diff options
Diffstat (limited to 'doc/examples/transcode_aac.c')
| -rw-r--r-- | doc/examples/transcode_aac.c | 95 |
1 files changed, 39 insertions, 56 deletions
diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c index 6206afe2e4..bf0128f68d 100644 --- a/doc/examples/transcode_aac.c +++ b/doc/examples/transcode_aac.c @@ -1,18 +1,18 @@ /* - * 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 */ @@ -21,7 +21,7 @@ * 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. * @author Andreas Unterweger (dustsigns@gmail.com) */ @@ -33,11 +33,12 @@ #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 kbit/s */ #define OUTPUT_BIT_RATE 48000 @@ -226,52 +227,46 @@ 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. */ 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); - } - /** + * 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; } @@ -390,30 +385,21 @@ static int init_converted_samples(uint8_t ***converted_input_samples, * The conversion happens on a per-frame basis, the size of which is specified * by frame_size. */ -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)); 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; } @@ -450,7 +436,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, - AVAudioResampleContext *resampler_context, + SwrContext *resampler_context, int *finished) { /** Temporary storage of the input samples of the frame read from the file. */ @@ -487,7 +473,7 @@ 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, + if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples, input_frame->nb_samples, resampler_context)) goto cleanup; @@ -649,7 +635,7 @@ 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; @@ -753,10 +739,7 @@ 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_close(output_codec_context); if (output_format_context) { |