[FFmpeg-cvslog] 32bit libmp3lame support.

[Peter Belkner]

ffmpeg | branch: master | Peter Belkner <pbelkner at snafu.de> | Tue Apr 19 12:00:01 2011 +0200| [39e69d0b88bf521097feeaa2ec5f390dd1e27723] | committer: Michael Niedermayer

32bit libmp3lame support.

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=39e69d0b88bf521097feeaa2ec5f390dd1e27723
---

 libavcodec/libmp3lame.c |   66 ++++++++++++++++++++++++++++++++++++++++++----
 1 files changed, 60 insertions(+), 6 deletions(-)

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 086c7dc..5cfb122 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -34,6 +34,10 @@ typedef struct Mp3AudioContext {
     int stereo;
     uint8_t buffer[BUFFER_SIZE];
     int buffer_index;
+    struct {
+        int *left;
+        int *right;
+    } s32_data;
 } Mp3AudioContext;
 
 static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
@@ -76,6 +80,19 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
     }
     avctx->coded_frame->key_frame= 1;
 
+    if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt && s->stereo) {
+        int nelem = 2 * avctx->frame_size;
+
+        if(! (s->s32_data.left = av_malloc(nelem * sizeof(int)))) {
+            av_freep(&avctx->coded_frame);
+            lame_close(s->gfp);
+
+            return AVERROR(ENOMEM);
+        }
+
+        s->s32_data.right = s->s32_data.left + avctx->frame_size;
+    }
+
     return 0;
 
 err_close:
@@ -150,17 +167,35 @@ static int MP3lame_encode_frame(AVCodecContext *avctx,
 
     /* lame 3.91 dies on '1-channel interleaved' data */
 
-    if(data){
+    if(!data){
+        lame_result= lame_encode_flush(
+                s->gfp,
+                s->buffer + s->buffer_index,
+                BUFFER_SIZE - s->buffer_index
+                );
+#if 2147483647 == INT_MAX
+    }else if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt){
         if (s->stereo) {
-            lame_result = lame_encode_buffer_interleaved(
+            int32_t *rp = data;
+            int32_t *mp = rp + 2*avctx->frame_size;
+            int *wpl = s->s32_data.left;
+            int *wpr = s->s32_data.right;
+
+            while (rp < mp) {
+                *wpl++ = *rp++;
+                *wpr++ = *rp++;
+            }
+
+            lame_result = lame_encode_buffer_int(
                 s->gfp,
-                data,
+                s->s32_data.left,
+                s->s32_data.right,
                 avctx->frame_size,
                 s->buffer + s->buffer_index,
                 BUFFER_SIZE - s->buffer_index
                 );
         } else {
-            lame_result = lame_encode_buffer(
+            lame_result = lame_encode_buffer_int(
                 s->gfp,
                 data,
                 data,
@@ -169,12 +204,26 @@ static int MP3lame_encode_frame(AVCodecContext *avctx,
                 BUFFER_SIZE - s->buffer_index
                 );
         }
+#endif
     }else{
-        lame_result= lame_encode_flush(
+        if (s->stereo) {
+            lame_result = lame_encode_buffer_interleaved(
                 s->gfp,
+                data,
+                avctx->frame_size,
                 s->buffer + s->buffer_index,
                 BUFFER_SIZE - s->buffer_index
                 );
+        } else {
+            lame_result = lame_encode_buffer(
+                s->gfp,
+                data,
+                data,
+                avctx->frame_size,
+                s->buffer + s->buffer_index,
+                BUFFER_SIZE - s->buffer_index
+                );
+        }
     }
 
     if(lame_result < 0){
@@ -210,6 +259,7 @@ static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
 {
     Mp3AudioContext *s = avctx->priv_data;
 
+    av_freep(&s->s32_data.left);
     av_freep(&avctx->coded_frame);
 
     lame_close(s->gfp);
@@ -226,7 +276,11 @@ AVCodec ff_libmp3lame_encoder = {
     MP3lame_encode_frame,
     MP3lame_encode_close,
     .capabilities= CODEC_CAP_DELAY,
-    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
+#if 2147483647 == INT_MAX
+    AV_SAMPLE_FMT_S32,
+#endif
+    AV_SAMPLE_FMT_NONE},
     .supported_samplerates= sSampleRates,
     .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
 };