кодирование — перекодирование в vorbis с использованием библиотек FFmpeg, переполнение стека

Я сделал тестовое приложение для перекодирования в формат vorbis (контейнер webm).

Пока что, основываясь на примерах FFmpeg, все работает, и выходной файл воспроизводится правильно, но звук в правом канале отсутствует. Я пытался смотреть на разные возможности, но пока не смог найти ответа.

Для справки, это код, который я использую:

#include "stdafx.h"#define MAX_AUDIO_PACKET_SIZE (128 * 1024)

#include <iostream>
#include <fstream>

#include <string>
#include <vector>
#include <map>

#include <deque>
#include <queue>

#include <math.h>
#include <stdlib.h>
#include <stdio.h>
#include <conio.h>

extern "C"{
#include "libavcodec/avcodec.h"#include "libavformat/avformat.h"#include "libavdevice/avdevice.h"#include "libswscale/swscale.h"#include "libavutil/dict.h"#include "libavutil/error.h"#include "libavutil/opt.h"#include <libavutil/fifo.h>
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
}
AVCodecID           outputAudioFormat = AV_CODEC_ID_VORBIS;static int sws_flags = SWS_BICUBIC;
#define STREAM_DURATION   50.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
#define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */AVFormatContext*    fmt_ctx= NULL;
int                    audio_stream_index = -1;
AVCodecContext *    codec_ctx_audio = NULL;
AVCodec*            codec_audio = NULL;
AVFrame*            decoded_frame = NULL;
uint8_t**            audio_dst_data = NULL;
int                    got_frame = 0;
int                    audiobufsize = 0;
AVPacket            input_packet;
int                    audio_dst_linesize = 0;
int                    audio_dst_bufsize = 0;
SwrContext *        swrContext = NULL;

AVOutputFormat *    output_format = NULL ;
AVFormatContext *    output_fmt_ctx= NULL;
AVStream *            audio_st = NULL;
AVStream*           video_st = NULL;
AVCodec *            audio_codec = NULL;
AVCodec*            video_codec = NULL;
double                audio_pts = 0.0;
AVFrame *            out_frame = avcodec_alloc_frame();

int                    audio_input_frame_size = 64;

uint8_t *            audio_data_buf = NULL;
uint8_t *            audio_out = NULL;
int                    audio_bit_rate;
int                    audio_sample_rate;
int                    audio_channels;
int                 sourceSampleRate=0;
int                 destSampleRate = 0;

int                 dst_nb_samples = 0;
int                 pivotIndex = 0;
int                 max_dst_nb_samples = 0;
int                 samples_count=0;int decode_packet();
int open_audio_input(char* src_filename);
int decode_frame();

int open_encoder(char* output_filename);
AVStream *add_audio_stream(AVFormatContext *oc, AVCodec **codec,
enum AVCodecID codec_id);
int open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st);
void close_audio(AVFormatContext *oc, AVStream *st);
void write_audio_frame(uint8_t ** audio_src_data, int audio_src_bufsize);

static AVFrame *frame;
static AVPicture src_picture, dst_picture;
static int frame_count;
/* Add an output stream. */
static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
AVStream *st;

/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}

st = avformat_new_stream(oc, *codec);
if (!st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
st->id = oc->nb_streams-1;
c = st->codec;

switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
c->sample_fmt  = (*codec)->sample_fmts ?
(*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
c->bit_rate    = 64000;
c->sample_rate = 44100;
c->channels    = 2;
break;

case AVMEDIA_TYPE_VIDEO:
c->codec_id = codec_id;

c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width    = 352;
c->height   = 288;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
c->time_base.den = STREAM_FRAME_RATE;
c->time_base.num = 1;
c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt       = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;

default:
break;
}

/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;

return st;
}static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
int ret;
AVCodecContext *c = st->codec;

/* open the codec */
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
//fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
}

/* allocate and init a re-usable frame */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame->format = c->pix_fmt;
frame->width = c->width;
frame->height = c->height;

/* Allocate the encoded raw picture. */
ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
if (ret < 0) {
//fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
exit(1);
}

/* If the output format is not YUV420P, then a temporary YUV420P
* picture is needed too. It is then converted to the required
* output format. */
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
if (ret < 0) {
//fprintf(stderr, "Could not allocate temporary picture: %s\n",
//        av_err2str(ret));
exit(1);
}
}

/* copy data and linesize picture pointers to frame */
*((AVPicture *)frame) = dst_picture;
}

int open_audio_input(char* src_filename)
{
int i =0;
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0)
{
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}

// Retrieve stream information
if(avformat_find_stream_info(fmt_ctx, NULL)<0)
return -1; // Couldn't find stream information

