On Wed, Oct 9, 2013 at 4:41 PM, Anton Khirnov <an...@khirnov.net> wrote:
> From: Andreas Unterweger <dustsi...@gmail.com>
>
> Signed-off-by: Anton Khirnov <an...@khirnov.net>
> ---
> Added some fixes, mostly cosmetics, but also some functional (using
> frame->nb_samples instead of codec context frame size, which might not be
> constant, or known at all; only write output packet it the encoder actually
> returned it).
>
> A certain Diego should also help us with integrating this with the build
> system.
> ---
> doc/examples/Makefile | 15 +
> doc/examples/mp3_aac.c | 722
> ++++++++++++++++++++++++++++++++++++++++++++++++
> 2 files changed, 737 insertions(+)
> create mode 100644 doc/examples/Makefile
> create mode 100644 doc/examples/mp3_aac.c
>
> diff --git a/doc/examples/Makefile b/doc/examples/Makefile
> new file mode 100644
> index 0000000..ddc53bb
> --- /dev/null
> +++ b/doc/examples/Makefile
> @@ -0,0 +1,15 @@
> +CC=gcc
> +
> +CFLAGS=-c -Wall
> +LDFLAGS=-lavformat -lavcodec -lavresample -lavutil -lvo-aacenc -lpthread -lm
> +
> +all: mp3_to_aac
> +
> +mp3_to_aac: mp3_aac
> + $(CC) mp3_aac.o -o mp3_to_aac.exe $(LDFLAGS)
> +
> +mp3_aac: mp3_aac.c
> + $(CC) $(CFLAGS) mp3_aac.c
> +
> +clean:
> + rm -rf *.o *.exe
What about this makefile:
CC=gcc
CFLAGS=-Wall -O2
LDLIBS=-lavformat -lavcodec -lavresample -lavutil -lvo-aacenc
-lpthread -lm -lz -lbz2
all: mp3_aac
clean:
rm -f mp3_aac
.PHONY: all clean
This way, you can (test) compile the example with these steps:
./configure --enable-gpl --enable-version3 --prefix=/tmp/libav-master
--enable-libvo-aacenc
make install
make -C docs/examples CPPFLAGS="-I/tmp/libav-master/include"
LDFLAGS="-L/tmp/libav-master/lib"
I'd also include these tree lines into the commit description, so that
that they are at hand when you need to retest/recompile the example
with latest head.
The rest of the patch looks good to me, but please note that I had to
add -lz and -lbz2 to get the executable linked.
cheers,
Reinhard
> diff --git a/doc/examples/mp3_aac.c b/doc/examples/mp3_aac.c
> new file mode 100644
> index 0000000..e7131ca
> --- /dev/null
> +++ b/doc/examples/mp3_aac.c
> @@ -0,0 +1,722 @@
> +/*
> + * This file is part of Libav.
> + *
> + * Libav 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,
> + * 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
> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301
> USA
> + */
> +
> +/**
> + * @file MP3 to AAC converter
> + * Convert an MP3 file to an AAC in an MP4 container using Libav.
> + * Requires libvo_aacenc to show how to use an external library
> + * Use -lavformat -lavcodec -lavresample -lavutil -lvo-aacenc -lpthread -lm
> + * @author Andreas Unterweger (dustsi...@gmail.com)
> + */
> +
> +#include <stdio.h>
> +#include <libavformat/avformat.h>
> +#include <libavformat/avio.h>
> +#include <libavcodec/avcodec.h>
> +#include <libavutil/audio_fifo.h>
> +#include <libavutil/frame.h>
> +#include <libavutil/opt.h>
> +#include <libavresample/avresample.h>
> +
> +/** The input file path */
> +#define INPUT_FILENAME "test.mp3"
> +/** The output file path */
> +#define OUTPUT_FILENAME "test.mp4"
> +/** The output codec name (AAC via libvo_aacenc) */
> +#define OUTPUT_CODEC_NAME "libvo_aacenc"
> +/** The output bit rate in kbit/s */
> +#define OUTPUT_BIT_RATE 48000
> +/** The number of output channels */
> +#define OUTPUT_CHANNELS 2
> +/** The audio sample output format */
> +#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
> +
> +/**
> + * Convert an error code into a text message.
