I'm trying to remake the doc/examples/transcoding.c so that in encodes opus audio. How can I do that?
This is what I have right now:
encoder = avcodec_find_encoder(AV_CODEC_ID_OPUS);
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);
}
enc_ctx->thread_count = 1;
/* enc_ctx->sample_rate = dec_ctx->sample_rate; */
enc_ctx->sample_rate = 48000;
/* enc_ctx->channel_layout = dec_ctx->channel_layout; */
enc_ctx->channel_layout = AV_CH_LAYOUT_MONO;
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};
enc_ctx->bit_rate = 32000;
Should I convert to 48000 Hz and to mono with filters, or encoder will figure it out by itself?
How to copy audio from one packet to another? (It looks like each packet should contain exactly 20ms of audio, that's 960 samples). This page https://ffmpeg.org/doxygen/trunk/structAVFrame.html says something about AVFrame.buf, how to copy all samples from there?
[libopus # 02ec1d60] more samples than frame size (avcodec_encode_audio2)
Related
I’m trying to transcode a video on my iOS app using FFMpeg/LibAv.
What I’m trying to accomplish is to transcode a video in order to resize each frame and possibly lower the bitrate in order to save valuable MB in the device.
The resulting video must be playable on all iPhone5+ devices.
After reading the documentation I found out that:
I do not need to encode/decode the audio stream -> I’ll copy as-is to the output file
I need to encode the video using the h264 codec (LibX264) with a profile supported by iOS (baseline profile with level 3.0 - https://trac.ffmpeg.org/wiki/Encode/H.264#Compatibility)
I’m also setting the picture format to YUV planar since it’s the only one supported by iOS
For the sake of testing I’m not using any filter (I’m using a dummy/passthrough) at all or even trying to lower the bitrate, I’m just trying to decode the video stream and encode it again
Most of the code is based on the transcoding.c and filtering.c available on the FFMpeg examples directory
FFMpeg-wise what I’m trying to achieve with LibAv is:
ffmpeg -i INPUT.MOV -c:v libx264 -preset ultrafast -profile:v baseline -level 3.0 -c:a copy output.MOV
(the resulting file - which can be found below - is playable on QuickTime if it’s generated by FFMpeg through the command line)
The original video was generated with a regular iPhone using iOS 8.2 but the problem is not device specific or iOS specific, it occurs on all videos generated with LibAv.
Although both resulting files are playable by VideoLan (VLC) the one I generated through LibAv is not playable by QuickTime even though I can’t find anything wrong with it.
As you can see below, I create the video stream with the proper video codec on the call to avformat_new_stream:
AVStream *out_stream; // output stream
AVStream *in_stream; // input stream
AVCodecContext *dec_ctx, *enc_ctx; // codec context for the stream
AVCodec *encoder; // codec used
int ret;
unsigned int i;
ofmt_ctx = NULL;
// Allocate an AVFormatContext for an output format. This will be the file header (similar to avformat_open_input but with an zero'ed memory)
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
if (!ofmt_ctx) {
av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
[self errorWith:kErrorCreatingOutputContext and:#"Could not create output context"];
return AVERROR_UNKNOWN;
}
// we must not use the AVCodecContext from the video stream directly! So we have to use avcodec_copy_context() to copy the context to a new location (after allocating memory for it, of course).
