lavc: remove disabled FF_API_CODEC_ID cruft

[ffmpeg] / libavcodec / mpegaudioenc.c

diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c
index 1cadef7175a71c7e699597f268a9ee0b0ebcbe68..76e7b88fce2d8bff3f3bf688afbff114fec3727d 100644 (file)
--- a/libavcodec/mpegaudioenc.c
+++ b/libavcodec/mpegaudioenc.c
@@ -24,6 +24,8 @@
  * The simplest mpeg audio layer 2 encoder.
  */
 
+#include "libavutil/channel_layout.h"
+
 #include "avcodec.h"
 #include "internal.h"
 #include "put_bits.h"
@@ -32,6 +34,7 @@
 #define WFRAC_BITS  14   /* fractional bits for window */
 
 #include "mpegaudio.h"
+#include "mpegaudiodsp.h"
 
 /* currently, cannot change these constants (need to modify
    quantization stage) */
@@ -75,36 +78,37 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
 
     if (channels <= 0 || channels > 2){
         av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
-        return -1;
+        return AVERROR(EINVAL);
     }
     bitrate = bitrate / 1000;
     s->nb_channels = channels;
     avctx->frame_size = MPA_FRAME_SIZE;
+    avctx->delay      = 512 - 32 + 1;
 
     /* encoding freq */
     s->lsf = 0;
     for(i=0;i<3;i++) {
-        if (ff_mpa_freq_tab[i] == freq)
+        if (avpriv_mpa_freq_tab[i] == freq)
             break;
-        if ((ff_mpa_freq_tab[i] / 2) == freq) {
+        if ((avpriv_mpa_freq_tab[i] / 2) == freq) {
             s->lsf = 1;
             break;
         }
     }
     if (i == 3){
         av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
-        return -1;
+        return AVERROR(EINVAL);
     }
     s->freq_index = i;
 
     /* encoding bitrate & frequency */
     for(i=0;i<15;i++) {
-        if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
+        if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
             break;
     }
     if (i == 15){
         av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
-        return -1;
+        return AVERROR(EINVAL);
     }
     s->bitrate_index = i;
 
@@ -180,9 +184,6 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
         total_quant_bits[i] = 12 * v;
     }
 
-    avctx->coded_frame= avcodec_alloc_frame();
-    avctx->coded_frame->key_frame= 1;
-
     return 0;
 }
 
@@ -725,14 +726,14 @@ static void encode_frame(MpegAudioContext *s,
     flush_put_bits(p);
 }
 
-static int MPA_encode_frame(AVCodecContext *avctx,
-                            unsigned char *frame, int buf_size, void *data)
+static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+                            const AVFrame *frame, int *got_packet_ptr)
 {
     MpegAudioContext *s = avctx->priv_data;
-    const short *samples = data;
+    const int16_t *samples = (const int16_t *)frame->data[0];
     short smr[MPA_MAX_CHANNELS][SBLIMIT];
     unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
-    int padding, i;
+    int padding, i, ret;
 
     for(i=0;i<s->nb_channels;i++) {
         filter(s, i, samples + i, s->nb_channels);
@@ -747,16 +748,20 @@ static int MPA_encode_frame(AVCodecContext *avctx,
     }
     compute_bit_allocation(s, smr, bit_alloc, &padding);
 
-    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
+    if ((ret = ff_alloc_packet(avpkt, MPA_MAX_CODED_FRAME_SIZE))) {
+        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+        return ret;
+    }
+
+    init_put_bits(&s->pb, avpkt->data, avpkt->size);
 
     encode_frame(s, bit_alloc, padding);
 
-    return put_bits_ptr(&s->pb) - s->pb.buf;
-}
+    if (frame->pts != AV_NOPTS_VALUE)
+        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
 
-static av_cold int MPA_encode_close(AVCodecContext *avctx)
-{
-    av_freep(&avctx->coded_frame);
+    avpkt->size = put_bits_count(&s->pb) / 8;
+    *got_packet_ptr = 1;
     return 0;
 }
 
@@ -766,15 +771,20 @@ static const AVCodecDefault mp2_defaults[] = {
 };
 
 AVCodec ff_mp2_encoder = {
-    .name           = "mp2",
-    .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = CODEC_ID_MP2,
-    .priv_data_size = sizeof(MpegAudioContext),
-    .init           = MPA_encode_init,
-    .encode         = MPA_encode_frame,
-    .close          = MPA_encode_close,
-    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
-    .supported_samplerates= (const int[]){44100, 48000,  32000, 22050, 24000, 16000, 0},
-    .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
-    .defaults       = mp2_defaults,
+    .name                  = "mp2",
+    .type                  = AVMEDIA_TYPE_AUDIO,
+    .id                    = AV_CODEC_ID_MP2,
+    .priv_data_size        = sizeof(MpegAudioContext),
+    .init                  = MPA_encode_init,
+    .encode2               = MPA_encode_frame,
+    .sample_fmts           = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+                                                            AV_SAMPLE_FMT_NONE },
+    .supported_samplerates = (const int[]){
+        44100, 48000,  32000, 22050, 24000, 16000, 0
+    },
+    .channel_layouts       = (const uint64_t[]){ AV_CH_LAYOUT_MONO,
+                                                 AV_CH_LAYOUT_STEREO,
+                                                 0 },
+    .long_name             = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
+    .defaults              = mp2_defaults,
 };