[FFmpeg-cvslog] Merge commit 'b200a2c8da403b5a5c8b50f8cb4a75fd4f0131b1'
James Almer
git at videolan.org
Thu Oct 26 23:10:47 EEST 2017
ffmpeg | branch: master | James Almer <jamrial at gmail.com> | Thu Oct 26 16:58:39 2017 -0300| [8a3d3b624075c56b844cbd0f3d58c6e291c8e18c] | committer: James Almer
Merge commit 'b200a2c8da403b5a5c8b50f8cb4a75fd4f0131b1'
* commit 'b200a2c8da403b5a5c8b50f8cb4a75fd4f0131b1':
examples: Fixed and extended Doxygen documentation
Merged-by: James Almer <jamrial at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8a3d3b624075c56b844cbd0f3d58c6e291c8e18c
---
doc/examples/transcode_aac.c | 361 +++++++++++++++++++++++++------------------
1 file changed, 209 insertions(+), 152 deletions(-)
diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c
index 6c2f4fb475..9fd5c00d60 100644
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@ -1,4 +1,6 @@
/*
+ * Copyright (c) 2013-2017 Andreas Unterweger
+ *
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
@@ -8,7 +10,7 @@
*
* 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
+ * 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
@@ -18,10 +20,11 @@
/**
* @file
- * simple audio converter
+ * Simple audio converter
*
* @example transcode_aac.c
* 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 at gmail.com)
*/
@@ -40,12 +43,18 @@
#include "libswresample/swresample.h"
-/** The output bit rate in kbit/s */
+/* The output bit rate in bit/s */
#define OUTPUT_BIT_RATE 96000
-/** The number of output channels */
+/* The number of output channels */
#define OUTPUT_CHANNELS 2
-/** Open an input file and the required decoder. */
+/**
+ * Open an input file and the required decoder.
+ * @param filename File to be opened
+ * @param[out] input_format_context Format context of opened file
+ * @param[out] input_codec_context Codec context of opened file
+ * @return Error code (0 if successful)
+ */
static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
@@ -54,7 +63,7 @@ static int open_input_file(const char *filename,
AVCodec *input_codec;
int error;
- /** Open the input file to read from it. */
+ /* 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",
@@ -63,7 +72,7 @@ static int open_input_file(const char *filename,
return error;
}
- /** Get information on the input file (number of streams etc.). */
+ /* 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",
av_err2str(error));
@@ -71,7 +80,7 @@ static int open_input_file(const char *filename,
return error;
}
- /** Make sure that there is only one stream in the input file. */
+ /* 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);
@@ -79,14 +88,14 @@ static int open_input_file(const char *filename,
return AVERROR_EXIT;
}
- /** Find a decoder for the audio stream. */
+ /* Find a decoder for the audio stream. */
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 */
+ /* Allocate a new decoding context. */
avctx = avcodec_alloc_context3(input_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate a decoding context\n");
@@ -94,7 +103,7 @@ static int open_input_file(const char *filename,
return AVERROR(ENOMEM);
}
- /** initialize the stream parameters with demuxer information */
+ /* 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);
@@ -102,7 +111,7 @@ static int open_input_file(const char *filename,
return error;
}
- /** Open the decoder for the audio stream to use it later. */
+ /* 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",
av_err2str(error));
@@ -111,7 +120,7 @@ static int open_input_file(const char *filename,
return error;
}
- /** Save the decoder context for easier access later. */
+ /* Save the decoder context for easier access later. */
*input_codec_context = avctx;
return 0;
@@ -121,6 +130,11 @@ static int open_input_file(const char *filename,
* 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.
