目录
- 参考
- 示例说明
- AVAudioFifo介绍
- 示例代码
1. 参考
- [1] FFmpeg/doc/examples/transcoding.c
- [2] FFmpeg/doc/examples/transcode_aac.c
2. 示例说明
示例提供了一个“解封装->解码->filtering->编码->封装”的处理流程。
示例修改自[1]，修改的地方：
- [1]中旧的编解码的API替换为新的API。
- 编码之前的音频数据经过AVAudioFifo处理，用于满足音频编码器对frame size的要求。否则音频编码为AAC的时候，会报more samples than frame size的错误。AVAudioFifo提供了一个先入先出的音频缓冲队列。
- 增加编码之后的AVPacket的pts和dts的针对编码器和解码器的time_base不一样的转换，否则编码出来的AVPacket的pts和dts不正确。
示例的流程图如下所示。
FFmpeg_transcoding.png
- 编解码API的详细介绍参见 FFmpeg音频解码。
- 转封装流程的详细介绍参见 FFmpeg转封装(remuxing)。
- 其中filtering流程的介绍见 FFmpeg libavfilter使用示例-处理YUV格式数据。
3. AVAudioFifo介绍
AVAudioFifo提供了一个先入先出的音频缓冲队列。
- 操作在样本级别而不是字节级别。
- 支持多通道的格式，不管是planar还是packed类型。
- 当写入一个已满的buffer时会自动重新分配内存。
AVAudioFifo主要的函数：
- av_audio_fifo_alloc()：根据采样格式、通道数和样本个数创建一个AVAudioFifo。
- av_audio_fifo_realloc()：根据新的样本个数为AVAudioFifo重新分配空间。
- av_audio_fifo_write(): 将数据写入AVAudioFifo。如果可用的空间小于传入nb_samples参数AVAudioFifo将自动重新分配空间，
- av_audio_fifo_size(): 获取当前AVAudioFifo中可供读取的样本数量。
- av_audio_fifo_read()：从AVAudioFifo读取数据。
av_audio_fifo_read()的声明在libavutil/audio_fifo.h，如下所示。
/**
* Read data from an AVAudioFifo.
*
* @see enum AVSampleFormat
* The documentation for AVSampleFormat describes the data layout.
*
* @param af AVAudioFifo to read from
* @param data audio data plane pointers
* @param nb_samples number of samples to read
* @return number of samples actually read, or negative AVERROR code
* on failure. The number of samples actually read will not
* be greater than nb_samples, and will only be less than
* nb_samples if av_audio_fifo_size is less than nb_samples.
*/
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples);
说明：
- data传入指向数据平面的指针，例如数据保存在AVFrame中，则传入AVFrame.data。
4. 示例代码
以下的代码修改自[1]。
/**
* @file
* API example for demuxing, decoding, filtering, encoding and muxing
* @example transcoding.c
*/
#include
#include
#include
#include
#include
#include
#include
#include
static AVFormatContext *ifmt_ctx;
static AVFormatContext *ofmt_ctx;
typedef struct FilteringContext {
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
} FilteringContext;
static FilteringContext *filter_ctx;
typedef struct StreamContext {
AVCodecContext *dec_ctx;
AVCodecContext *enc_ctx;
AVAudioFifo *fifo;
int64_t pts_audio;
} StreamContext;
static StreamContext *stream_ctx;
static int open_input_file(const char *filename)
{
int ret;
unsigned int i;
ifmt_ctx = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
stream_ctx = av_mallocz_array(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
if (!stream_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *stream = ifmt_ctx->streams[i];
AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
AVCodecContext *codec_ctx;
if (!dec) {
av_log(NULL, AV_LOG_ERROR, "Failed to find decoder for stream #%u\n", i);
return AVERROR_DECODER_NOT_FOUND;
}
codec_ctx = avcodec_alloc_context3(dec);
if (!codec_ctx) {
av_log(NULL, AV_LOG_ERROR, "Failed to allocate the decoder context for stream #%u\n", i);
return AVERROR(ENOMEM);
}
ret = avcodec_parameters_to_context(codec_ctx, stream->codecpar);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy decoder parameters to input decoder context "
"for stream #%u\n", i);
return ret;
}
/* Reencode video & audio and remux subtitles etc. */
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO)
codec_ctx->framerate = av_guess_frame_rate(ifmt_ctx, stream, NULL);
/* Open decoder */
ret = avcodec_open2(codec_ctx, dec, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
return ret;
}
}
stream_ctx[i].dec_ctx = codec_ctx;
}
av_dump_format(ifmt_ctx, 0, filename, 0);
return 0;
}
static int open_output_file(const char *filename)
{
AVStream *out_stream;
AVStream *in_stream;
AVCodecContext *dec_ctx, *enc_ctx;
AVCodec *encoder;
int ret;
unsigned int i;
ofmt_ctx = NULL;
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
if (!ofmt_ctx) {
av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
return AVERROR_UNKNOWN;
}
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
return AVERROR_UNKNOWN;
}
in_stream = ifmt_ctx->streams[i];
dec_ctx = stream_ctx[i].dec_ctx;
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* in this example, we choose transcoding to same codec */
encoder = avcodec_find_encoder(dec_ctx->codec_id);
if (!encoder) {
av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
enc_ctx = avcodec_alloc_context3(encoder);
if (!enc_ctx) {
av_log(NULL, AV_LOG_FATAL, "Failed to allocate the encoder context\n");
return AVERROR(ENOMEM);
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
* streams easily using filters */
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
enc_ctx->height = dec_ctx->height;
enc_ctx->width = dec_ctx->width;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
/* take first format from list of supported formats */
if (encoder->pix_fmts)
enc_ctx->pix_fmt = encoder->pix_fmts[0];
else
enc_ctx->pix_fmt = dec_ctx->pix_fmt;
/* video time_base can be set to whatever is handy and supported by encoder */
enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
