CODEC_ID_MP3LAME is obsolete

Originally committed as revision 2232 to svn://svn.ffmpeg.org/ffmpeg/trunk

diff --git a/libavcodec/mp3lameaudio.c b/libavcodec/mp3lameaudio.c
index 4e1660d..0a121eb 100644 (file)
--- a/libavcodec/mp3lameaudio.c
+++ b/libavcodec/mp3lameaudio.c
@@ -99,7 +99,7 @@ int MP3lame_encode_close(AVCodecContext *avctx)
 AVCodec mp3lame_encoder = {
     "mp3",
     CODEC_TYPE_AUDIO,
-    CODEC_ID_MP3LAME,
+    CODEC_ID_MP3,
     sizeof(Mp3AudioContext),
     MP3lame_encode_init,
     MP3lame_encode_frame,

diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c
index 61fc5b8..43ec55d 100644 (file)
--- a/libavcodec/mpegaudiodec.c
+++ b/libavcodec/mpegaudiodec.c
@@ -2597,7 +2597,7 @@ AVCodec mp3_decoder =
 {
     "mp3",
     CODEC_TYPE_AUDIO,
-    CODEC_ID_MP3LAME,
+    CODEC_ID_MP3,
     sizeof(MPADecodeContext),
     decode_init,
     NULL,

diff --git a/libavformat/asf.c b/libavformat/asf.c
index c05f28e..0a22fad 100644 (file)
--- a/libavformat/asf.c
+++ b/libavformat/asf.c
@@ -870,7 +870,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
                        asf_st->ds_span = 0; // disable descrambling
                }
                 switch (st->codec.codec_id) {
-                case CODEC_ID_MP3LAME:
+                case CODEC_ID_MP3:
                     st->codec.frame_size = MPA_FRAME_SIZE;
                     break;
                 case CODEC_ID_PCM_S16LE:
@@ -1252,7 +1252,7 @@ static AVOutputFormat asf_oformat = {
     "asf,wmv",
     sizeof(ASFContext),
 #ifdef CONFIG_MP3LAME
-    CODEC_ID_MP3LAME,
+    CODEC_ID_MP3,
 #else
     CODEC_ID_MP2,
 #endif
@@ -1269,7 +1269,7 @@ static AVOutputFormat asf_stream_oformat = {
     "asf,wmv",
     sizeof(ASFContext),
 #ifdef CONFIG_MP3LAME
-    CODEC_ID_MP3LAME,
+    CODEC_ID_MP3,
 #else
     CODEC_ID_MP2,
 #endif

diff --git a/libavformat/avienc.c b/libavformat/avienc.c
index 103860b..f5ecfa8 100644 (file)
--- a/libavformat/avienc.c
+++ b/libavformat/avienc.c
@@ -240,7 +240,7 @@ static void parse_specific_params(AVCodecContext *stream, int *au_byterate, int
         *au_ssize = 1;
         *au_scale = 1;
         *au_byterate = stream->bit_rate / 8;
-    case CODEC_ID_MP3LAME:
+    case CODEC_ID_MP3:
         *au_ssize = 1;
         *au_scale = 1;
         *au_byterate = stream->bit_rate / 8;    
@@ -570,7 +570,7 @@ static int avi_write_idx1(AVFormatContext *s)
                     if (nb_frames < stream->frame_number)
                         nb_frames = stream->frame_number;
                 } else {
-                    if (stream->codec_id == CODEC_ID_MP2 || stream->codec_id == CODEC_ID_MP3LAME) {
+                    if (stream->codec_id == CODEC_ID_MP2 || stream->codec_id == CODEC_ID_MP3) {
                         put_le32(pb, stream->frame_number);
                         nb_frames += stream->frame_number;
                     } else {
@@ -675,7 +675,7 @@ static int avi_write_trailer(AVFormatContext *s)
                  if (nb_frames < stream->frame_number)
                      nb_frames = stream->frame_number;
              } else {
-                 if (stream->codec_id == CODEC_ID_MP2 || stream->codec_id == CODEC_ID_MP3LAME) {
+                 if (stream->codec_id == CODEC_ID_MP2 || stream->codec_id == CODEC_ID_MP3) {
                      nb_frames += stream->frame_number;
                 }
             }

diff --git a/libavformat/flvdec.c b/libavformat/flvdec.c
index b15e143..e443fd5 100644 (file)
--- a/libavformat/flvdec.c
+++ b/libavformat/flvdec.c
@@ -64,7 +64,7 @@ static int flv_read_header(AVFormatContext *s,
         if (!st)
             return AVERROR_NOMEM;
         st->codec.codec_type = CODEC_TYPE_AUDIO;
-        st->codec.codec_id = CODEC_ID_MP3LAME;
+        st->codec.codec_id = CODEC_ID_MP3;
     }
 
     offset = get_be32(&s->pb);