// Dump information about file onto standard error
av_dump_format(fmt_ctx, 0, src_filename, 0);

// Find the first video stream
for(i=0; i<fmt_ctx->nb_streams; i++)
{
if(fmt_ctx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO)
{
audio_stream_index=i;
break;
}
}
if ( audio_stream_index != -1 )
{
// Get a pointer to the codec context for the audio stream
codec_ctx_audio=fmt_ctx->streams[audio_stream_index]->codec;

// Find the decoder for the video stream
codec_audio=avcodec_find_decoder(codec_ctx_audio->codec_id);
if(codec_audio==NULL) {
fprintf(stderr, "Unsupported audio codec!\n");
return -1; // Codec not found
}

// Open codec
AVDictionary *codecDictOptions = NULL;
if(avcodec_open2(codec_ctx_audio, codec_audio, &codecDictOptions)<0)
return -1; // Could not open codec

// Set up SWR context once you've got codec information
swrContext = swr_alloc();
av_opt_set_int(swrContext, "in_channel_layout",  codec_ctx_audio->channel_layout, 0);
av_opt_set_int(swrContext, "out_channel_layout", codec_ctx_audio->channel_layout,  0);
av_opt_set_int(swrContext, "in_sample_rate",     codec_ctx_audio->sample_rate, 0);
av_opt_set_int(swrContext, "out_sample_rate",    codec_ctx_audio->sample_rate, 0);
av_opt_set_sample_fmt(swrContext, "in_sample_fmt",  codec_ctx_audio->sample_fmt, 0);
if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
{
av_opt_set_sample_fmt(swrContext, "out_sample_fmt", AV_SAMPLE_FMT_FLTP,  0);
}
else
{
av_opt_set_sample_fmt(swrContext, "out_sample_fmt", AV_SAMPLE_FMT_S16,  0);
}
int rv = swr_init(swrContext);

sourceSampleRate    =   destSampleRate = codec_ctx_audio->sample_rate;

// Allocate audio frame
if ( decoded_frame == NULL ) decoded_frame = avcodec_alloc_frame();
int nb_planes = 0;
AVStream* audio_stream = fmt_ctx->streams[audio_stream_index];
nb_planes = av_sample_fmt_is_planar(codec_ctx_audio->sample_fmt) ? codec_ctx_audio->channels : 1;
int tempSize =  sizeof(uint8_t *) * nb_planes;
audio_dst_data = (uint8_t**)av_mallocz(tempSize);
if (!audio_dst_data)
{
fprintf(stderr, "Could not allocate audio data buffers\n");
}
else
{
for ( int i = 0 ; i < nb_planes ; i ++ )
{
audio_dst_data[i] = NULL;
}
}
}
}

int decode_frame()
{
int rv = 0;
got_frame = 0;
if ( fmt_ctx == NULL  )
{
return rv;
}
int ret = 0;
audiobufsize = 0;
rv = av_read_frame(fmt_ctx, &input_packet);
if ( rv < 0 )
{
return rv;
}
rv = decode_packet();
// Free the input_packet that was allocated by av_read_frame
av_free_packet(&input_packet);
return rv;
}

int decode_packet()
{
int rv = 0;
int ret = 0;

//audio stream?
if(input_packet.stream_index == audio_stream_index)
{
avcodec_get_frame_defaults(decoded_frame);
while( input_packet.size > 0 )
{
int result = avcodec_decode_audio4(codec_ctx_audio, decoded_frame, &got_frame, &input_packet);
if ( result < 0)
{
fprintf(stderr, "Error decoding audio frame\n");
//return ret;
}
else
{
if ( got_frame )
{
dst_nb_samples = (int)av_rescale_rnd(swr_get_delay(swrContext, sourceSampleRate) + decoded_frame->nb_samples, sourceSampleRate, destSampleRate, AV_ROUND_UP);
if ( dst_nb_samples > max_dst_nb_samples )
{
max_dst_nb_samples = dst_nb_samples;
if ( audio_dst_data[0] )
{
av_freep(&audio_dst_data[0]);
audio_dst_data[0] = NULL;
}
}
if ( audio_dst_data[0] == NULL )
{
if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
{
ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, codec_ctx_audio->channels,
decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 0);
}
else
{
ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, codec_ctx_audio->channels,
decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_S16, 0);
}
}
/* TODO: extend return code of the av_samples_* functions so that this call is not needed */
int resampled  = swr_convert(swrContext, audio_dst_data, out_frame->nb_samples,
(const uint8_t **)(decoded_frame->extended_data), decoded_frame->nb_samples);
char str[900]="";
sprintf(str,"out_frame->nb_samples:\t%d; decoded_frame->nb_samples:\t%d",out_frame->nb_samples,decoded_frame->nb_samples );
if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
{
audio_dst_bufsize  = av_samples_get_buffer_size(&audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 1);
}
else
{
audio_dst_bufsize  = av_samples_get_buffer_size(&audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_S16, 1);
}

input_packet.size -= result;
input_packet.data += result;
}
else
{
input_packet.size   =   0;
input_packet.data   =   NULL;
}
}
}
}
return rv;
}

int open_encoder(char* output_filename )
{
int rv = 0;

