mirror of
https://github.com/xenia-project/FFmpeg.git
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b945fed629
Explicitly identify decoder/encoder wrappers with a common name. This
saves API users from guessing by the name suffix. For example, they
don't have to guess that "h264_qsv" is the h264 QSV implementation, and
instead they can just check the AVCodec .codec and .wrapper_name fields.
Explicitly mark AVCodec entries that are hardware decoders or most
likely hardware decoders with new AV_CODEC_CAPs. The purpose is allowing
API users listing hardware decoders in a more generic way. The proposed
AVCodecHWConfig does not provide this information fully, because it's
concerned with decoder configuration, not information about the fact
whether the hardware is used or not.
AV_CODEC_CAP_HYBRID exists specifically for QSV, which can have software
implementations in case the hardware is not capable.
Based on a patch by Philip Langdale <philipl@overt.org>.
Merges Libav commit 47687a2f8a
.
196 lines
5.6 KiB
C
196 lines
5.6 KiB
C
/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <wavpack/wavpack.h>
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#include <string.h>
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#include "libavutil/attributes.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "audio_frame_queue.h"
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#include "avcodec.h"
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#include "internal.h"
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#define WV_DEFAULT_BLOCK_SIZE 32768
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typedef struct LibWavpackContext {
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const AVClass *class;
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WavpackContext *wv;
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AudioFrameQueue afq;
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AVPacket *pkt;
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int user_size;
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int got_output;
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} LibWavpackContext;
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static int wavpack_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
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const AVFrame *frame, int *got_output)
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{
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LibWavpackContext *s = avctx->priv_data;
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int ret;
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s->got_output = 0;
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s->pkt = pkt;
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s->user_size = pkt->size;
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if (frame) {
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ret = ff_af_queue_add(&s->afq, frame);
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if (ret < 0)
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return ret;
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ret = WavpackPackSamples(s->wv, (int32_t*)frame->data[0], frame->nb_samples);
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if (!ret) {
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av_log(avctx, AV_LOG_ERROR, "Error encoding a frame: %s\n",
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WavpackGetErrorMessage(s->wv));
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return AVERROR_UNKNOWN;
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}
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}
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if (!s->got_output &&
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(!frame || frame->nb_samples < avctx->frame_size)) {
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ret = WavpackFlushSamples(s->wv);
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if (!ret) {
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av_log(avctx, AV_LOG_ERROR, "Error flushing the encoder: %s\n",
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WavpackGetErrorMessage(s->wv));
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return AVERROR_UNKNOWN;
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}
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}
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if (s->got_output) {
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ff_af_queue_remove(&s->afq, avctx->frame_size, &pkt->pts, &pkt->duration);
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*got_output = 1;
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}
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return 0;
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}
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static int encode_callback(void *id, void *data, int32_t count)
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{
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AVCodecContext *avctx = id;
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LibWavpackContext *s = avctx->priv_data;
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int ret, offset = s->pkt->size;
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if (s->user_size) {
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if (s->user_size - count < s->pkt->size) {
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av_log(avctx, AV_LOG_ERROR, "Provided packet too small.\n");
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return 0;
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}
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s->pkt->size += count;
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} else {
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ret = av_grow_packet(s->pkt, count);
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if (ret < 0) {
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av_log(avctx, AV_LOG_ERROR, "Error allocating output packet.\n");
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return 0;
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}
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}
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memcpy(s->pkt->data + offset, data, count);
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s->got_output = 1;
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return 1;
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}
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static av_cold int wavpack_encode_init(AVCodecContext *avctx)
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{
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LibWavpackContext *s = avctx->priv_data;
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WavpackConfig config = { 0 };
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int ret;
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s->wv = WavpackOpenFileOutput(encode_callback, avctx, NULL);
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if (!s->wv) {
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av_log(avctx, AV_LOG_ERROR, "Error allocating the encoder.\n");
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return AVERROR(ENOMEM);
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}
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if (!avctx->frame_size)
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avctx->frame_size = WV_DEFAULT_BLOCK_SIZE;
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config.bytes_per_sample = 4;
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config.bits_per_sample = 32;
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config.block_samples = avctx->frame_size;
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config.channel_mask = avctx->channel_layout;
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config.num_channels = avctx->channels;
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config.sample_rate = avctx->sample_rate;
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if (avctx->compression_level != FF_COMPRESSION_DEFAULT) {
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if (avctx->compression_level >= 3) {
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config.flags |= CONFIG_VERY_HIGH_FLAG;
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if (avctx->compression_level >= 8)
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config.xmode = 6;
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else if (avctx->compression_level >= 7)
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config.xmode = 5;
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else if (avctx->compression_level >= 6)
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config.xmode = 4;
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else if (avctx->compression_level >= 5)
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config.xmode = 3;
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else if (avctx->compression_level >= 4)
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config.xmode = 2;
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} else if (avctx->compression_level >= 2)
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config.flags |= CONFIG_HIGH_FLAG;
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else if (avctx->compression_level < 1)
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config.flags |= CONFIG_FAST_FLAG;
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}
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ret = WavpackSetConfiguration(s->wv, &config, -1);
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if (!ret)
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goto fail;
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ret = WavpackPackInit(s->wv);
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if (!ret)
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goto fail;
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ff_af_queue_init(avctx, &s->afq);
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return 0;
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fail:
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av_log(avctx, AV_LOG_ERROR, "Error configuring the encoder: %s.\n",
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WavpackGetErrorMessage(s->wv));
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WavpackCloseFile(s->wv);
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return AVERROR_UNKNOWN;
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}
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static av_cold int wavpack_encode_close(AVCodecContext *avctx)
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{
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LibWavpackContext *s = avctx->priv_data;
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WavpackCloseFile(s->wv);
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ff_af_queue_close(&s->afq);
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return 0;
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}
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AVCodec ff_libwavpack_encoder = {
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.name = "libwavpack",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_WAVPACK,
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.priv_data_size = sizeof(LibWavpackContext),
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.init = wavpack_encode_init,
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.encode2 = wavpack_encode_frame,
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.close = wavpack_encode_close,
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.capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
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AV_SAMPLE_FMT_NONE },
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.wrapper_name = "libwavpack",
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};
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