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https://github.com/FFmpeg/FFmpeg.git
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Add avcodec_decode_audio4().
Deprecate avcodec_decode_audio3(). Implement audio support in avcodec_default_get_buffer(). Implement the new audio decoder API in all audio decoders.
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@ -41,6 +41,7 @@
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#include "amrwbdata.h"
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typedef struct {
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AVFrame avframe; ///< AVFrame for decoded samples
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AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream
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enum Mode fr_cur_mode; ///< mode index of current frame
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uint8_t fr_quality; ///< frame quality index (FQI)
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@ -102,6 +103,9 @@ static av_cold int amrwb_decode_init(AVCodecContext *avctx)
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for (i = 0; i < 4; i++)
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ctx->prediction_error[i] = MIN_ENERGY;
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avcodec_get_frame_defaults(&ctx->avframe);
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avctx->coded_frame = &ctx->avframe;
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return 0;
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}
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@ -1062,15 +1066,15 @@ static void update_sub_state(AMRWBContext *ctx)
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LP_ORDER_16k * sizeof(float));
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}
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static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
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AVPacket *avpkt)
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static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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AMRWBContext *ctx = avctx->priv_data;
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AMRWBFrame *cf = &ctx->frame;
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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int expected_fr_size, header_size;
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float *buf_out = data;
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float *buf_out;
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float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing
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float fixed_gain_factor; // fixed gain correction factor (gamma)
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float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
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@ -1080,7 +1084,15 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
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float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band
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float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis
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float hb_gain;
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int sub, i;
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int sub, i, ret;
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/* get output buffer */
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ctx->avframe.nb_samples = 4 * AMRWB_SFR_SIZE_16k;
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if ((ret = avctx->get_buffer(avctx, &ctx->avframe)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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buf_out = (float *)ctx->avframe.data[0];
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header_size = decode_mime_header(ctx, buf);
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expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
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@ -1088,7 +1100,7 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
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if (buf_size < expected_fr_size) {
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av_log(avctx, AV_LOG_ERROR,
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"Frame too small (%d bytes). Truncated file?\n", buf_size);
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*data_size = 0;
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*got_frame_ptr = 0;
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return buf_size;
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}
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@ -1219,8 +1231,8 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
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memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
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memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
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/* report how many samples we got */
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*data_size = 4 * AMRWB_SFR_SIZE_16k * sizeof(float);
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*got_frame_ptr = 1;
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*(AVFrame *)data = ctx->avframe;
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return expected_fr_size;
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}
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@ -1232,6 +1244,7 @@ AVCodec ff_amrwb_decoder = {
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.priv_data_size = sizeof(AMRWBContext),
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.init = amrwb_decode_init,
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.decode = amrwb_decode_frame,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate WideBand"),
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.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
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};
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