(pb, audio_enc->codec_id == CODEC_ID_PCM_S8 ? 8 : 16);
+        put_amf_double(pb, audio_enc->codec_id == CODEC_ID_PCM_U8 ? 8 : 16);
 
         put_amf_string(pb, "stereo");
         put_amf_bool(pb, audio_enc->channels == 2);
-- 
1.6.5.2