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authorAnton Khirnov <anton@khirnov.net>2016-02-10 14:17:21 +0100
committerAnton Khirnov <anton@khirnov.net>2016-02-24 10:08:34 +0100
commitac6d53589f3631ae08467c784fb371a15c957f01 (patch)
treedc8309172f5ec3d5622f7c9d547419c87f0d62b1 /doc
parenta9e1f2cc61cbd5606a087a60565e87923c39de5a (diff)
downloadffmpeg-ac6d53589f3631ae08467c784fb371a15c957f01.tar.gz
examples/transcode_aac: convert to codecpar
Diffstat (limited to 'doc')
-rw-r--r--doc/examples/transcode_aac.c66
1 files changed, 49 insertions, 17 deletions
diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c
index 3eebfb9d02..be86fe5d3b 100644
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@ -61,6 +61,7 @@ static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
{
+ AVCodecContext *avctx;
AVCodec *input_codec;
int error;
@@ -90,23 +91,39 @@ static int open_input_file(const char *filename,
}
/** Find a decoder for the audio stream. */
- if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
+ if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
fprintf(stderr, "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
+ /** allocate a new decoding context */
+ avctx = avcodec_alloc_context3(input_codec);
+ if (!avctx) {
+ fprintf(stderr, "Could not allocate a decoding context\n");
+ avformat_close_input(input_format_context);
+ return AVERROR(ENOMEM);
+ }
+
+ /** initialize the stream parameters with demuxer information */
+ error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
+ if (error < 0) {
+ avformat_close_input(input_format_context);
+ avcodec_free_context(&avctx);
+ return error;
+ }
+
/** Open the decoder for the audio stream to use it later. */
- if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
- input_codec, NULL)) < 0) {
+ if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
fprintf(stderr, "Could not open input codec (error '%s')\n",
get_error_text(error));
+ avcodec_free_context(&avctx);
avformat_close_input(input_format_context);
return error;
}
/** Save the decoder context for easier access later. */
- *input_codec_context = (*input_format_context)->streams[0]->codec;
+ *input_codec_context = avctx;
return 0;
}
@@ -121,6 +138,7 @@ static int open_output_file(const char *filename,
AVFormatContext **output_format_context,
AVCodecContext **output_codec_context)
{
+ AVCodecContext *avctx = NULL;
AVIOContext *output_io_context = NULL;
AVStream *stream = NULL;
AVCodec *output_codec = NULL;
@@ -160,27 +178,31 @@ static int open_output_file(const char *filename,
}
/** Create a new audio stream in the output file container. */
- if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
+ if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
- /** Save the encoder context for easier access later. */
- *output_codec_context = stream->codec;
+ avctx = avcodec_alloc_context3(output_codec);
+ if (!avctx) {
+ fprintf(stderr, "Could not allocate an encoding context\n");
+ error = AVERROR(ENOMEM);
+ goto cleanup;
+ }
/**
* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion.
*/
- (*output_codec_context)->channels = OUTPUT_CHANNELS;
- (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
- (*output_codec_context)->sample_rate = input_codec_context->sample_rate;
- (*output_codec_context)->sample_fmt = output_codec->sample_fmts[0];
- (*output_codec_context)->bit_rate = OUTPUT_BIT_RATE;
+ avctx->channels = OUTPUT_CHANNELS;
+ avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
+ avctx->sample_rate = input_codec_context->sample_rate;
+ avctx->sample_fmt = output_codec->sample_fmts[0];
+ avctx->bit_rate = OUTPUT_BIT_RATE;
/** Allow the use of the experimental AAC encoder */
- (*output_codec_context)->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
+ avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/** Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
@@ -191,18 +213,28 @@ static int open_output_file(const char *filename,
* Mark the encoder so that it behaves accordingly.
*/
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
- (*output_codec_context)->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+ avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/** Open the encoder for the audio stream to use it later. */
- if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
+ if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
get_error_text(error));
goto cleanup;
}
+ error = avcodec_parameters_from_context(stream->codecpar, avctx);
+ if (error < 0) {
+ fprintf(stderr, "Could not initialize stream parameters\n");
+ goto cleanup;
+ }
+
+ /** Save the encoder context for easier access later. */
+ *output_codec_context = avctx;
+
return 0;
cleanup:
+ avcodec_free_context(&avctx);
avio_close((*output_format_context)->pb);
avformat_free_context(*output_format_context);
*output_format_context = NULL;
@@ -773,13 +805,13 @@ cleanup:
avresample_free(&resample_context);
}
if (output_codec_context)
- avcodec_close(output_codec_context);
+ avcodec_free_context(&output_codec_context);
if (output_format_context) {
avio_close(output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
- avcodec_close(input_codec_context);
+ avcodec_free_context(&input_codec_context);
if (input_format_context)
avformat_close_input(&input_format_context);