normalize calls to ff_alloc_packet2

- check ret < 0
- remove excessive error log
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

libavcodec/aacenc.c

@@ -570,10 +570,8 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         }
         start_ch += chans;
     }
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels))) {
-        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)
         return ret;
-    }
     do {
         int frame_bits;
 

libavcodec/ac3enc_template.c

@@ -435,7 +435,7 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt,
 
     ff_ac3_quantize_mantissas(s);
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size)) < 0)
         return ret;
     ff_ac3_output_frame(s, avpkt->data);
 

libavcodec/adpcmenc.c

@@ -494,7 +494,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
     else
         pkt_size = avctx->block_align;
-    if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size)) < 0)
         return ret;
     dst = avpkt->data;
 

libavcodec/alacenc.c

@@ -613,7 +613,7 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     else
         max_frame_size = s->max_coded_frame_size;
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size)) < 0)
         return ret;
 
     /* use verbatim mode for compression_level 0 */

libavcodec/dcaenc.c

@@ -497,7 +497,7 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     const int16_t *samples;
     int ret, real_channel = 0;
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)) < 0)
         return ret;
 
     samples = (const int16_t *)frame->data[0];

libavcodec/flacenc.c

@@ -1276,7 +1276,7 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         }
     }
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, frame_bytes)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, frame_bytes)) < 0)
         return ret;
 
     out_bytes = write_frame(s, avpkt);

libavcodec/g722enc.c

@@ -368,7 +368,7 @@ static int g722_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     int nb_samples, out_size, ret;
 
     out_size = (frame->nb_samples + 1) / 2;
-    if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)) < 0)
         return ret;
 
     nb_samples = frame->nb_samples - (frame->nb_samples & 1);

libavcodec/g723_1.c

@@ -2458,7 +2458,7 @@ static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         offset += LPC_ORDER;
     }
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 24)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 24)) < 0)
         return ret;
 
     *got_packet_ptr = 1;

libavcodec/g726.c

@@ -362,7 +362,7 @@ static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     int i, ret, out_size;
 
     out_size = (frame->nb_samples * c->code_size + 7) / 8;
-    if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)) < 0)
         return ret;
     init_put_bits(&pb, avpkt->data, avpkt->size);
 

libavcodec/libaacplus.c

@@ -104,7 +104,7 @@ static int aacPlus_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
     int32_t *input_buffer = (int32_t *)frame->data[0];
     int ret;
 
-    if ((ret = ff_alloc_packet2(avctx, pkt, s->max_output_bytes)))
+    if ((ret = ff_alloc_packet2(avctx, pkt, s->max_output_bytes)) < 0)
         return ret;
 
     pkt->size = aacplusEncEncode(s->aacplus_handle, input_buffer,

libavcodec/libfaac.c

@@ -184,10 +184,8 @@ static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     int num_samples  = frame ? frame->nb_samples : 0;
     void *samples    = frame ? frame->data[0]    : NULL;
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, (7 + 768) * avctx->channels))) {
-        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+    if ((ret = ff_alloc_packet2(avctx, avpkt, (7 + 768) * avctx->channels)) < 0)
         return ret;
-    }
 
     bytes_written = faacEncEncode(s->faac_handle, samples,
                                   num_samples * avctx->channels,

libavcodec/libfdk-aacenc.c

@@ -339,7 +339,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     }
 
     /* The maximum packet size is 6144 bits aka 768 bytes per channel. */
-    if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))) < 0)
         return ret;
 
     out_ptr                   = avpkt->data;

libavcodec/libgsm.c

@@ -107,7 +107,7 @@ static int libgsm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     gsm_signal *samples = (gsm_signal *)frame->data[0];
     struct gsm_state *state = avctx->priv_data;
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, avctx->block_align)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, avctx->block_align)) < 0)
         return ret;
 
