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Diffstat (limited to 'doc/examples/transcode_aac.c')
-rw-r--r--doc/examples/transcode_aac.c144
1 files changed, 60 insertions, 84 deletions
diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c
index 44d5af6b04..9fd5c00d60 100644
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@ -1,20 +1,20 @@
/*
* Copyright (c) 2013-2017 Andreas Unterweger
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -23,7 +23,7 @@
* Simple audio converter
*
* @example transcode_aac.c
- * Convert an input audio file to AAC in an MP4 container using Libav.
+ * Convert an input audio file to AAC in an MP4 container using FFmpeg.
* Formats other than MP4 are supported based on the output file extension.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
@@ -36,11 +36,12 @@
#include "libavcodec/avcodec.h"
#include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
-#include "libavresample/avresample.h"
+#include "libswresample/swresample.h"
/* The output bit rate in bit/s */
#define OUTPUT_BIT_RATE 96000
@@ -48,18 +49,6 @@
#define OUTPUT_CHANNELS 2
/**
- * Convert an error code into a text message.
- * @param error Error code to be converted
- * @return Corresponding error text (not thread-safe)
- */
-static char *get_error_text(const int error)
-{
- static char error_buffer[255];
- av_strerror(error, error_buffer, sizeof(error_buffer));
- return error_buffer;
-}
-
-/**
* Open an input file and the required decoder.
* @param filename File to be opened
* @param[out] input_format_context Format context of opened file
@@ -78,7 +67,7 @@ static int open_input_file(const char *filename,
if ((error = avformat_open_input(input_format_context, filename, NULL,
NULL)) < 0) {
fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
- filename, get_error_text(error));
+ filename, av_err2str(error));
*input_format_context = NULL;
return error;
}
@@ -86,7 +75,7 @@ static int open_input_file(const char *filename,
/* Get information on the input file (number of streams etc.). */
if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
fprintf(stderr, "Could not open find stream info (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
avformat_close_input(input_format_context);
return error;
}
@@ -125,7 +114,7 @@ static int open_input_file(const char *filename,
/* Open the decoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
fprintf(stderr, "Could not open input codec (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
avcodec_free_context(&avctx);
avformat_close_input(input_format_context);
return error;
@@ -162,7 +151,7 @@ static int open_output_file(const char *filename,
if ((error = avio_open(&output_io_context, filename,
AVIO_FLAG_WRITE)) < 0) {
fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
- filename, get_error_text(error));
+ filename, av_err2str(error));
return error;
}
@@ -228,7 +217,7 @@ static int open_output_file(const char *filename,
/* Open the encoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
goto cleanup;
}
@@ -245,7 +234,7 @@ static int open_output_file(const char *filename,
cleanup:
avcodec_free_context(&avctx);
- avio_close((*output_format_context)->pb);
+ avio_closep(&(*output_format_context)->pb);
avformat_free_context(*output_format_context);
*output_format_context = NULL;
return error < 0 ? error : AVERROR_EXIT;
@@ -280,7 +269,7 @@ static int init_input_frame(AVFrame **frame)
/**
* Initialize the audio resampler based on the input and output codec settings.
* If the input and output sample formats differ, a conversion is required
- * libavresample takes care of this, but requires initialization.
+ * libswresample takes care of this, but requires initialization.
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
* @param[out] resample_context Resample context for the required conversion
@@ -288,45 +277,42 @@ static int init_input_frame(AVFrame **frame)
*/
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
- AVAudioResampleContext **resample_context)
+ SwrContext **resample_context)
{
- /* Only initialize the resampler if it is necessary, i.e.,
- * if and only if the sample formats differ. */
- if (input_codec_context->sample_fmt != output_codec_context->sample_fmt ||
- input_codec_context->channels != output_codec_context->channels) {
int error;
- /* Create a resampler context for the conversion. */
- if (!(*resample_context = avresample_alloc_context())) {
- fprintf(stderr, "Could not allocate resample context\n");
- return AVERROR(ENOMEM);
- }
-
- /* Set the conversion parameters.
+ /*
+ * Create a resampler context for the conversion.
+ * Set the conversion parameters.
* Default channel layouts based on the number of channels
* are assumed for simplicity (they are sometimes not detected
* properly by the demuxer and/or decoder).
*/
- av_opt_set_int(*resample_context, "in_channel_layout",
- av_get_default_channel_layout(input_codec_context->channels), 0);
- av_opt_set_int(*resample_context, "out_channel_layout",
- av_get_default_channel_layout(output_codec_context->channels), 0);
- av_opt_set_int(*resample_context, "in_sample_rate",
- input_codec_context->sample_rate, 0);
- av_opt_set_int(*resample_context, "out_sample_rate",
- output_codec_context->sample_rate, 0);
- av_opt_set_int(*resample_context, "in_sample_fmt",
- input_codec_context->sample_fmt, 0);
- av_opt_set_int(*resample_context, "out_sample_fmt",
- output_codec_context->sample_fmt, 0);
+ *resample_context = swr_alloc_set_opts(NULL,
+ av_get_default_channel_layout(output_codec_context->channels),
+ output_codec_context->sample_fmt,
+ output_codec_context->sample_rate,
+ av_get_default_channel_layout(input_codec_context->channels),
+ input_codec_context->sample_fmt,
+ input_codec_context->sample_rate,
+ 0, NULL);
+ if (!*resample_context) {
+ fprintf(stderr, "Could not allocate resample context\n");
+ return AVERROR(ENOMEM);
+ }
+ /*
+ * Perform a sanity check so that the number of converted samples is
+ * not greater than the number of samples to be converted.