> + * @param error Error code to be converted
> + * @return Corresponding error text (not thread-safe)
> + */
> +static char *const 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. */
> +static int open_input_file(const char *filename,
> + AVFormatContext **input_format_context,
> + AVCodecContext **input_codec_context)
> +{
> + AVCodec *input_codec;
> + int error;
> +
> + /** Open the input file to read from it. */
> + 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));
> + *input_format_context = NULL;
> + return error;
> + }
> +
> + /** 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));
> + avformat_close_input(input_format_context);
> + return error;
> + }
> +
> + /** Make sure that there is only one stream in the input file. */
> + if ((*input_format_context)->nb_streams != 1) {
> + fprintf(stderr, "Expected one audio input stream, but found %d\n",
> + (*input_format_context)->nb_streams);
> + avformat_close_input(input_format_context);
> + return AVERROR_EXIT;
> + }
> +
> + /** Find a decoder for the audio stream. */
> + if (!(input_codec =
> avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
> + fprintf(stderr, "Could not find input codec\n");
> + avformat_close_input(input_format_context);
> + return AVERROR_EXIT;
> + }
> +
> + /** 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) {
> + fprintf(stderr, "Could not open input codec (error '%s')\n",
> get_error_text(error));
> + 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;
> +
> + return 0;
> +}
> +
> +/**
> + * Open an output file and the required encoder.
> + * Also set some basic encoder parameters.
> + * Some of these parameters are based on the input file's parameters.
> + */
> +static int open_output_file(const char *filename,
> + AVCodecContext *input_codec_context,
> + AVFormatContext **output_format_context,
> + AVCodecContext **output_codec_context)
> +{
> + AVIOContext *output_io_context = NULL;
> + AVStream *stream = NULL;
> + AVCodec *output_codec = NULL;
> + int error;
> +
> + /** Open the output file to write to it. */
> + 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));
> + return error;
> + }
> +
> + /** Create a new format context for the output container format. */
> + if (!(*output_format_context = avformat_alloc_context())) {
> + fprintf(stderr, "Could not allocate output format context\n");
> + return AVERROR(ENOMEM);
> + }
> +
> + /** Associate the output file (pointer) with the container format
> context. */
> + (*output_format_context)->pb = output_io_context;
> +
> + /** Guess the desired container format based on the file extension. */
> + if (!((*output_format_context)->oformat = av_guess_format(NULL,
> filename, NULL))) {
> + fprintf(stderr, "Could not find output file format\n");
> + goto cleanup;
> + }
> +
> + /** Find the encoder to be used by its name. */
> + if (!(output_codec = avcodec_find_encoder_by_name(OUTPUT_CODEC_NAME))) {
> + fprintf(stderr, "Could not find output codec '%s'\n",
> OUTPUT_CODEC_NAME);
> + goto cleanup;
> + }
> +
> + /** Create a new audio stream in the output file container. */
> + if (!(stream = avformat_new_stream(*output_format_context,
> output_codec))) {
> + fprintf(stderr, "Could not create new stream\n");
> + error = AVERROR(ENOMEM);
> + goto cleanup;
> + }
> +
> + /** Save the encoder context for easiert access later. */
> + *output_codec_context = stream->codec;
> +
> + /**
> + * 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)->sample_rate = input_codec_context->sample_rate;
> + (*output_codec_context)->sample_fmt = AV_SAMPLE_FMT_S16;
> + (*output_codec_context)->bit_rate = OUTPUT_BIT_RATE;
> +
> + /**
> + * Some container formats (like MP4) require global headers to be present
> + * Mark the encoder so that it behaves accordingly.
