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https://github.com/FFmpeg/FFmpeg.git
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g722enc: use AVCodec.encode2()
FATE reference updated due timestamp rounding because of resampling from 44100 Hz to 16000 Hz in avconv.
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@ -28,6 +28,7 @@
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*/
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#include "avcodec.h"
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#include "internal.h"
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#include "g722.h"
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#define FREEZE_INTERVAL 128
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@ -50,6 +51,9 @@ static av_cold int g722_encode_close(AVCodecContext *avctx)
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av_freep(&c->node_buf[i]);
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av_freep(&c->nodep_buf[i]);
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}
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#if FF_API_OLD_ENCODE_AUDIO
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av_freep(&avctx->coded_frame);
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#endif
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return 0;
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}
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@ -104,6 +108,7 @@ static av_cold int g722_encode_init(AVCodecContext * avctx)
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a common packet size for VoIP applications */
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avctx->frame_size = 320;
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}
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avctx->delay = 22;
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if (avctx->trellis) {
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/* validate trellis */
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@ -116,6 +121,14 @@ static av_cold int g722_encode_init(AVCodecContext * avctx)
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}
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}
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#if FF_API_OLD_ENCODE_AUDIO
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avctx->coded_frame = avcodec_alloc_frame();
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if (!avctx->coded_frame) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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#endif
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return 0;
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error:
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g722_encode_close(avctx);
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@ -345,27 +358,36 @@ static void g722_encode_no_trellis(G722Context *c,
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encode_byte(c, dst++, &samples[i]);
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}
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static int g722_encode_frame(AVCodecContext *avctx,
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uint8_t *dst, int buf_size, void *data)
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static int g722_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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G722Context *c = avctx->priv_data;
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const int16_t *samples = data;
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int nb_samples;
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const int16_t *samples = (const int16_t *)frame->data[0];
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int nb_samples, out_size, ret;
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nb_samples = avctx->frame_size - (avctx->frame_size & 1);
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if (avctx->trellis)
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g722_encode_trellis(c, avctx->trellis, dst, nb_samples, samples);
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else
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g722_encode_no_trellis(c, dst, nb_samples, samples);
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/* handle last frame with odd frame_size */
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if (nb_samples < avctx->frame_size) {
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int16_t last_samples[2] = { samples[nb_samples], samples[nb_samples] };
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encode_byte(c, &dst[nb_samples >> 1], last_samples);
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out_size = (frame->nb_samples + 1) / 2;
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if ((ret = ff_alloc_packet(avpkt, out_size))) {
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av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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return ret;
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}
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return (avctx->frame_size + 1) >> 1;
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nb_samples = frame->nb_samples - (frame->nb_samples & 1);
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if (avctx->trellis)
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g722_encode_trellis(c, avctx->trellis, avpkt->data, nb_samples, samples);
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else
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g722_encode_no_trellis(c, avpkt->data, nb_samples, samples);
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/* handle last frame with odd frame_size */
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if (nb_samples < frame->nb_samples) {
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int16_t last_samples[2] = { samples[nb_samples], samples[nb_samples] };
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encode_byte(c, &avpkt->data[nb_samples >> 1], last_samples);
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}
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if (frame->pts != AV_NOPTS_VALUE)
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avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
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*got_packet_ptr = 1;
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return 0;
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}
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AVCodec ff_adpcm_g722_encoder = {
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@ -375,7 +397,7 @@ AVCodec ff_adpcm_g722_encoder = {
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.priv_data_size = sizeof(G722Context),
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.init = g722_encode_init,
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.close = g722_encode_close,
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.encode = g722_encode_frame,
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.encode2 = g722_encode_frame,
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.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
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.long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
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.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
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@ -1,4 +1,4 @@
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7b0492eee76b04b710990235f97a0bf2 *./tests/data/acodec/g722.wav
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f30e8e99cfd3f38ba66f1d4131602a19 *./tests/data/acodec/g722.wav
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48053 ./tests/data/acodec/g722.wav
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b5568e0e3930ff563824156e8e1015f0 *./tests/data/g722.acodec.out.wav
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stddev: 8939.44 PSNR: 17.30 MAXDIFF:40370 bytes: 191980/ 1058400
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