// iterate over all input streams
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
in_stream = ifmt_ctx->streams[i]; // input stream
dec_ctx = in_stream->codec; // get the codec context for the decoder
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
// lets use h264
encoder = avcodec_find_encoder(AV_CODEC_ID_H264);
if (!encoder) {
[self errorWith:kErrorCodecNotFound and:#"H264 Codec Not Found"];
return AVERROR_UNKNOWN;
}
out_stream = avformat_new_stream(ofmt_ctx, encoder); // create a new stream with h264 codec
if (!out_stream) {
av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
[self errorWith:kErrorAllocateOutputStream and:#"Failed allocating output stream"];
return AVERROR_UNKNOWN;
}
enc_ctx = out_stream->codec; // pointer to the stream codec context
/* 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->width = dec_ctx->width;
enc_ctx->height = dec_ctx->height;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
enc_ctx->pix_fmt = AV_PIX_FMT_YUV420P;
enc_ctx->time_base = dec_ctx->time_base;
av_opt_set(enc_ctx->priv_data, "preset", "ultrafast", 0);
av_opt_set(enc_ctx->priv_data, "profile", "baseline", 0);
av_opt_set(enc_ctx->priv_data, "level", "3.0", 0);
}
out_stream->time_base = in_stream->time_base;
AVDictionaryEntry *tag = NULL;
while ((tag = av_dict_get(in_stream->metadata, "", tag, AV_DICT_IGNORE_SUFFIX))) {
printf("%s=%s\n", tag->key, tag->value);
char *k = av_strdup(tag->key); // if your strings are already allocated,
char *v = av_strdup(tag->value); // you can avoid copying them like this
av_dict_set(&out_stream->metadata, k, v, 0);
}
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);
[self errorWith:kErrorCantOpenOutputFile and:[NSString stringWithFormat:#"Cannot open video encoder for stream #%u",i]];
return ret;
}
}
else if(dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
// if we cant figure out the stream type, fail
av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
[self errorWith:kErrorUnknownStream and:[NSString stringWithFormat:#"Elementary stream #%d is of unknown type, cannot proceed",i]];
return AVERROR_INVALIDDATA;
}
else {
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
[self errorWith:kErrorAllocateOutputStream and:#"Failed allocating output stream"];
return AVERROR_UNKNOWN;
}
enc_ctx = out_stream->codec;
/* this stream must be remuxed */
// copies ifmt_ctx->streams[i]->codec into ofmt_ctx->streams[i]->codec - Copy the settings of the source AVCodecContext into the destination AVCodecContext.
ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
ifmt_ctx->streams[i]->codec);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
[self errorWith:kErrorCopyStreamFailed and:#"Copying stream context failed"];
return ret;
}
}
// dunno what this is for
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
// Create and initialize a AVIOContext for accessing the
// resource indicated by url.
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);
[self errorWith:kErrorCantOpenOutputFile and:[NSString stringWithFormat:#"Could not open output file '%s'", filename]];
return ret;
}
}
/* init muxer, write output file header */
// Allocate the stream private data and write the stream header to an output media file.
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
[self errorWith:kErrorOutFileCantWriteHeader and:#"Error occurred when opening output file"];
return ret;
}
return 0;
You can find the files here:
Original final: https://www.dropbox.com/s/2jjs1uy2pu2veyy/IMG_5705.MOV?dl=0
File generated with FFMpeg - https://www.dropbox.com/s/9hfmq3fcifgpfqc/local-ffmpeg.MOV?dl=0
File generated by code - https://www.dropbox.com/s/rttvny39rj7ejpf/generated-by-Ze.MOV?dl=0
Thank you so much,
Ze
sI have program, that succefully shows h264 stream using SDL: I'm getting h264 frame, decode it using ffmpeg and draw using SDL.
Also I can write frames to file (using fwrite) and play this file through ffplay.
But I want to mux data to the avi and face some problems in av_write_frame.
Here is my code:
...
/*Initializing format context - outFormatContext is the member of my class*/
AVOutputFormat *outFormat;
outFormat = av_guess_format(NULL,"out.avi",NULL);
outFormat->video_codec = AV_CODEC_ID_H264;
outFormat->audio_codec = AV_CODEC_ID_NONE;
avformat_alloc_output_context2(&outFormatContext, outFormat, NULL, "out.avi");
AVCodec *outCodec;
AVStream *outStream = add_stream(outFormatContext, &outCodec, outFormatContext->oformat->video_codec);
avcodec_open2(outStream->codec, outCodec, NULL);
av_dump_format(outFormatContext, 0, "out.avi", 1);
if (avio_open(&outFormatContext->pb, "out.avi", AVIO_FLAG_WRITE) < 0)
throw Exception("Couldn't open file");
if (avformat_write_header(outFormatContext, NULL) < 0)
throw Exception("Couldn't write to file");
//I don't have exceptions here - so there is 6KB header in out.avi.