> + */
> + if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
> + (*output_codec_context)->flags |= 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) {
> + fprintf(stderr, "Could not open output codec (error '%s')\n",
> get_error_text(error));
> + goto cleanup;
> + }
> +
> + return 0;
> +cleanup:
> + avio_close((*output_format_context)->pb);
> + avformat_free_context(*output_format_context);
> + *output_format_context = NULL;
> + return error < 0 ? error : AVERROR_EXIT;
> +}
> +
> +/** Initialize one data packet for reading or writing. */
> +static void init_packet(AVPacket *packet)
> +{
> + av_init_packet(packet);
> + /** Set the packet data and size so that it is recognized as being
> empty. */
> + packet->data = NULL;
> + packet->size = 0;
> +}
> +
> +/** Initialize one audio frame for reading from the input file */
> +static int init_input_frame(AVFrame **frame)
> +{
> + if (!(*frame = avcodec_alloc_frame())) {
> + fprintf(stderr, "Could not allocate input frame\n");
> + return AVERROR(ENOMEM);
> + }
> + return 0;
> +}
> +
> +/**
> + * 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.
> + */
> +static int init_resampler(AVCodecContext *input_codec_context,
> + AVCodecContext *output_codec_context,
> + AVAudioResampleContext **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.
> + * 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);
> +
> + /** Open the resampler with the specified parameters. */
> + if ((error = avresample_open(*resample_context)) < 0) {
> + fprintf(stderr, "Could not open resample context\n");
> + avresample_free(resample_context);
> + return error;
> + }
> + }
> + return 0;
> +}
> +
> +/** Initialize a FIFO buffer for the audio samples to be encoded. */
> +static int init_fifo(AVAudioFifo **fifo)
> +{
> + /** Create the FIFO buffer based on the specified output sample format.
> */
> + if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS,
> 1))) {
> + fprintf(stderr, "Could not allocate FIFO\n");
> + return AVERROR(ENOMEM);
> + }
> + return 0;
> +}
> +
> +/** Write the header of the output file container. */
> +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));
> + return error;
> + }
> + return 0;
> +}
> +
> +/** Decode one audio frame from the input file. */
> +static int decode_audio_frame(AVFrame *frame,
> + AVFormatContext *input_format_context,
> + AVCodecContext *input_codec_context,
> + int *data_present, int *finished)
> +{
> + /** Packet used for temporary storage. */
> + AVPacket input_packet;
> + int error;
> + init_packet(&input_packet);
> +
> + /** 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 (error == AVERROR_EOF)
> + *finished = 1;
> + else {
> + fprintf(stderr, "Could not read frame (error '%s')\n",
> get_error_text(error));
> + return error;
> + }
> + }
> +
> + /**
> + * Decode the audio frame stored in the temporary packet.
> + * The input audio stream decoder is used to do this.
> + * If we are at the end of the file, pass an empty packet to the decoder
> + * to flush it.
> + */
> + 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_free_packet(&input_packet);
> + return error;
> + }
> +
> + /**
> + * If the decoder has not been flushed completely, we are not finished,
> + * so that this function has to be called again.
> + */
> + if (*finished && *data_present)
> + *finished = 0;
> + av_free_packet(&input_packet);
> + return 0;
> +}
> +
> +/**
> + * Initialize a temporary storage for the specified number of audio samples.
> + * The conversion requires temporary storage due to the different format.
> + * The number of audio samples to be allocated is specified in frame_size.
> + */
> +static int init_converted_samples(uint8_t ***converted_input_samples,
> + AVCodecContext *output_codec_context,
> + int frame_size)
> +{
> + int error;
> +
> + /**
> + * Allocate as many pointers as there are audio channels.
> + * Each pointer will later point to the audio samples of the
> corresponding
> + * channels (although it may be NULL for interleaved formats).
