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<h2 class="center">An ffmpeg and SDL Tutorial</h2>
<a href="http://dranger.com/ffmpeg/ffmpegtutorial_all.html#tutorial01.html">Part 1</a><br>
<a href="http://dranger.com/ffmpeg/ffmpegtutorial_all.html#tutorial02.html">Part 2</a><br>
<a href="http://dranger.com/ffmpeg/ffmpegtutorial_all.html#tutorial03.html">Part 3</a><br>
<a href="http://dranger.com/ffmpeg/ffmpegtutorial_all.html#tutorial04.html">Part 4</a><br>
<a href="http://dranger.com/ffmpeg/ffmpegtutorial_all.html#tutorial05.html">Part 5</a><br>
<a href="http://dranger.com/ffmpeg/ffmpegtutorial_all.html#tutorial06.html">Part 6</a><br>
<a href="http://dranger.com/ffmpeg/ffmpegtutorial_all.html#tutorial07.html">Part 7</a><br>
<a href="http://dranger.com/ffmpeg/ffmpegtutorial_all.html#tutorial08.html">Part 8</a><br>
<a href="http://dranger.com/ffmpeg/ffmpegtutorial_all.html#end.html">End</a><br>
<a name="ffmpeg.html"></a>
<p>ffmpeg is a wonderful library for creating video applications or even general purpose utilities. ffmpeg takes care of all the hard work of video processing by doing all the decoding, encoding, muxing and demuxing for you. This can make media applications much simpler to write. It's simple, written in C, fast, and can decode almost any codec you'll find in use today, as well as encode several other formats.
</p>
<p>
The only prolem is that documentation is basically nonexistent. There is a single tutorial that shows the basics of ffmpeg and auto-generated doxygen documents. That's it. So, when I decided to learn about ffmpeg, and in the process about how digital video and audio applications work, I decided to document the process and present it as a tutorial.</p>
<p>There is a sample program that comes with ffmpeg called ffplay. It is a simple C program that implements a complete video player using ffmpeg. This tutorial will begin with an updated version of the original tutorial, written by Martin Bhme (I have <span class="crossout">stolen</span> liberally borrowed from that work), and work from there to developing a working video player, based on Fabrice Bellard's ffplay.c. In each tutorial, I'll introduce a new idea (or two) and explain how we implement it. Each tutorial will have a C file so you can download it, compile it, and follow along at home. The source files will show you how the real program works, how we move all the pieces around, as well as showing you the technical details that are unimportant to the tutorial. By the time we are finished, we will have a working video player written in less than 1000 lines of code!
</p>
<p>In making the player, we will be using SDL to output the audio and video of the media file. SDL is an excellent cross-platform multimedia library that's used in MPEG playback software, emulators, and many video games. You will need to download and install the SDL development libraries for your system in order to compile the programs in this tutorial.
</p>
<p>
This tutorial is meant for people with a decent programming background. At the very least you should know C and have some idea about concepts like queues, mutexes, and so on. You should know some basics about multimedia; things like waveforms and such, but you don't need to know a lot, as I explain a lot of those concepts in this tutorial.
</p>
<p>
There are printable HTML files along the way as well as old school ASCII files. You can also get a tarball of the text files and source code or just the source. You can get a printable page of the full thing in HTML or in text.
</p>
<p><b>UPDATE:</b> I've fixed a code error in Tutorial 7 and 8, as well as adding -lavutil.</p>
<p>
Please feel free to email me with bugs, questions, comments, ideas, features, whatever, at <em>dranger at gmail dot com</em>.
</p>
<p><em><b>>></b> Proceed with the tutorial!</em></p>
<a name="tutorial01.html"></a>
<h2>Tutorial 01: Making Screencaps</h2>
<span class="codelink">Code: tutorial01.c</span>
<h3>Overview</h3>
<p>
Movie files have a few basic components. First, the file itself is called a <b>container</b>, and the type of container determines where the information in the file goes. Examples of containers are AVI and Quicktime. Next, you have a bunch of <b>streams</b>; for example, you usually have an audio stream and a video stream. (A "stream" is just a fancy word for "a succession of data elements made available over time".) The data elements in a stream are called <b>frames</b>. Each stream is encoded by a different kind of <b>codec</b>. The codec defines how the actual data is COded and DECoded - hence the name CODEC. Examples of codecs are DivX and MP3. <b>Packets</b> are then read from the stream. Packets are pieces of data that can contain bits of data that are decoded into raw frames that we can finally manipulate for our application. For our purposes, each packet contains complete frames, or multiple frames in the case of audio.