/* allocate the output media context */
AVOutputFormat *opfmt = NULL;

avformat_alloc_output_context2(&output_fmt_ctx, opfmt, NULL, output_filename);
if (!output_fmt_ctx) {
printf("Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&output_fmt_ctx, NULL, "mpeg", output_filename);
}
if (!output_fmt_ctx) {
rv = -1;
}
else
{
output_format = output_fmt_ctx->oformat;
}

/* Add the audio stream using the default format codecs
* and initialize the codecs. */
audio_st = NULL;

if ( output_fmt_ctx )
{
if (output_format->audio_codec != AV_CODEC_ID_NONE)
{
audio_st = add_audio_stream(output_fmt_ctx, &audio_codec, output_format->audio_codec);
}

/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */

if (audio_st)
{
rv = open_audio(output_fmt_ctx, audio_codec, audio_st);
if ( rv < 0 ) return rv;
}

av_dump_format(output_fmt_ctx, 0, output_filename, 1);
/* open the output file, if needed */
if (!(output_format->flags & AVFMT_NOFILE))
{
if (avio_open(&output_fmt_ctx->pb, output_filename, AVIO_FLAG_WRITE) < 0) {
fprintf(stderr, "Could not open '%s'\n", output_filename);
rv = -1;
}
else
{
/* Write the stream header, if any. */
if (avformat_write_header(output_fmt_ctx, NULL) < 0)
{
fprintf(stderr, "Error occurred when opening output file\n");
rv = -1;
}
}
}
}

return rv;
}

AVStream *add_audio_stream(AVFormatContext *oc, AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
AVStream *st;

/* find the audio encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find codec\n");
exit(1);
}

st = avformat_new_stream(oc, *codec);
if (!st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
st->id = 1;

c = st->codec;

/* put sample parameters */
if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
{
c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
}
else
{
c->sample_fmt  = AV_SAMPLE_FMT_S16;
}

c->bit_rate    = audio_bit_rate;
c->sample_rate = audio_sample_rate;
c->channels    = audio_channels;

// some formats want stream headers to be separate
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;

return st;
}

int open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
int ret=0;
AVCodecContext *c;

st->duration = fmt_ctx->duration;
c = st->codec;

/* open it */
ret = avcodec_open2(c, codec, NULL) ;
if ( ret < 0)
{
fprintf(stderr, "could not open codec\n");
return -1;
//exit(1);
}

if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
audio_input_frame_size = 10000;
else
audio_input_frame_size = c->frame_size;
out_frame->nb_samples = audio_input_frame_size;
int tempSize = audio_input_frame_size *
av_get_bytes_per_sample(c->sample_fmt) *
c->channels;
return ret;
}

void close_audio(AVFormatContext *oc, AVStream *st)
{
avcodec_close(st->codec);
}

void write_audio_frame(uint8_t ** audio_dst_data, int audio_dst_bufsize)
{
AVFormatContext *oc = output_fmt_ctx;
AVStream *st = audio_st;
if ( oc == NULL || st == NULL ) return;
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
int got_packet=0, ret=0;

av_init_packet(&pkt);
c = st->codec;

out_frame->nb_samples = audio_input_frame_size;

AVRational r;
r.num = 1;
r.den = c->sample_rate;
out_frame->pts = av_rescale_q(samples_count, (AVRational)r, c->time_base);
avcodec_fill_audio_frame(out_frame, c->channels, c->sample_fmt,
audio_dst_data[0], audio_dst_bufsize, 0);
samples_count += out_frame->nb_samples;

ret = avcodec_encode_audio2(c, &pkt, out_frame, &got_packet);
if (ret < 0)
{
return;
}

if (!got_packet)
return;