> + */
> + if (!(*converted_input_samples = calloc(output_codec_context->channels,
> sizeof(**converted_input_samples)))) {
> + fprintf(stderr, "Could not allocate converted input sample
> pointers\n");
> + return AVERROR(ENOMEM);
> + }
> +
> + /**
> + * Allocate memory for the samples of all channels in one consecutive
> + * block for convenience.
> + */
> + if ((error = av_samples_alloc(*converted_input_samples, NULL,
> output_codec_context->channels,
> + frame_size,
> output_codec_context->sample_fmt, 0)) < 0) {
> + fprintf(stderr, "Could not allocate converted input samples (error
> '%s')\n", get_error_text(error));
> + av_freep(&converted_input_samples[0]);
> + free(converted_input_samples);
> + return error;
> + }
> + return 0;
> +}
> +
> +/**
> + * Convert the input audio samples into the output sample format.
> + * 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,
> + uint8_t **converted_data, const int frame_size,
> + AVAudioResampleContext *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) {
> + 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;
> +}
> +
> +/** Add converted input audio samples to the FIFO buffer for later
> processing. */
> +static int add_samples_to_fifo(AVAudioFifo *fifo,
> + uint8_t **converted_input_samples,
> + const int frame_size)
> +{
> + int error;
> +
> + /**
> + * Make the FIFO as large as it needs to be to hold both,
> + * the old and the new samples.
> + */
> + if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) +
> frame_size)) < 0) {
> + fprintf(stderr, "Could not reallocate FIFO\n");
> + return error;
> + }
> +
> + /** Store the new samples in the FIFO buffer. */
> + if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
> frame_size) < frame_size) {
> + fprintf(stderr, "Could not write data to FIFO\n");
> + return AVERROR_EXIT;
> + }
> + return 0;
> +}
> +
> +/**
> + * Read one audio frame from the input file, decodes, converts and stores
> + * it in the FIFO buffer.
> + */
> +static int read_decode_convert_and_store(AVAudioFifo *fifo, AVFormatContext
> *input_format_context,
> + AVCodecContext *input_codec_context,
> + AVCodecContext
> *output_codec_context,
> + AVAudioResampleContext
> *resampler_context, int *finished)
> +{
> + /** Temporary storage of the input samples of the frame read from the
> file. */
> + AVFrame *input_frame = NULL;
> + /** Temporary storage for the converted input samples. */
> + uint8_t **converted_input_samples = NULL;
> + int data_present;
> + int ret = AVERROR_EXIT;
> +
> + /** Initialize temporary storage for one input frame. */
> + if (init_input_frame(&input_frame))
> + goto cleanup;
> + /** Decode one frame worth of audio samples. */
> + if (decode_audio_frame(input_frame, input_format_context,
> input_codec_context, &data_present, finished))
> + goto cleanup;
> + /**
> + * If we are at the end of the file and there are no more samples
> + * in the decoder which are delayed, we are actually finished.
> + * This must not be treated as an error.
> + */
> + if (*finished && !data_present) {
> + ret = 0;
> + goto cleanup;
> + }
> + /** If there is decoded data, convert and store it */
> + if (data_present) {
> + /** Initialize the temporary storage for the converted input
> samples. */
> + if (init_converted_samples(&converted_input_samples,
> output_codec_context,
> + input_frame->nb_samples))
> + goto cleanup;
> +
> + /**
> + * 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, resampler_context))
> + goto cleanup;
> +
> + /** Add the converted input samples to the FIFO buffer for later
> processing. */
> + if (add_samples_to_fifo(fifo, converted_input_samples,
> input_frame->nb_samples))
> + goto cleanup;
> + ret = 0;
> + }
> + ret = 0;
> +
> +cleanup:
> + if (converted_input_samples) {
> + av_freep(&converted_input_samples[0]);
> + free(converted_input_samples);
> + }
> + av_frame_free(&input_frame);
> +
> + return ret;
> +}
> +
> +/**
> + * Initialize one input frame for writing to the output file.