...
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))
throw("Could not find encoder");
st = avformat_new_stream(oc, *codec);
if (!st)
throw ("Could not allocate stream");
st->id = oc->nb_streams-1;
c = st->codec;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 1920;
c->height = 1080;
c->pix_fmt = PIX_FMT_YUV420P;
c->flags = 0;
c->time_base.num = 1;
c->time_base.den = 25;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
return st;
}
...
/* Part of decoding loop. There is AVPacket packet - h264 packet;
int ret = av_write_frame(outFormatContext, &packet); //it return -22 code - Invadlid argument;
if (avcodec_decode_video2(pCodecCtx, pFrame, &frameDecoded, &packet) < 0)
return;
if (frameDecoded)
{
//SDL stuff
}
Also i tried to use avcodec_encode_video2 (encode pFrame back to the H264) next to the SDL stuff but encoding is not working - i've got empty packets :( It is the second problem.
Using av_interleaved_write_frame causes acces violation.
Code of the muxing part i copied from ffmpeg muxing example (https://www.ffmpeg.org/doxygen/2.1/doc_2examples_2muxing_8c-example.html)
I would like to develop an application which would be able to convert YUV frames into RGBA frames using the ffmpeg library.
I have begun writing this code:
void Decode::video_encode_example(const char *filename, int codec_id)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int i, ret, x, y, got_output;
FILE *f;
AVFrame *frame;
AVPacket pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
printf("Encode video file %s\n", filename);
/* find the mpeg1 video encoder */
codec = avcodec_find_encoder((enum AVCodecID)codec_id);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(2);
}
/* put sample parameters */
c->bit_rate = 400000;
/* resolution must be a multiple of two */
c->width = 352; // Avant c'était du 352x288
c->height = 288;
/* frames per second */
c->time_base = (AVRational){1,25};
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
c->gop_size = 10;
c->max_b_frames = 1;
printf("Avant\n");
c->pix_fmt = PIX_FMT_RGBA;// Avant c'était AV_PIX_FMT_YUV420P
printf("Après\n");
if (codec_id == AV_CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(3);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(4);
}
frame = avcodec_alloc_frame();// Dans une version plus récente c'est av_frame_alloc
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(5);
}
frame->format = c->pix_fmt;
frame->width = c->width;
frame->height = c->height;
However, each time I run this application, the following error appears in my Linux terminal:
[mpeg2video # 0x10c7040] Specified pix_fmt is not supported
Could you help me please ?
I'm not sure how you believe your code is relevant to your question; your question suggests you'd like to do a pixel format conversion from YUV to RGB, for which you could e.g. use ffmpeg's libswscale. However, your code is creating a MPEG-1/2 encoder object and tries to encode RGB input data into MPEG-1/2. This is not possible, ffmpeg's MPEG-1/2 encoders only support YUV420P. I'm not quite sure what to recommend other than to figure out whether you want to encode MPEG-1/2 video, in which case your input should be YUV420P, not RGBA, or whether you want to do pixel format conversion, in which case you should use libswscale...
I want to realize an application that firstly decode a multi-media file(such as test.mp4 file, video codec id is H264), get a video stream and an audio stream, then make some different in the audio stream, at last encode the video stream(use libx264) and audio stream into a result file(result.mp4). To promote the efficiency, i omitted the decode and encode of video stream, i get the video packet via function "av_read_frame", then output it directly into the result file via function "av_write_frame". But there is no picture in the output file, and the size of output file is fairly small.
I tracked the ffmpeg code and found that in the function "av_write_frame->mov_write_packet->ff_mov_write_packet", it will call function "ff_avc_parse_nal_units" to obtain the size of nal unit, but the return value is very small(such as 208 bytes).
I find that the H264 stream in the MP4 file is not stored in Annex-B format, so it can't find start code(0x000001）， now my problem is how can I change the H264 stream to Annex-B format, and make it work?
I added start code at the beginning of every frame manually, but it still not work.