diff --git a/libavformat/flvenc.c b/libavformat/flvenc.c
index 72abb65..65bdf4b 100644 (file)
--- a/libavformat/flvenc.c
+++ b/libavformat/flvenc.c
@@ -268,7 +268,7 @@ static int flv_write_packet(AVFormatContext *s, int stream_index,
     }
     else if (enc->codec_type == CODEC_TYPE_AUDIO) {
 #ifdef CONFIG_MP3LAME
-        if (enc->codec_id == CODEC_ID_MP3LAME ) {
+        if (enc->codec_id == CODEC_ID_MP3 ) {
             int c=0;
             for (;c<size;c++) {
                 flv->audioFifo[(flv->audioOutPos+c)%AUDIO_FIFO_SIZE] = buf[c];
@@ -366,7 +366,7 @@ static AVOutputFormat flv_oformat = {
     "flv",
     sizeof(FLVContext),
 #ifdef CONFIG_MP3LAME
-    CODEC_ID_MP3LAME,
+    CODEC_ID_MP3,
 #else // CONFIG_MP3LAME
     CODEC_ID_NONE,
 #endif // CONFIG_MP3LAME

diff --git a/libavformat/nut.c b/libavformat/nut.c
index 92f478e..2b52d5b 100644 (file)
--- a/libavformat/nut.c
+++ b/libavformat/nut.c
@@ -567,7 +567,7 @@ static AVOutputFormat nut_oformat = {
 #ifdef CONFIG_VORBIS
     CODEC_ID_VORBIS,
 #elif defined(CONFIG_MP3LAME)
-    CODEC_ID_MP3LAME,
+    CODEC_ID_MP3,
 #else
     CODEC_ID_MP2, /* AC3 needs liba52 decoder */
 #endif

diff --git a/libavformat/raw.c b/libavformat/raw.c
index 2354ad1..141d327 100644 (file)
--- a/libavformat/raw.c
+++ b/libavformat/raw.c
@@ -129,7 +129,7 @@ static int mp3_read_header(AVFormatContext *s,
            break;
            
     if( pos < s->pb.buffer_size-1 && (s->pb.buffer[pos+1] & 6) == 2 )
-       st->codec.codec_id = CODEC_ID_MP3LAME;
+       st->codec.codec_id = CODEC_ID_MP3;
 
     /* the parameters will be extracted from the compressed bitstream */
     return 0;
@@ -259,7 +259,7 @@ AVOutputFormat mp3_oformat = {
     "audio/x-mpeg",
     "mp3",
     0,
-    CODEC_ID_MP3LAME,
+    CODEC_ID_MP3,
     0,
     raw_write_header,
     raw_write_packet,

diff --git a/libavformat/rtp.c b/libavformat/rtp.c
index d9f99f1..46e1fd8 100644 (file)
--- a/libavformat/rtp.c
+++ b/libavformat/rtp.c
@@ -170,7 +170,7 @@ int rtp_get_payload_type(AVCodecContext *codec)
         }
         break;
     case CODEC_ID_MP2:
-    case CODEC_ID_MP3LAME:
+    case CODEC_ID_MP3:
         payload_type = RTP_PT_MPEGAUDIO;
         break;
     case CODEC_ID_MJPEG:
@@ -390,7 +390,7 @@ static int rtp_write_header(AVFormatContext *s1)
 
     switch(st->codec.codec_id) {
     case CODEC_ID_MP2:
-    case CODEC_ID_MP3LAME:
+    case CODEC_ID_MP3:
         s->buf_ptr = s->buf + 4;
         s->cur_timestamp = 0;
         break;
@@ -650,7 +650,7 @@ static int rtp_write_packet(AVFormatContext *s1, int stream_index,
         rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
         break;
     case CODEC_ID_MP2:
-    case CODEC_ID_MP3LAME:
+    case CODEC_ID_MP3:
         rtp_send_mpegaudio(s1, buf1, size);
         break;
     case CODEC_ID_MPEG1VIDEO:

diff --git a/libavformat/wav.c b/libavformat/wav.c
index 5f7c5dd..f71dd55 100644 (file)
--- a/libavformat/wav.c
+++ b/libavformat/wav.c
@@ -21,7 +21,7 @@
 
 const CodecTag codec_wav_tags[] = {
     { CODEC_ID_MP2, 0x50 },
-    { CODEC_ID_MP3LAME, 0x55 },
+    { CODEC_ID_MP3, 0x55 },
     { CODEC_ID_AC3, 0x2000 },
     { CODEC_ID_PCM_S16LE, 0x01 },
     { CODEC_ID_PCM_U8, 0x01 }, /* must come after s16le in this list */
@@ -55,7 +55,7 @@ int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
         enc->codec_id == CODEC_ID_PCM_ALAW ||
         enc->codec_id == CODEC_ID_PCM_MULAW) {
         bps = 8;
-    } else if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3LAME) {
+    } else if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3) {
         bps = 0;
     } else if (enc->codec_id == CODEC_ID_ADPCM_IMA_WAV || enc->codec_id == CODEC_ID_ADPCM_MS) {
         bps = 4;
@@ -63,7 +63,7 @@ int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
         bps = 16;
     }
     
-    if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3LAME) {
+    if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3) {
         blkalign = 1;
         //blkalign = 144 * enc->bit_rate/enc->sample_rate;
     } else if (enc->block_align != 0) { /* specified by the codec */
@@ -79,7 +79,7 @@ int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
     put_le32(pb, bytespersec); /* bytes per second */
     put_le16(pb, blkalign); /* block align */
     put_le16(pb, bps); /* bits per sample */
-    if (enc->codec_id == CODEC_ID_MP3LAME) {
+    if (enc->codec_id == CODEC_ID_MP3) {
         put_le16(pb, 12); /* wav_extra_size */
         hdrsize += 12;
         put_le16(pb, 1); /* wID */