+ * @param filename File to be opened
+ * @param input_codec_context Codec context of input file
+ * @param[out] output_format_context Format context of output file
+ * @param[out] output_codec_context Codec context of output file
+ * @return Error code (0 if successful)
*/
static int open_output_file(const char *filename,
AVCodecContext *input_codec_context,
@@ -133,7 +147,7 @@ static int open_output_file(const char *filename,
AVCodec *output_codec = NULL;
int error;
- /** Open the output file to write to it. */
+ /* 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",
@@ -141,16 +155,16 @@ static int open_output_file(const char *filename,
return error;
}
- /** Create a new format context for the output container format. */
+ /* 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. */
+ /* 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. */
+ /* 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");
@@ -160,13 +174,13 @@ static int open_output_file(const char *filename,
av_strlcpy((*output_format_context)->filename, filename,
sizeof((*output_format_context)->filename));
- /** Find the encoder to be used by its name. */
+ /* 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. */
+ /* Create a new audio stream in the output file container. */
if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
@@ -180,31 +194,27 @@ static int open_output_file(const char *filename,
goto cleanup;
}
- /**
- * Set the basic encoder parameters.
- * The input file's sample rate is used to avoid a sample rate conversion.
- */
+ /* Set the basic encoder parameters.
+ * The input file's sample rate is used to avoid a sample rate conversion. */
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 */
+ /* Allow the use of the experimental AAC encoder. */
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
- /** Set the sample rate for the container. */
+ /* Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.num = 1;
- /**
- * Some container formats (like MP4) require global headers to be present
- * Mark the encoder so that it behaves accordingly.
- */
+ /* 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)
avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
- /** Open the encoder for the audio stream to use it later. */
+ /* 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",
av_err2str(error));
@@ -217,7 +227,7 @@ static int open_output_file(const char *filename,
goto cleanup;
}
- /** Save the encoder context for easier access later. */
+ /* Save the encoder context for easier access later. */
*output_codec_context = avctx;
return 0;
@@ -230,16 +240,23 @@ cleanup:
return error < 0 ? error : AVERROR_EXIT;
}
-/** Initialize one data packet for reading or writing. */
+/**
+ * Initialize one data packet for reading or writing.
+ * @param packet Packet to be initialized
+ */
static void init_packet(AVPacket *packet)
{
av_init_packet(packet);
- /** Set the packet data and size so that it is recognized as being empty. */
+ /* 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 */
+/**
+ * Initialize one audio frame for reading from the input file.
+ * @param[out] frame Frame to be initialized
+ * @return Error code (0 if successful)
+ */
static int init_input_frame(AVFrame **frame)
{
if (!(*frame = av_frame_alloc())) {
@@ -253,6 +270,10 @@ 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
* 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
+ * @return Error code (0 if successful)
*/
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
@@ -260,7 +281,7 @@ static int init_resampler(AVCodecContext *input_codec_context,
{
int error;
- /**
+ /*
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
@@ -279,14 +300,14 @@ static int init_resampler(AVCodecContext *input_codec_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. */
+ /* Open the resampler with the specified parameters. */
if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(resample_context);
@@ -295,10 +316,15 @@ static int init_resampler(AVCodecContext *input_codec_context,
return 0;
}
-/** Initialize a FIFO buffer for the audio samples to be encoded. */
+/**
+ * Initialize a FIFO buffer for the audio samples to be encoded.
+ * @param[out] fifo Sample buffer
+ * @param output_codec_context Codec context of the output file
+ * @return Error code (0 if successful)
+ */
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
{
- /** Create the FIFO buffer based on the specified output sample format. */
+ /* Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
output_codec_context->channels, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
@@ -307,7 +333,11 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
return 0;
}
-/** Write the header of the output file container. */
+/**
+ * Write the header of the output file container.
+ * @param output_format_context Format context of the output file
+ * @return Error code (0 if successful)
+ */
static int write_output_file_header(AVFormatContext *output_format_context)
{
int error;
@@ -319,20 +349,32 @@ static int write_output_file_header(AVFormatContext *output_format_context)
return 0;
}
-/** Decode one audio frame from the input file. */
+/**
+ * Decode one audio frame from the input file.