Anyone can give me any hint?Thanks very much.
Following is the codes similar with my:
// write the stream header, if any
av_write_header(pFormatCtxEnc);
.........
/**
* Init of Encoder and Decoder
*/
bool KeyFlag = false;
bool KeyFlagEx = false;
// Read frames and save frames to disk
int iPts = 1;
av_init_packet(&packet);
while(av_read_frame(pFormatCtxDec, &packet)>=0)
{
if (packet.flags == 1)
KeyFlag = true;
if (!KeyFlag)
continue;
if (m_bStop)
{
break;
}
// Is this a packet from the video stream?
if(packet.stream_index == videoStream)
{
currentframeNum ++;
if (progressCB != NULL && currentframeNum%20 == 0)
{
float fpercent = (float)currentframeNum/frameNum;
progressCB(fpercent,m_pUser);
}
if (currentframeNum >= beginFrame && currentframeNum <= endFrane)
{
if (packet.flags == 1)
KeyFlagEx = true;
if (!KeyFlagEx)
continue;
packet.dts = iPts ++;
av_write_frame(pFormatCtxEnc, &packet);
}
}
// Free the packet that was allocated by av_read_frame
}
// write the trailer, if any
av_write_trailer(pFormatCtxEnc);
/**
* Release of encoder and decoder
*/
return true;
You might try this: libavcodec/h264_mp4toannexb_bsf.c. It converts bitstream without start codes to bitstream with start codes.
Using your source file, does ffmpeg -i src.mp4 -vcodec copy -an dst.mp4 work? Does it work if you add -bsf h264_mp4toannexb? (all using the same version/build of ffmpeg as you are trying to use programmatically of course)
Converting 3gp (amr) to mp3 using ffmpeg api calls
I try to use libavformat (ffmpeg) to build my own function that converts 3gp audio files (recorded with an android mobile device) into mp3 files.
I use av_read_frame() to read a frame from the input file and use avcodec_decode_audio3() to decode the data
into a buffer and use this buffer to encode the data into mp3 with avcodec_encode_audio.
This seems to give me a correct result for converting wav to mp3 and mp3 to wav (Or decode one mp3 and encode to another mp3) but not for amr to mp3.
My resulting mp3 file seems to has the right length but only consists of noise.
In another post I read that amr-decoder does not use the same sample format than mp3 does.
AMR uses FLT and mp3 S16 or S32 und that I have to do resampling.
So I call av_audio_resample_init() and audio_resample for each frame that has been decoded.
But that does not solve my problem completely. Now I can hear my recorded voice and unsterstand what I was saying, but the quality is very low and there is still a lot of noise.
I am not sure if I set the parameters of av_audio_resample correctly, especially the last 4 parameters (I think not) or if I miss something else.
ReSampleContext* reSampleContext = av_audio_resample_init(1, 1, 44100, 8000, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT, 0, 0, 0, 0.0);
while(1)
{
if(av_read_frame(ic, &avpkt) < 0)
{
break;
}
out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
int count;
count = avcodec_decode_audio3(audio_stream->codec, (short *)decodedBuffer, &out_size, &avpkt);
if(count < 0)
{
break;
}
if((audio_resample(reSampleContext, (short *)resampledBuffer, (short *)decodedBuffer, out_size / 4)) < 0)
{
fprintf(stderr, "Error\n");
exit(1);
}
out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
pktOut.size = avcodec_encode_audio(c, outbuf, out_size, (short *)resampledBuffer);
if(c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE)
{
pktOut.pts = av_rescale_q(c->coded_frame->pts, c->time_base, outStream->time_base);
//av_res
}
pktOut.pts = AV_NOPTS_VALUE;
pktOut.dts = AV_NOPTS_VALUE;
pktOut.flags |= AV_PKT_FLAG_KEY;
pktOut.stream_index = audio_stream->index;
pktOut.data = outbuf;
if(av_write_frame(oc, &pktOut) != 0)
{
fprintf(stderr, "Error while writing audio frame\n");
exit(1);
}
}