> + * The frame will be exactly frame_size samples large.
> + */
> +static int init_output_frame(AVFrame **frame,
> + AVCodecContext *output_codec_context,
> + int frame_size)
> +{
> + int error;
> +
> + /** Create a new frame to store the audio samples. */
> + if (!(*frame = av_frame_alloc())) {
> + fprintf(stderr, "Could not allocate output frame\n");
> + return AVERROR_EXIT;
> + }
> +
> + /**
> + * Set the frame's parameters, especially its size and format.
> + * av_frame_get_buffer needs this to allocate memory for the
> + * audio samples of the frame.
> + * Default channel layouts based on the number of channels
> + * are assumed for simplicity.
> + */
> + (*frame)->nb_samples = frame_size;
> + (*frame)->channel_layout =
> av_get_default_channel_layout(output_codec_context->channels);
> + (*frame)->format = output_codec_context->sample_fmt;
> +
> + /**
> + * Allocate the samples of the created frame. This call will make
> + * sure that the audio frame can hold as many samples as specified.
> + */
> + if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
> + fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
> + get_error_text(error));
> + av_frame_free(frame);
> + return error;
> + }
> +
> + return 0;
> +}
> +
> +/** Encode one frame worth of audio to the output file. */
> +static int encode_audio_frame(AVFrame *frame, AVFormatContext
> *output_format_context,
> + AVCodecContext *output_codec_context, int
> *data_present)
> +{
> + /** Packet used for temporary storage. */
> + AVPacket output_packet;
> + int error;
> + init_packet(&output_packet);
> +
> + /**
> + * Encode the audio frame and store it in the temporary packet.
> + * The output audio stream encoder is used to do this.
> + */
> + 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_free_packet(&output_packet);
> + return error;
> + }
> +
> + /** Write one audio frame from the temporary packet to the output file.
> */
> + 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_free_packet(&output_packet);
> + return error;
> + }
> +
> + av_free_packet(&output_packet);
> + }
> +
> + return 0;
> +}
> +
> +/**
> + * Load one audio frame from the FIFO buffer, encode and write it to the
> + * output file.
> + */
> +static int load_encode_and_write(AVAudioFifo *fifo, AVFormatContext
> *output_format_context,
> + AVCodecContext *output_codec_context)
> +{
> + /** Temporary storage of the output samples of the frame written to the
> file. */
> + AVFrame *output_frame;
> + /**
> + * Use the maximum number of possible samples per frame.
> + * If there is less than the maximum possible frame size in the FIFO
> + * buffer use this number. Otherwise, use the maximum possible frame size
> + */
> + const int frame_size = FFMIN(av_audio_fifo_size(fifo),
> output_codec_context->frame_size);
> + int data_written;
> +
> + /** Initialize temporary storage for one output frame. */
> + if (init_output_frame(&output_frame, output_codec_context, frame_size))
> + return AVERROR_EXIT;
> +
> + /**
> + * Read as many samples from the FIFO buffer as required to fill the
> frame.
> + * The samples are stored in the frame temporarily.
> + */
> + if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) <
> frame_size) {
> + fprintf(stderr, "Could not read data from FIFO\n");
> + av_frame_free(&output_frame);
> + return AVERROR_EXIT;
> + }
> +
> + /** Encode one frame worth of audio samples. */
> + if (encode_audio_frame(output_frame, output_format_context,
> output_codec_context, &data_written)) {
> + av_frame_free(&output_frame);
> + return AVERROR_EXIT;
> + }
> + av_frame_free(&output_frame);
> + return 0;
> +}
> +
> +/** Write the trailer of the output file container. */
> +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));
> + return error;
> + }
> + return 0;
> +}
> +
> +/** Convert an MP3 to an AAC file in an MP4 container. */
> +int main(void)
> +{
> + AVFormatContext *input_format_context = NULL, *output_format_context
> = NULL;
> + AVCodecContext *input_codec_context = NULL, *output_codec_context =
> NULL;
> + AVAudioResampleContext *resample_context = NULL;
> + AVAudioFifo *fifo = NULL;
> + int ret = AVERROR_EXIT;
> +
> + /** Register all codecs and formats so that they can be used. */
> + av_register_all();
> + /** Open the input file for reading. */
> + if (open_input_file(INPUT_FILENAME, &input_format_context,
> &input_codec_context))
> + goto cleanup;
> + /** Open the output file for writing. */
> + if (open_output_file(OUTPUT_FILENAME, input_codec_context,
> &output_format_context, &output_codec_context))
> + goto cleanup;
> + /** Initialize the resampler to be able to convert audio sample formats.