+ * @param frame Audio frame to be decoded
+ * @param input_format_context Format context of the input file
+ * @param input_codec_context Codec context of the input file
+ * @param[out] data_present Indicates whether data has been decoded
+ * @param[out] finished Indicates whether the end of file has
+ * been reached and all data has been
+ * decoded. If this flag is false, there
+ * is more data to be decoded, i.e., this
+ * function has to be called again.
+ * @return Error code (0 if successful)
+ */
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. */
+ /* 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. */
+ /* 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 at 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 {
@@ -342,12 +384,10 @@ static int decode_audio_frame(AVFrame *frame,
}
}
- /**
- * Decode the audio frame stored in the temporary packet.
+ /* 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.
- */
+ * 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",
@@ -356,10 +396,8 @@ static int decode_audio_frame(AVFrame *frame,
return error;
}
- /**
- * If the decoder has not been flushed completely, we are not finished,
- * so that this function has to be called again.
- */
+ /* 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_packet_unref(&input_packet);
@@ -370,6 +408,13 @@ static int decode_audio_frame(AVFrame *frame,
* 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.
+ * @param[out] converted_input_samples Array of converted samples. The
+ * dimensions are reference, channel
+ * (for multi-channel audio), sample.
+ * @param output_codec_context Codec context of the output file
+ * @param frame_size Number of samples to be converted in
+ * each round
+ * @return Error code (0 if successful)
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
AVCodecContext *output_codec_context,
@@ -377,8 +422,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
{
int error;
- /**
- * Allocate as many pointers as there are audio channels.
+ /* 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).
*/
@@ -388,10 +432,8 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
return AVERROR(ENOMEM);
}
- /**
- * Allocate memory for the samples of all channels in one consecutive
- * block for convenience.
- */
+ /* 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,
@@ -408,8 +450,15 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
/**
* 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.
+ * The conversion happens on a per-frame basis, the size of which is
+ * specified by frame_size.
+ * @param input_data Samples to be decoded. The dimensions are
+ * channel (for multi-channel audio), sample.
+ * @param[out] converted_data Converted samples. The dimensions are channel
+ * (for multi-channel audio), sample.
+ * @param frame_size Number of samples to be converted
+ * @param resample_context Resample context for the conversion
+ * @return Error code (0 if successful)
*/
static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
@@ -417,7 +466,7 @@ static int convert_samples(const uint8_t **input_data,
{
int error;
- /** Convert the samples using the resampler. */
+ /* Convert the samples using the resampler. */
if ((error = swr_convert(resample_context,
converted_data, frame_size,
input_data , frame_size)) < 0) {
@@ -429,23 +478,28 @@ static int convert_samples(const uint8_t **input_data,
return 0;
}
-/** Add converted input audio samples to the FIFO buffer for later processing. */
+/**
+ * Add converted input audio samples to the FIFO buffer for later processing.
+ * @param fifo Buffer to add the samples to
+ * @param converted_input_samples Samples to be added. The dimensions are channel
+ * (for multi-channel audio), sample.
+ * @param frame_size Number of samples to be converted
+ * @return Error code (0 if successful)
+ */
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.
- */
+ /* 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. */
+ /* 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");
@@ -455,8 +509,20 @@ static int add_samples_to_fifo(AVAudioFifo *fifo,
}
/**
- * Read one audio frame from the input file, decodes, converts and stores
+ * Read one audio frame from the input file, decode, convert and store
* it in the FIFO buffer.
+ * @param fifo Buffer used for temporary storage
+ * @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 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,
+ * there is more data to be decoded,
+ * i.e., this function has to be called
+ * again.
+ * @return Error code (0 if successful)
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
@@ -465,45 +531,41 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
SwrContext *resampler_context,
int *finished)
{
- /** Temporary storage of the input samples of the frame read from the file. */
+ /* Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = NULL;
- /** Temporary storage for the converted input samples. */
+ /* 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. */
+ /* Initialize temporary storage for one input frame. */
if (init_input_frame(&input_frame))
goto cleanup;
- /** Decode one frame worth of audio samples. */
+ /* 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
+ /* 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.
