[FFmpeg-cvslog] Add an audio transcoding example.
Andreas Unterweger
git at videolan.org
Wed Nov 27 10:38:39 CET 2013
ffmpeg | branch: master | Andreas Unterweger <dustsigns at gmail.com> | Tue Oct 8 13:10:46 2013 +0200| [10421bcf0ab5d48fa3d84de803e657b77fe7d3c0] | committer: Anton Khirnov
Add an audio transcoding example.
Signed-off-by: Anton Khirnov <anton at khirnov.net>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=10421bcf0ab5d48fa3d84de803e657b77fe7d3c0
---
configure | 2 +
doc/Makefile | 3 +-
doc/examples/transcode_aac.c | 769 ++++++++++++++++++++++++++++++++++++++++++
3 files changed, 773 insertions(+), 1 deletion(-)
diff --git a/configure b/configure
index eddf40b..6dcfd1b 100755
--- a/configure
+++ b/configure
@@ -1043,6 +1043,7 @@ COMPONENT_LIST="
EXAMPLE_LIST="
output_example
+ transcode_aac_example
"
EXTERNAL_LIBRARY_LIST="
@@ -1952,6 +1953,7 @@ yadif_filter_deps="gpl"
# examples
output_example_deps="avcodec avformat avutil swscale"
+transcode_aac_example_deps="avcodec avformat avresample"
# libraries
avcodec_deps="avutil"
diff --git a/doc/Makefile b/doc/Makefile
index fb15896..3cd67df 100644
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -16,7 +16,8 @@ DOCS-$(CONFIG_TEXI2HTML) += $(HTMLPAGES)
DOCS = $(DOCS-yes)
DOC_EXAMPLES-$(CONFIG_OUTPUT_EXAMPLE) += output
-ALL_DOC_EXAMPLES = output
+DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
+ALL_DOC_EXAMPLES = output transcode_aac
DOC_EXAMPLES := $(DOC_EXAMPLES-yes:%=doc/examples/%$(EXESUF))
ALL_DOC_EXAMPLES := $(ALL_DOC_EXAMPLES:%=doc/examples/%$(EXESUF))
diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c
new file mode 100644
index 0000000..46f61d8
--- /dev/null
+++ b/doc/examples/transcode_aac.c
@@ -0,0 +1,769 @@
+/*
+ * 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 simple audio converter
+ * Convert an input audio file to AAC in an MP4 container using Libav.
+ * @author Andreas Unterweger (dustsigns at gmail.com)
+ */
+
+#include <stdio.h>
+
+#include "libavformat/avformat.h"
+#include "libavformat/avio.h"
+
+#include "libavcodec/avcodec.h"
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avstring.h"
+#include "libavutil/frame.h"
+#include "libavutil/opt.h"
+
+#include "libavresample/avresample.h"
+
+/** 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;
+ }
+
+ av_strlcpy((*output_format_context)->filename, filename,
+ sizeof((*output_format_context)->filename));
+
+ /** Find the encoder to be used by its name. */
+ if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
+ fprintf(stderr, "Could not find an AAC encoder.\n");
+ 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)->channel_layout = av_get_default_channel_layout(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 = av_frame_alloc())) {
+ 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 = output_codec_context->channel_layout;
+ (*frame)->format = output_codec_context->sample_fmt;
+ (*frame)->sample_rate = output_codec_context->sample_rate;
+
+ /**
+ * 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 audio file to an AAC file in an MP4 container. */
+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;
+ AVAudioFifo *fifo = NULL;
+ int ret = AVERROR_EXIT;
+
+ if (argc < 3) {
+ fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
+ exit(1);
+ }
+
+ /** Register all codecs and formats so that they can be used. */
+ av_register_all();
+ /** Open the input file for reading. */
+ if (open_input_file(argv[1], &input_format_context,
+ &input_codec_context))
+ goto cleanup;
+ /** Open the output file for writing. */
+ if (open_output_file(argv[2], 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;
+}
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