av_log(NULL, AV_LOG_DEBUG, "enc_ctx->time_base=%d/%d\n", enc_ctx->time_base.num, enc_ctx->time_base.den);
} else {
enc_ctx->sample_rate = dec_ctx->sample_rate;
enc_ctx->channel_layout = dec_ctx->channel_layout;
enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
/* take first format from list of supported formats */
enc_ctx->sample_fmt = encoder->sample_fmts[0];
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
av_log(NULL, AV_LOG_DEBUG, "enc_ctx->time_base=%d/%d, "\
"enc_ctx->sample_fmt=%s, dec_ctx->sample_fmt=%s\n",
enc_ctx->time_base.num, enc_ctx->time_base.den,
av_get_sample_fmt_name(enc_ctx->sample_fmt),
av_get_sample_fmt_name(dec_ctx->sample_fmt));
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/* Third parameter can be used to pass settings to encoder */
ret = avcodec_open2(enc_ctx, encoder, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
return ret;
}
ret = avcodec_parameters_from_context(out_stream->codecpar, enc_ctx);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
return ret;
}
out_stream->time_base = enc_ctx->time_base;
stream_ctx[i].enc_ctx = enc_ctx;
stream_ctx[i].fifo = av_audio_fifo_alloc(enc_ctx->sample_fmt, enc_ctx->channels, enc_ctx->frame_size);
stream_ctx[i].pts_audio = 0;
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
return AVERROR_INVALIDDATA;
} else {
/* if this stream must be remuxed */
ret = avcodec_parameters_copy(out_stream->codecpar, in_stream->codecpar);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying parameters for stream #%u failed\n", i);
return ret;
}
out_stream->time_base = in_stream->time_base;
}
}
av_dump_format(ofmt_ctx, 0, filename, 1);
if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
return ret;
}
}
/* init muxer, write output file header */
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
return ret;
}
return 0;
}
static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
AVCodecContext *enc_ctx, const char *filter_spec)
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = NULL;
const AVFilter *buffersink = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVFilterGraph *filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
buffersrc = avfilter_get_by_name("buffer");
buffersink = avfilter_get_by_name("buffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num,
dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
(uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
buffersrc = avfilter_get_by_name("abuffer");
buffersink = avfilter_get_by_name("abuffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout =
av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt),
dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
(uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
(uint8_t*)&enc_ctx->channel_layout,
sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
(uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
} else {
ret = AVERROR_UNKNOWN;
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if (!outputs->name || !inputs->name) {
ret = AVERROR(ENOMEM);
goto end;
}
if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Fill FilteringContext */
fctx->buffersrc_ctx = buffersrc_ctx;
fctx->buffersink_ctx = buffersink_ctx;
fctx->filter_graph = filter_graph;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
/**
* 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. */
if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
av_log(NULL, AV_LOG_ERROR, "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) {
av_log(NULL, AV_LOG_ERROR, "Could not write data to FIFO\n");
return AVERROR_EXIT;
}
return 0;
}
static int init_filters(void)
{
const char *filter_spec;
unsigned int i;
int ret;
filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
if (!filter_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
filter_ctx[i].buffersrc_ctx = NULL;
filter_ctx[i].buffersink_ctx = NULL;
filter_ctx[i].filter_graph = NULL;
if (!(ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO))
continue;
if (ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO)
filter_spec = "null"; /* passthrough (dummy) filter for video */
else
filter_spec = "anull"; /* passthrough (dummy) filter for audio */
ret = init_filter(&filter_ctx[i], stream_ctx[i].dec_ctx,
stream_ctx[i].enc_ctx, filter_spec);
if (ret)
return ret;
}
return 0;
}
/**
* 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,
int frame_size)
{
int error;
/* Create a new frame to store the audio samples. */
if (!(*frame = av_frame_alloc())) {
av_log(NULL, AV_LOG_ERROR, "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) {
av_log(NULL, AV_LOG_ERROR, "Could not allocate output frame samples (error '%s')\n",
av_err2str(error));
av_frame_free(frame);
return error;
}
return 0;
}
static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index) {
int ret;
int got_frame_local;
AVPacket enc_pkt;
/* encode filtered frame */
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
if (stream_ctx[stream_index].enc_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if (filt_frame) {
