[FFmpeg-devel] [PATCH] lavf/swf: support 8-bit audio.

[Clément Bœsch]

---
 libavformat/swfdec.c | 11 ++++++++---
 1 file changed, 8 insertions(+), 3 deletions(-)

diff --git a/libavformat/swfdec.c b/libavformat/swfdec.c
index 8fb4aeb..69e5f04 100644
--- a/libavformat/swfdec.c
+++ b/libavformat/swfdec.c
@@ -184,7 +184,7 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
             len -= 8;
         } else if (tag == TAG_STREAMHEAD || tag == TAG_STREAMHEAD2) {
             /* streaming found */
-            int sample_rate_code;
+            int sample_rate_code, is_16bits;
 
             for (i=0; i<s->nb_streams; i++) {
                 st = s->streams[i];
@@ -210,12 +210,15 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
             ast->codec->codec_id = ff_codec_get_id(swf_audio_codec_tags, (v>>4) & 15);
             ast->need_parsing = AVSTREAM_PARSE_FULL;
             sample_rate_code= (v>>2) & 3;
+            is_16bits = v>>1 & 1;
+            if (!is_16bits)
+                ast->codec->codec_id = AV_CODEC_ID_PCM_U8;
             ast->codec->sample_rate = 44100 >> (3 - sample_rate_code);
             avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate);
             len -= 4;
         } else if (tag == TAG_DEFINESOUND) {
             /* audio stream */
-            int sample_rate_code;
+            int sample_rate_code, is_16bits;
             int ch_id = avio_rl16(pb);
 
             for (i=0; i<s->nb_streams; i++) {
@@ -224,7 +227,6 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
                     goto skip;
             }
 
-            // FIXME: 8-bit uncompressed PCM audio will be interpreted as 16-bit
             // FIXME: The entire audio stream is stored in a single chunk/tag. Normally,
             // these are smaller audio streams in DEFINESOUND tags, but it's technically
             // possible they could be huge. Break it up into multiple packets if it's big.
@@ -238,6 +240,9 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
             ast->codec->codec_id = ff_codec_get_id(swf_audio_codec_tags, (v>>4) & 15);
             ast->need_parsing = AVSTREAM_PARSE_FULL;
             sample_rate_code= (v>>2) & 3;
+            is_16bits = v>>1 & 1;
+            if (!is_16bits)
+                ast->codec->codec_id = AV_CODEC_ID_PCM_U8;
             ast->codec->sample_rate = 44100 >> (3 - sample_rate_code);
             avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate);
             ast->duration = avio_rl32(pb); // number of samples
-- 
1.8.1.4