- */
+ * 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 there is decoded data, convert and store it. */
if (data_present) {
- /** Initialize the temporary storage for the converted input samples. */
+ /* 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.
- */
+ /* Convert the input samples to the desired output sample format.
+ * This requires a temporary storage provided by converted_input_samples. */
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. */
+ /* 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;
@@ -524,6 +586,10 @@ cleanup:
/**
* Initialize one input frame for writing to the output file.
* The frame will be exactly frame_size samples large.
+ * @param[out] frame Frame to be initialized
+ * @param output_codec_context Codec context of the output file
+ * @param frame_size Size of the frame
+ * @return Error code (0 if successful)
*/
static int init_output_frame(AVFrame **frame,
AVCodecContext *output_codec_context,
@@ -531,28 +597,24 @@ static int init_output_frame(AVFrame **frame,
{
int error;
- /** Create a new frame to store the audio samples. */
+ /* 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.
+ /* 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.
- */
+ * 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.
- */
+ /* 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 not allocate output frame samples (error '%s')\n",
av_err2str(error));
@@ -563,30 +625,36 @@ static int init_output_frame(AVFrame **frame,
return 0;
}
-/** Global timestamp for the audio frames */
+/* Global timestamp for the audio frames. */
static int64_t pts = 0;
-/** Encode one frame worth of audio to the output file. */
+/**
+ * Encode one frame worth of audio to the output file.
+ * @param frame Samples to be encoded
+ * @param output_format_context Format context of the output file
+ * @param output_codec_context Codec context of the output file
+ * @param[out] data_present Indicates whether data has been
+ * decoded
+ * @return Error code (0 if successful)
+ */
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
int *data_present)
{
- /** Packet used for temporary storage. */
+ /* Packet used for temporary storage. */
AVPacket output_packet;
int error;
init_packet(&output_packet);
- /** Set a timestamp based on the sample rate for the container. */
+ /* Set a timestamp based on the sample rate for the container. */
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
- /**
- * Encode the audio frame and store it in the temporary packet.
- * The output audio stream encoder is used to do this.
- */
+ /* 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",
@@ -595,7 +663,7 @@ static int encode_audio_frame(AVFrame *frame,
return error;
}
- /** Write one audio frame from the temporary packet to the output file. */
+ /* 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",
@@ -613,37 +681,37 @@ static int encode_audio_frame(AVFrame *frame,
/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
+ * @param fifo Buffer used for temporary storage
+ * @param output_format_context Format context of the output file
+ * @param output_codec_context Codec context of the output file
+ * @return Error code (0 if successful)
*/
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. */
+ /* 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.
+ /* 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
- */
+ * 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. */
+ /* 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.
- */
+ /* 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. */
+ /* 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);
@@ -653,7 +721,11 @@ static int load_encode_and_write(AVAudioFifo *fifo,
return 0;
}
-/** Write the trailer of the output file container. */
+/**
+ * Write the trailer of the output file container.
+ * @param output_format_context Format context of the output file
+ * @return Error code (0 if successful)
+ */
static int write_output_file_trailer(AVFormatContext *output_format_context)
{
int error;
@@ -665,7 +737,6 @@ static int write_output_file_trailer(AVFormatContext *output_format_context)
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;
@@ -674,89 +745,75 @@ int main(int argc, char **argv)
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
- if (argc < 3) {
+ 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. */
+ /* Register all codecs and formats so that they can be used. */
av_register_all();
- /** Open the input file for reading. */
+ /* 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. */
+ /* 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. */
+ /* 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. */
+ /* Initialize the FIFO buffer to store audio samples to be encoded. */
if (init_fifo(&fifo, output_codec_context))
goto cleanup;
- /** Write the header of the output file container. */
+ /* 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.
- */
+ /* 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. */
+ /* 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
+ /* 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.
- */
+ * 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.