/* rescale output packet timestamp values from codec to stream timebase */
pkt.pts = av_rescale_q_rnd(pkt.pts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
pkt.dts = av_rescale_q_rnd(pkt.dts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
pkt.duration = av_rescale_q(pkt.duration, c->time_base, st->time_base);
pkt.stream_index = st->index;

char str[999]="";
sprintf(str,"out_frame->nb_samples:\t%d",out_frame->nb_samples);
/* Write the compressed frame to the media file. */
ret = av_interleaved_write_frame(oc, &pkt);
if (ret != 0)
{
exit(1);
}
av_free_packet(&pkt);
}

void write_delayed_frames(AVFormatContext *oc, AVStream *st)
{
AVCodecContext *c = st->codec;
int got_output = 0;
int ret = 0;
AVPacket pkt;
pkt.data = NULL;
pkt.size = 0;
av_init_packet(&pkt);
int i = 0;
for (got_output = 1; got_output; i++)
{
ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
if (ret < 0)
{
fprintf(stderr, "error encoding frame\n");
exit(1);
}
static int64_t tempPts = 0;
static int64_t tempDts = 0;
/* If size is zero, it means the image was buffered. */
if (got_output)
{
pkt.pts = av_rescale_q_rnd(pkt.pts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
pkt.dts = av_rescale_q_rnd(pkt.dts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
pkt.duration = av_rescale_q(pkt.duration, c->time_base, st->time_base);
pkt.stream_index = st->index;
if ( c && c->coded_frame && c->coded_frame->key_frame)
pkt.flags |= AV_PKT_FLAG_KEY;
/* Write the compressed frame to the media file. */
ret = av_interleaved_write_frame(oc, &pkt);
}
else
{
ret = 0;
}
av_free_packet(&pkt);
}
}

int main(int argc, char **argv)
{
/* register all formats and codecs */
av_register_all();
avcodec_register_all();
avformat_network_init();
avdevice_register_all();
int i =0;
int ret=0;
char src_filename[90] = "test.mp2";
char dst_filename[90] = "output.webm";
outputAudioFormat = AV_CODEC_ID_VORBIS;
open_audio_input(src_filename);
if ( codec_ctx_audio->bit_rate == 0 ) codec_ctx_audio->bit_rate = 112000;
audio_bit_rate        = codec_ctx_audio->bit_rate;
audio_sample_rate    = codec_ctx_audio->sample_rate;
audio_channels        = codec_ctx_audio->channels;
open_encoder( dst_filename );
int frames= 0;
while(1)
{
int rv = decode_frame();
if ( rv < 0 )
{
break;
}

if (audio_st)
{
audio_pts = audio_st->pts.val * av_q2d(audio_st->time_base);
}
else
{
audio_pts = 0.0;
}
if ( codec_ctx_audio )
{
if ( got_frame )
{
write_audio_frame( audio_dst_data, audio_dst_bufsize );
frames++;
}
}
printf("\naudio_pts: %f", audio_pts);
}
while(1)
{
dst_nb_samples = (int)av_rescale_rnd(swr_get_delay(swrContext, sourceSampleRate) + decoded_frame->nb_samples, sourceSampleRate, destSampleRate, AV_ROUND_UP);
if ( dst_nb_samples > max_dst_nb_samples )
{
max_dst_nb_samples = dst_nb_samples;
if ( audio_dst_data[0] )
{
av_freep(&audio_dst_data[0]);
audio_dst_data[0] = NULL;
}
}
if ( audio_dst_data[0] == NULL )
{
if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
{
ret = av_samples_alloc(audio_dst_data, NULL, codec_ctx_audio->channels,
decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 0);
}
else
{
ret = av_samples_alloc(audio_dst_data, NULL, codec_ctx_audio->channels,
decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_S16, 0);
}
}
int resampled = swr_convert(swrContext, audio_dst_data, out_frame->nb_samples,NULL, 0);
if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
{
audio_dst_bufsize  = av_samples_get_buffer_size(&audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 1);
}
else
{
audio_dst_bufsize  = av_samples_get_buffer_size(&audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_S16, 1);
}
if ( audio_dst_bufsize <= 0 ) break;
audio_pts = audio_st->pts.val * av_q2d(audio_st->time_base);
printf("\naudio_pts: %f", audio_pts);
write_audio_frame( audio_dst_data, audio_dst_bufsize );
}
write_delayed_frames( output_fmt_ctx, audio_st );
av_write_trailer(output_fmt_ctx);
close_audio( output_fmt_ctx, audio_st);
swr_free(&swrContext);
avcodec_free_frame(&out_frame);
getch();
return 0;
}

Работая под Windows 7, Zeranoe FFmpeg 32 битная сборка:

libavutil      52. 62.100 / 52. 62.100
libavcodec     55. 47.101 / 55. 47.101
libavformat    55. 22.103 / 55. 22.103
libavdevice    55.  5.102 / 55.  5.102
libavfilter     4.  1.100 /  4.  1.100
libswscale      2.  5.101 /  2.  5.101
libswresample   0. 17.104 /  0. 17.104
libpostproc    52.  3.100 / 52.  3.100

Может ли кто-нибудь указать на то место, где я мог бы неправильно понимать вещи?

Спасибо за любое руководство заранее!