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https://gitee.com/openharmony/third_party_ffmpeg
synced 2024-11-27 05:00:37 +00:00
normalize calls to ff_alloc_packet2
- check ret < 0 - remove excessive error log Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
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c257fe1fde
commit
bcaf64b605
@ -570,10 +570,8 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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}
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start_ch += chans;
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}
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if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels))) {
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av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)
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return ret;
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}
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do {
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int frame_bits;
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@ -435,7 +435,7 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt,
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ff_ac3_quantize_mantissas(s);
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if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size)) < 0)
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return ret;
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ff_ac3_output_frame(s, avpkt->data);
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@ -494,7 +494,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
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else
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pkt_size = avctx->block_align;
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if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size)) < 0)
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return ret;
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dst = avpkt->data;
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@ -613,7 +613,7 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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else
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max_frame_size = s->max_coded_frame_size;
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if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size)) < 0)
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return ret;
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/* use verbatim mode for compression_level 0 */
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@ -497,7 +497,7 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const int16_t *samples;
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int ret, real_channel = 0;
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if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)) < 0)
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return ret;
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samples = (const int16_t *)frame->data[0];
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@ -1276,7 +1276,7 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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}
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}
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if ((ret = ff_alloc_packet2(avctx, avpkt, frame_bytes)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, frame_bytes)) < 0)
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return ret;
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out_bytes = write_frame(s, avpkt);
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@ -368,7 +368,7 @@ static int g722_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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int nb_samples, out_size, ret;
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out_size = (frame->nb_samples + 1) / 2;
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if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)) < 0)
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return ret;
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nb_samples = frame->nb_samples - (frame->nb_samples & 1);
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@ -2458,7 +2458,7 @@ static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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offset += LPC_ORDER;
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}
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if ((ret = ff_alloc_packet2(avctx, avpkt, 24)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, 24)) < 0)
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return ret;
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*got_packet_ptr = 1;
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@ -362,7 +362,7 @@ static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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int i, ret, out_size;
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out_size = (frame->nb_samples * c->code_size + 7) / 8;
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if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)) < 0)
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return ret;
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init_put_bits(&pb, avpkt->data, avpkt->size);
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@ -104,7 +104,7 @@ static int aacPlus_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
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int32_t *input_buffer = (int32_t *)frame->data[0];
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int ret;
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if ((ret = ff_alloc_packet2(avctx, pkt, s->max_output_bytes)))
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if ((ret = ff_alloc_packet2(avctx, pkt, s->max_output_bytes)) < 0)
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return ret;
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pkt->size = aacplusEncEncode(s->aacplus_handle, input_buffer,
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@ -184,10 +184,8 @@ static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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int num_samples = frame ? frame->nb_samples : 0;
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void *samples = frame ? frame->data[0] : NULL;
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if ((ret = ff_alloc_packet2(avctx, avpkt, (7 + 768) * avctx->channels))) {
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av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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if ((ret = ff_alloc_packet2(avctx, avpkt, (7 + 768) * avctx->channels)) < 0)
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return ret;
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}
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bytes_written = faacEncEncode(s->faac_handle, samples,
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num_samples * avctx->channels,
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@ -339,7 +339,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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}
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/* The maximum packet size is 6144 bits aka 768 bytes per channel. */
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if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))))
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if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))) < 0)
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return ret;
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out_ptr = avpkt->data;
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@ -107,7 +107,7 @@ static int libgsm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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gsm_signal *samples = (gsm_signal *)frame->data[0];
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struct gsm_state *state = avctx->priv_data;
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if ((ret = ff_alloc_packet2(avctx, avpkt, avctx->block_align)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, avctx->block_align)) < 0)
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return ret;
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switch(avctx->codec_id) {
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@ -254,7 +254,7 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
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s->buffer_index);
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if (len <= s->buffer_index) {
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if ((ret = ff_alloc_packet2(avctx, avpkt, len)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0)
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return ret;
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memcpy(avpkt->data, s->buffer, len);
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s->buffer_index -= len;
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@ -247,7 +247,7 @@ static int amr_nb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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s->enc_bitrate = avctx->bit_rate;
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}
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if ((ret = ff_alloc_packet2(avctx, avpkt, 32)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, 32)) < 0)
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return ret;
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if (frame) {
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@ -304,7 +304,7 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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/* write output if all frames for the packet have been encoded */
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if (s->pkt_frame_count == s->frames_per_packet) {
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s->pkt_frame_count = 0;
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if ((ret = ff_alloc_packet2(avctx, avpkt, speex_bits_nbytes(&s->bits))))
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if ((ret = ff_alloc_packet2(avctx, avpkt, speex_bits_nbytes(&s->bits))) < 0)
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return ret;
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ret = speex_bits_write(&s->bits, avpkt->data, avpkt->size);
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speex_bits_reset(&s->bits);
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@ -94,7 +94,7 @@ static int twolame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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TWOLAMEContext *s = avctx->priv_data;
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int ret;
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if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0)
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return ret;
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if (frame) {
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@ -161,7 +161,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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return ret;
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}
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if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))))
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if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))) < 0)
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return ret;
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input.Buffer = samples;
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@ -121,7 +121,7 @@ static int amr_wb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const int16_t *samples = (const int16_t *)frame->data[0];
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int size, ret;
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if ((ret = ff_alloc_packet2(avctx, avpkt, MAX_PACKET_SIZE)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, MAX_PACKET_SIZE)) < 0)
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return ret;
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if (s->last_bitrate != avctx->bit_rate) {
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@ -348,7 +348,7 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
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if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes)) < 0)
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return ret;
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av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
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@ -754,7 +754,7 @@ static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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}
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compute_bit_allocation(s, smr, bit_alloc, &padding);
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if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0)
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return ret;
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init_put_bits(&s->pb, avpkt->data, avpkt->size);
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@ -401,7 +401,7 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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s->last_frame = 1;
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}
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if ((ret = ff_alloc_packet2(avctx, avpkt, NELLY_BLOCK_LEN)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, NELLY_BLOCK_LEN)) < 0)
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return ret;
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encode_block(s, avpkt->data, avpkt->size);
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@ -107,7 +107,7 @@ static int pcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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n = frame->nb_samples * avctx->channels;
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samples = (const short *)frame->data[0];
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if ((ret = ff_alloc_packet2(avctx, avpkt, n * sample_size)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, n * sample_size)) < 0)
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return ret;
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dst = avpkt->data;
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@ -458,7 +458,7 @@ static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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if (ractx->last_frame)
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return 0;
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if ((ret = ff_alloc_packet2(avctx, avpkt, FRAMESIZE)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, FRAMESIZE)) < 0)
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return ret;
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/**
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@ -173,7 +173,7 @@ static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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else
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data_size = avctx->channels * avctx->frame_size;
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if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size)) < 0)
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return ret;
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out = avpkt->data;
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@ -632,7 +632,7 @@ static int sonic_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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int ret;
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const short *samples = (const int16_t*)frame->data[0];
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if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000)) < 0)
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return ret;
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init_put_bits(&pb, avpkt->data, avpkt->size);
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@ -1028,7 +1028,7 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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return 0;
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samples = 1 << (venc->log2_blocksize[0] - 1);
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if ((ret = ff_alloc_packet2(avctx, avpkt, 8192)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, 8192)) < 0)
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return ret;
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init_put_bits(&pb, avpkt->data, avpkt->size);
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@ -366,7 +366,7 @@ static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
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}
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}
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if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE)))
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if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE)) < 0)
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return ret;
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total_gain= 128;
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