     switch(avctx->codec_id) {

libavcodec/libmp3lame.c

@@ -254,7 +254,7 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
             s->buffer_index);
     if (len <= s->buffer_index) {
-        if ((ret = ff_alloc_packet2(avctx, avpkt, len)))
+        if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0)
             return ret;
         memcpy(avpkt->data, s->buffer, len);
         s->buffer_index -= len;

libavcodec/libopencore-amr.c

@@ -247,7 +247,7 @@ static int amr_nb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         s->enc_bitrate = avctx->bit_rate;
     }
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 32)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 32)) < 0)
         return ret;
 
     if (frame) {

libavcodec/libspeexenc.c

@@ -304,7 +304,7 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     /* write output if all frames for the packet have been encoded */
     if (s->pkt_frame_count == s->frames_per_packet) {
         s->pkt_frame_count = 0;
-        if ((ret = ff_alloc_packet2(avctx, avpkt, speex_bits_nbytes(&s->bits))))
+        if ((ret = ff_alloc_packet2(avctx, avpkt, speex_bits_nbytes(&s->bits))) < 0)
             return ret;
         ret = speex_bits_write(&s->bits, avpkt->data, avpkt->size);
         speex_bits_reset(&s->bits);

libavcodec/libtwolame.c

@@ -94,7 +94,7 @@ static int twolame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     TWOLAMEContext *s = avctx->priv_data;
     int ret;
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0)
         return ret;
 
     if (frame) {

libavcodec/libvo-aacenc.c

@@ -161,7 +161,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             return ret;
     }
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))) < 0)
         return ret;
 
     input.Buffer  = samples;

libavcodec/libvo-amrwbenc.c

@@ -121,7 +121,7 @@ static int amr_wb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     const int16_t *samples = (const int16_t *)frame->data[0];
     int size, ret;
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, MAX_PACKET_SIZE)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, MAX_PACKET_SIZE)) < 0)
         return ret;
 
     if (s->last_bitrate != avctx->bit_rate) {

libavcodec/libvorbisenc.c

@@ -348,7 +348,7 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 
     av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes)) < 0)
         return ret;
     av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
 

libavcodec/mpegaudioenc.c

@@ -754,7 +754,7 @@ static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     }
     compute_bit_allocation(s, smr, bit_alloc, &padding);
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0)
         return ret;
 
     init_put_bits(&s->pb, avpkt->data, avpkt->size);

libavcodec/nellymoserenc.c

@@ -401,7 +401,7 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         s->last_frame = 1;
     }
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, NELLY_BLOCK_LEN)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, NELLY_BLOCK_LEN)) < 0)
         return ret;
     encode_block(s, avpkt->data, avpkt->size);
 

libavcodec/pcm.c

@@ -107,7 +107,7 @@ static int pcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     n           = frame->nb_samples * avctx->channels;
     samples     = (const short *)frame->data[0];
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, n * sample_size)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, n * sample_size)) < 0)
         return ret;
     dst = avpkt->data;
 

libavcodec/ra144enc.c

@@ -458,7 +458,7 @@ static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     if (ractx->last_frame)
         return 0;
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, FRAMESIZE)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, FRAMESIZE)) < 0)
         return ret;
 
     /**

libavcodec/roqaudioenc.c

@@ -173,7 +173,7 @@ static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     else
         data_size = avctx->channels * avctx->frame_size;
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size)) < 0)
         return ret;
     out = avpkt->data;
 

libavcodec/sonic.c

@@ -632,7 +632,7 @@ static int sonic_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     int ret;
     const short *samples = (const int16_t*)frame->data[0];
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000)) < 0)
         return ret;
 
     init_put_bits(&pb, avpkt->data, avpkt->size);

libavcodec/vorbisenc.c

@@ -1028,7 +1028,7 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         return 0;
     samples = 1 << (venc->log2_blocksize[0] - 1);
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192)) < 0)
         return ret;
 
     init_put_bits(&pb, avpkt->data, avpkt->size);

libavcodec/wmaenc.c

@@ -366,7 +366,7 @@ static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
         }
     }
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE)) < 0)
         return ret;
 
     total_gain= 128;