- */
+ /* 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 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.
+ /* 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.
- */
+ * 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.
- */
+ /* 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 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. */
+ /* Flush the encoder as it may have delayed frames. */
do {
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
@@ -766,7 +823,7 @@ int main(int argc, char **argv)
}
}
- /** Write the trailer of the output file container. */
+ /* Write the trailer of the output file container. */
if (write_output_file_trailer(output_format_context))
goto cleanup;
ret = 0;
======================================================================
diff --cc doc/examples/transcode_aac.c
index 6c2f4fb475,44d5af6b04..9fd5c00d60
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@@ -1,14 -1,16 +1,16 @@@
/*
+ * 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
+ * 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
@@@ -18,10 -20,11 +20,11 @@@
/**
* @file
- * simple audio converter
+ * 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 at gmail.com)
*/
@@@ -38,14 -40,32 +41,20 @@@
#include "libavutil/frame.h"
#include "libavutil/opt.h"
-#include "libavresample/avresample.h"
+#include "libswresample/swresample.h"
- /** The output bit rate in kbit/s */
+ /* The output bit rate in bit/s */
#define OUTPUT_BIT_RATE 96000
- /** The number of output channels */
+ /* The number of output channels */
#define OUTPUT_CHANNELS 2
- /** Open an input file and the required decoder. */
+ /**
- * 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
+ * @param[out] input_codec_context Codec context of opened file
+ * @return Error code (0 if successful)
+ */
static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
@@@ -63,10 -83,10 +72,10 @@@
return error;
}
- /** Get information on the input file (number of streams etc.). */
+ /* 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;
}
@@@ -102,10 -122,10 +111,10 @@@
return error;
}
- /** Open the decoder for the audio stream to use it later. */
+ /* 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;
@@@ -204,10 -225,10 +214,10 @@@ static int open_output_file(const char
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
- /** Open the encoder for the audio stream to use it later. */
+ /* 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;
}
@@@ -252,46 -280,53 +269,50 @@@ static int init_input_frame(AVFrame **f
/**
* 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
+ * @return Error code (0 if successful)
*/
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. */
+ /* 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;
}
@@@ -330,9 -386,9 +372,9 @@@ static int decode_audio_frame(AVFrame *
int error;
init_packet(&input_packet);
- /** Read one audio frame from the input file into a temporary 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 at the end of the file, flush the decoder below. */
- /* 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 {
@@@ -408,21 -464,27 +450,28 @@@ static int init_converted_samples(uint8
/**
* 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.
+ * The conversion happens on a per-frame basis, the size of which is
+ * specified by frame_size.
+ * @param input_data Samples to be decoded. The dimensions are
+ * channel (for multi-channel audio), sample.
+ * @param[out] converted_data Converted samples. The dimensions are channel
+ * (for multi-channel audio), sample.
+ * @param frame_size Number of samples to be converted
+ * @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. */
+ /* 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;
}
@@@ -455,19 -530,31 +509,31 @@@ static int add_samples_to_fifo(AVAudioF
}
/**
- * Read one audio frame from the input file, decodes, converts and stores
+ * Read one audio frame from the input file, decode, convert and store
* it in the FIFO buffer.
+ * @param fifo Buffer used for temporary storage
+ * @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,
+ * there is more data to be decoded,
+ * i.e., this function has to be called
+ * again.
+ * @return Error code (0 if successful)
*/
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. */
+ /* Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = NULL;
- /** Temporary storage for the converted input samples. */
+ /* Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present;
int ret = AVERROR_EXIT;
@@@ -495,15 -580,13 +559,13 @@@
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.
- */
+ /* 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. */
+ /* 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;
@@@ -549,13 -634,11 +613,11 @@@ static int init_output_frame(AVFrame **
(*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.
- */
+ /* 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 not allocate output frame samples (error '%s')\n",
- get_error_text(error));
+ av_err2str(error));
av_frame_free(frame);
return error;
}
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