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https://github.com/xenia-project/FFmpeg.git
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Merge commit '0946c754d99c05413e813ee515039adcf0f9232a'
* commit '0946c754d99c05413e813ee515039adcf0f9232a': examples/decode_audio: use a parser for splitting the input Merged-by: Clément Bœsch <cboesch@gopro.com>
This commit is contained in:
commit
87e16e2b44
@ -39,14 +39,46 @@
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#define AUDIO_INBUF_SIZE 20480
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#define AUDIO_INBUF_SIZE 20480
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#define AUDIO_REFILL_THRESH 4096
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#define AUDIO_REFILL_THRESH 4096
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static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
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FILE *outfile)
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{
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int len, got_frame;
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while (pkt->size > 0) {
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len = avcodec_decode_audio4(dec_ctx, frame, &got_frame, pkt);
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if (len < 0) {
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fprintf(stderr, "Error while decoding\n");
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exit(1);
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}
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if (got_frame) {
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int i, ch;
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/* if a frame has been decoded, output it */
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int data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
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if (data_size < 0) {
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/* This should not occur, checking just for paranoia */
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fprintf(stderr, "Failed to calculate data size\n");
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exit(1);
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}
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for (i = 0; i < frame->nb_samples; i++)
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for (ch = 0; ch < dec_ctx->channels; ch++)
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fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
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}
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pkt->size -= len;
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pkt->data += len;
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}
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}
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int main(int argc, char **argv)
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int main(int argc, char **argv)
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{
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{
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const char *outfilename, *filename;
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const char *outfilename, *filename;
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const AVCodec *codec;
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const AVCodec *codec;
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AVCodecContext *c= NULL;
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AVCodecContext *c= NULL;
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int len;
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AVCodecParserContext *parser = NULL;
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int len, ret;
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FILE *f, *outfile;
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FILE *f, *outfile;
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uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
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uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
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uint8_t *data;
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size_t data_size;
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AVPacket avpkt;
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AVPacket avpkt;
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AVFrame *decoded_frame = NULL;
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AVFrame *decoded_frame = NULL;
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@ -69,6 +101,12 @@ int main(int argc, char **argv)
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exit(1);
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exit(1);
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}
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}
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parser = av_parser_init(codec->id);
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if (!parser) {
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fprintf(stderr, "Parser not found\n");
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exit(1);
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}
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c = avcodec_alloc_context3(codec);
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c = avcodec_alloc_context3(codec);
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if (!c) {
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if (!c) {
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fprintf(stderr, "Could not allocate audio codec context\n");
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fprintf(stderr, "Could not allocate audio codec context\n");
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@ -93,12 +131,10 @@ int main(int argc, char **argv)
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}
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}
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/* decode until eof */
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/* decode until eof */
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avpkt.data = inbuf;
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data = inbuf;
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avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
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data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
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while (avpkt.size > 0) {
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int got_frame = 0;
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while (data_size > 0) {
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if (!decoded_frame) {
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if (!decoded_frame) {
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if (!(decoded_frame = av_frame_alloc())) {
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if (!(decoded_frame = av_frame_alloc())) {
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fprintf(stderr, "Could not allocate audio frame\n");
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fprintf(stderr, "Could not allocate audio frame\n");
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@ -106,39 +142,26 @@ int main(int argc, char **argv)
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}
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}
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}
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}
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len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
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ret = av_parser_parse2(parser, c, &avpkt.data, &avpkt.size,
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if (len < 0) {
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data, data_size,
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fprintf(stderr, "Error while decoding\n");
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AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
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if (ret < 0) {
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fprintf(stderr, "Error while parsing\n");
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exit(1);
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exit(1);
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}
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}
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if (got_frame) {
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data += ret;
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int i, ch;
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data_size -= ret;
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/* if a frame has been decoded, output it */
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int data_size = av_get_bytes_per_sample(c->sample_fmt);
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if (avpkt.size)
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if (data_size < 0) {
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decode(c, &avpkt, decoded_frame, outfile);
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/* This should not occur, checking just for paranoia */
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fprintf(stderr, "Failed to calculate data size\n");
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if (data_size < AUDIO_REFILL_THRESH) {
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exit(1);
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memmove(inbuf, data, data_size);
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}
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data = inbuf;
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for (i=0; i<decoded_frame->nb_samples; i++)
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len = fread(data + data_size, 1,
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for (ch=0; ch<c->channels; ch++)
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AUDIO_INBUF_SIZE - data_size, f);
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fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
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}
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avpkt.size -= len;
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avpkt.data += len;
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avpkt.dts =
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avpkt.pts = AV_NOPTS_VALUE;
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if (avpkt.size < AUDIO_REFILL_THRESH) {
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/* Refill the input buffer, to avoid trying to decode
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* incomplete frames. Instead of this, one could also use
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* a parser, or use a proper container format through
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* libavformat. */
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memmove(inbuf, avpkt.data, avpkt.size);
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avpkt.data = inbuf;
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len = fread(avpkt.data + avpkt.size, 1,
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AUDIO_INBUF_SIZE - avpkt.size, f);
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if (len > 0)
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if (len > 0)
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avpkt.size += len;
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data_size += len;
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}
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}
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}
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}
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@ -146,6 +169,7 @@ int main(int argc, char **argv)
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fclose(f);
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fclose(f);
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avcodec_free_context(&c);
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avcodec_free_context(&c);
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av_parser_close(parser);
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av_frame_free(&decoded_frame);
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av_frame_free(&decoded_frame);
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return 0;
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return 0;
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