> */
> + if (init_resampler(input_codec_context, output_codec_context,
> &resample_context))
> + goto cleanup;
> + /** Initialize the FIFO buffer to store audio samples to be encoded. */
> + if (init_fifo(&fifo))
> + goto cleanup;
> + /** Write the header of the output file container. */
> + if (write_output_file_header(output_format_context))
> + goto cleanup;
> +
> + /**
> + * Loop as long as we have input samples to read or output samples
> + * to write; abort as soon as we have neither.
> + */
> + while (1) {
> + /** Use the encoder's desired frame size for processing. */
> + const int output_frame_size = output_codec_context->frame_size;
> + int finished = 0;
> +
> + /**
> + * Make sure that there is one frame worth of samples in the FIFO
> + * buffer so that the encoder can do its work.
> + * Since the decoder's and the encoder's frame size may differ, we
> + * need to FIFO buffer to store as many frames worth of input samples
> + * that they make up at least one frame worth of output samples.
> + */
> + while (av_audio_fifo_size(fifo) < output_frame_size) {
> + /**
> + * Decode one frame worth of audio samples, convert it to the
> + * output sample format and put it into the FIFO buffer.
> + */
> + if (read_decode_convert_and_store(fifo, input_format_context,
> input_codec_context, output_codec_context, resample_context, &finished))
> + goto cleanup;
> +
> + /**
> + * If we are at the end of the input file, we continue
> + * encoding the remaining audio samples to the output file.
> + */
> + if (finished)
> + break;
> + }
> +
> + /**
> + * If we have enough samples for the encoder, we encode them.
> + * At the end of the file, we pass the remaining samples to
> + * the encoder.
> + */
> + while (av_audio_fifo_size(fifo) >= output_frame_size || (finished &&
> av_audio_fifo_size(fifo) > 0))
> + /**
> + * Take one frame worth of audio samples from the FIFO buffer,
> + * encode it and write it to the output file.
> + */
> + if (load_encode_and_write(fifo, output_format_context,
> output_codec_context))
> + goto cleanup;
> +
> + /**
> + * If we are at the end of the input file and have encoded
> + * all remaining samples, we can exit this loop and finish.
> + */
> + if (finished) {
> + int data_written;
> + /** Flush the encoder as it may have delayed frames. */
> + do {
> + if (encode_audio_frame(NULL, output_format_context,
> output_codec_context, &data_written))
> + goto cleanup;
> + } while (data_written);
> + break;
> + }
> + }
> +
> + /** Write the trailer of the output file container. */
> + if (write_output_file_trailer(output_format_context))
> + goto cleanup;
> + ret = 0;
> +
> +cleanup:
> + if (fifo)
> + av_audio_fifo_free(fifo);
> + if (resample_context) {
> + avresample_close(resample_context);
> + avresample_free(&resample_context);
> + }
> + if (output_codec_context)
> + avcodec_close(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);
> + if (input_format_context)
> + avformat_close_input(&input_format_context);
> + return ret;
> +}
> --
> 1.7.10.4
>
> _______________________________________________
> libav-devel mailing list
> libav-devel@libav.org
> https://lists.libav.org/mailman/listinfo/libav-devel
--
regards,
Reinhard
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