filt_frame->pts = stream_ctx[stream_index].pts_audio;
stream_ctx[stream_index].pts_audio += filt_frame->nb_samples;
}
}
ret = avcodec_send_frame(stream_ctx[stream_index].enc_ctx, filt_frame);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error submitting the frame to the encoder, %s\n", av_err2str(ret));
return ret;
}
while (1) {
ret = avcodec_receive_packet(stream_ctx[stream_index].enc_ctx, &enc_pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
av_log(NULL, AV_LOG_INFO, "Error of EAGAIN or EOF\n");
return 0;
} else if (ret < 0){
av_log(NULL, AV_LOG_ERROR, "Error during encoding, %s\n", av_err2str(ret));
return ret;
} else {
/* prepare packet for muxing */
enc_pkt.stream_index = stream_index;
av_packet_rescale_ts(&enc_pkt,
stream_ctx[stream_index].dec_ctx->time_base,
stream_ctx[stream_index].enc_ctx->time_base);
av_packet_rescale_ts(&enc_pkt,
stream_ctx[stream_index].enc_ctx->time_base,
ofmt_ctx->streams[stream_index]->time_base);
AVRational enc_timebase = stream_ctx[stream_index].enc_ctx->time_base;
AVRational ofmt_timebase = ofmt_ctx->streams[stream_index]->time_base;
av_log(NULL, AV_LOG_DEBUG, "Muxing frame, enc_pkt->dts=%ld, enc_pkt->pts=%ld,"\
" enc_ctx->time_base=%d/%d, ofmt_ctx->time_base=%d/%d\n", enc_pkt.dts, enc_pkt.pts,
enc_timebase.num, enc_timebase.den, ofmt_timebase.num, ofmt_timebase.den);
/* mux encoded frame */
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
av_packet_unref(&enc_pkt);
if (ret < 0)
return ret;
}
}
return ret;
}
static int encode_write_frame_fifo(AVFrame *filt_frame, unsigned int stream_index) {
int ret = 0;
if (stream_ctx[stream_index].enc_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
const int output_frame_size = stream_ctx[stream_index].enc_ctx->frame_size;
/* 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. */
add_samples_to_fifo(stream_ctx[stream_index].fifo, filt_frame->data, filt_frame->nb_samples);
int audio_fifo_size = av_audio_fifo_size(stream_ctx[stream_index].fifo);
if (audio_fifo_size < stream_ctx[stream_index].enc_ctx->frame_size) {
/* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer. */
return 0;
}
/* 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(stream_ctx[stream_index].fifo) >= stream_ctx[stream_index].enc_ctx->frame_size) {
/* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file. */
/* 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(stream_ctx[stream_index].fifo), stream_ctx[stream_index].enc_ctx->frame_size);
AVFrame *output_frame;
/* Initialize temporary storage for one output frame. */
if (init_output_frame(&output_frame, stream_ctx[stream_index].enc_ctx, frame_size) < 0) {
av_log(NULL, AV_LOG_ERROR, "init_output_frame failed\n");
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(stream_ctx[stream_index].fifo, (void **)output_frame->data, frame_size) < frame_size) {
av_log(NULL, AV_LOG_ERROR, "Could not read data from FIFO\n");
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
ret = encode_write_frame(output_frame, stream_index);
av_frame_free(&output_frame);
}
} else {
ret = encode_write_frame(filt_frame, stream_index);
}
return ret;
}
static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
{
int ret;
AVFrame *filt_frame;
av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
/* push the decoded frame into the filtergraph */
ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
frame, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
return ret;
}
/* pull filtered frames from the filtergraph */
while (1) {
filt_frame = av_frame_alloc();
if (!filt_frame) {
return AVERROR(ENOMEM);
}
av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
filt_frame);
if (ret < 0) {
/* if no more frames for output - returns AVERROR(EAGAIN)
* if flushed and no more frames for output - returns AVERROR_EOF
* rewrite retcode to 0 to show it as normal procedure completion
*/
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
ret = 0;
goto cleanup;
}
filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame_fifo(filt_frame, stream_index);
if (ret < 0)
goto cleanup;
}
cleanup:
av_frame_free(&filt_frame);
return ret;
}
static int flush_encoder(unsigned int stream_index)
{
int ret;
if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
AV_CODEC_CAP_DELAY))
return 0;
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
ret = encode_write_frame(NULL, stream_index);
return ret;
}
int main(int argc, char **argv)
{
int ret;
//av_log_set_level(AV_LOG_DEBUG);
AVPacket packet = { .data = NULL, .size = 0 };
AVFrame *frame = NULL;
enum AVMediaType type;
unsigned int stream_index;
unsigned int i;
int got_frame;
if (argc != 3) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s 说明：
- 需要h264编码的时候，在FFmpeg编译的时候需要加入编码器的支持。否则会报如下错误。类似，如果需要使用其他第三方编码器的情况，也需要加入对应的支持。编译方式参考FFmpeg编译-Linux平台
[h264_v4l2m2m @ 0x205a940] Could not find a valid device
[h264_v4l2m2m @ 0x205a940] can't configure encoder
Cannot open video encoder for stream #0
Error occurred: Invalid argument