+ * If the sample rates differ, this case has to be handled differently
+ */
+ av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
/* Open the resampler with the specified parameters. */
- if ((error = avresample_open(*resample_context)) < 0) {
+ if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
- avresample_free(resample_context);
+ swr_free(resample_context);
return error;
}
- }
return 0;
}
@@ -357,7 +343,7 @@ static int write_output_file_header(AVFormatContext *output_format_context)
int error;
if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
fprintf(stderr, "Could not write output file header (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
return error;
}
return 0;
@@ -388,12 +374,12 @@ static int decode_audio_frame(AVFrame *frame,
/* Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
- /* If we are the the end of the file, flush the decoder below. */
+ /* If we are at the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
fprintf(stderr, "Could not read frame (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
return error;
}
}
@@ -405,7 +391,7 @@ static int decode_audio_frame(AVFrame *frame,
if ((error = avcodec_decode_audio4(input_codec_context, frame,
data_present, &input_packet)) < 0) {
fprintf(stderr, "Could not decode frame (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
av_packet_unref(&input_packet);
return error;
}
@@ -454,7 +440,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
output_codec_context->sample_fmt, 0)) < 0) {
fprintf(stderr,
"Could not allocate converted input samples (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return error;
@@ -474,28 +460,21 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
* @param resample_context Resample context for the conversion
* @return Error code (0 if successful)
*/
-static int convert_samples(uint8_t **input_data,
+static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
- AVAudioResampleContext *resample_context)
+ SwrContext *resample_context)
{
int error;
/* Convert the samples using the resampler. */
- if ((error = avresample_convert(resample_context, converted_data, 0,
- frame_size, input_data, 0, frame_size)) < 0) {
+ if ((error = swr_convert(resample_context,
+ converted_data, frame_size,
+ input_data , frame_size)) < 0) {
fprintf(stderr, "Could not convert input samples (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
return error;
}
- /* Perform a sanity check so that the number of converted samples is
- * not greater than the number of samples to be converted.
- * If the sample rates differ, this case has to be handled differently. */
- if (avresample_available(resample_context)) {
- fprintf(stderr, "Converted samples left over\n");
- return AVERROR_EXIT;
- }
-
return 0;
}
@@ -536,7 +515,7 @@ static int add_samples_to_fifo(AVAudioFifo *fifo,
* @param input_format_context Format context of the input file
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
- * @param resample_context Resample context for the conversion
+ * @param resampler_context Resample context for the conversion
* @param[out] finished Indicates whether the end of file has
* been reached and all data has been
* decoded. If this flag is false,
@@ -549,7 +528,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
- AVAudioResampleContext *resample_context,
+ SwrContext *resampler_context,
int *finished)
{
/* Temporary storage of the input samples of the frame read from the file. */
@@ -582,8 +561,8 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
/* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples. */
- if (convert_samples(input_frame->extended_data, converted_input_samples,
- input_frame->nb_samples, resample_context))
+ if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
+ input_frame->nb_samples, resampler_context))
goto cleanup;
/* Add the converted input samples to the FIFO buffer for later processing. */
@@ -638,7 +617,7 @@ static int init_output_frame(AVFrame **frame,
* sure that the audio frame can hold as many samples as specified. */
if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
av_frame_free(frame);
return error;
}
@@ -679,7 +658,7 @@ static int encode_audio_frame(AVFrame *frame,
if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
frame, data_present)) < 0) {
fprintf(stderr, "Could not encode frame (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
av_packet_unref(&output_packet);
return error;
}
@@ -688,7 +667,7 @@ static int encode_audio_frame(AVFrame *frame,
if (*data_present) {
if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
av_packet_unref(&output_packet);
return error;
}
@@ -752,7 +731,7 @@ static int write_output_file_trailer(AVFormatContext *output_format_context)
int error;
if ((error = av_write_trailer(output_format_context)) < 0) {
fprintf(stderr, "Could not write output file trailer (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
return error;
}
return 0;
@@ -762,7 +741,7 @@ int main(int argc, char **argv)
{
AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
- AVAudioResampleContext *resample_context = NULL;
+ SwrContext *resample_context = NULL;
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
@@ -852,14 +831,11 @@ int main(int argc, char **argv)
cleanup:
if (fifo)
av_audio_fifo_free(fifo);
- if (resample_context) {
- avresample_close(resample_context);
- avresample_free(&resample_context);
- }
+ swr_free(&resample_context);
if (output_codec_context)
avcodec_free_context(&output_codec_context);
if (output_format_context) {
- avio_close(output_format_context->pb);
+ avio_closep(&output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-21 14:27:32 +0000