</p>
<p>
At its very basic level, dealing with video and audio streams is very easy:
</p><pre>10 OPEN video_stream FROM video.avi
20 READ packet FROM video_stream INTO frame
30 IF frame NOT COMPLETE GOTO 20
40 DO SOMETHING WITH frame
50 GOTO 20
</pre>
Handling multimedia with ffmpeg is pretty much as simple as this program, although some programs might have a very complex "DO SOMETHING" step. So in this tutorial, we're going to open a file, read from the video stream inside it, and our DO SOMETHING is going to be writing the frame to a PPM file.
<p></p>
<h3>Opening the File</h3>
<p>
First, let's see how we open a file in the first place. With ffmpeg, you have to first initialize the library. (Note that some systems might have to use <ffmpeg/avcodec.h> and <ffmpeg/avformat.h> instead.)
</p><pre>#include <avcodec.h>
#include <avformat.h>
...
int main(int argc, charg *argv[]) {
av_register_all();
</pre>
This registers all available file formats and codecs with the library so they will be used automatically when a file with the corresponding format/codec is opened. Note that you only need to call av_register_all() once, so we do it here in main(). If you like, it's possible to register only certain individual file formats and codecs, but there's usually no reason why you would have to do that.
<p></p>
<p>
Now we can actually open the file:
</p><pre>AVFormatContext *pFormatCtx;
// Open video file
if(av_open_input_file(&pFormatCtx, argv[1], NULL, 0, NULL)!=0)
return -1; // Couldn't open file
</pre>
We get our filename from the first argument. This function reads the file header and stores information about the file format in the AVFormatContext structure we have given it. The last three arguments are used to specify the file format, buffer size, and format options, but by setting this to NULL or 0, libavformat will auto-detect these.
<p>This function only looks at the header, so next we need to check out the stream information in the file.:
</p><pre>// Retrieve stream information
if(av_find_stream_info(pFormatCtx)<0)
return -1; // Couldn't find stream information
</pre>
This function populates <tt>pFormatCtx->streams</tt> with the proper information. We introduce a handy debugging function to show us what's inside:
<pre>// Dump information about file onto standard error
dump_format(pFormatCtx, 0, argv[1], 0);
</pre>
Now <tt>pFormatCtx->streams</tt> is just an array of pointers, of size <tt>pFormatCtx->nb_streams</tt>, so let's walk through it until we find a video stream.
<pre>int i;
AVCodecContext *pCodecCtx;
// Find the first video stream
videoStream=-1;
for(i=0; i<pFormatCtx->nb_streams; i++)
if(pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_VIDEO) {
videoStream=i;
break;
}
if(videoStream==-1)
return -1; // Didn't find a video stream
// Get a pointer to the codec context for the video stream
pCodecCtx=pFormatCtx->streams[videoStream]->codec;
</pre>
The stream's information about the codec is in what we call the "codec context." This contains all the information about the codec that the stream is using, and now we have a pointer to it. But we still have to find the actual codec and open it:
<pre>AVCodec *pCodec;
// Find the decoder for the video stream
pCodec=avcodec_find_decoder(pCodecCtx->codec_id);
if(pCodec==NULL) {
fprintf(stderr, "Unsupported codec!\n");
return -1; // Codec not found
}
// Open codec
if(avcodec_open(pCodecCtx, pCodec)<0)
return -1; // Could not open codec
</pre>
Some of you might remember from the old tutorial that there were two other parts to this code: adding <tt>CODEC_FLAG_TRUNCATED</tt> to <tt>pCodecCtx->flags</tt> and adding a hack to correct grossly incorrect frame rates. These two fixes aren't in ffplay.c anymore, so I have to assume that they are not necessary anymore. There's another difference to point out since we removed that code: <tt>pCodecCtx->time_base</tt> now holds the frame rate information. <tt>time_base</tt> is a struct that has the numerator and denominator (AVRational). We represent the frame rate as a fraction because many codecs have non-integer frame rates (like NTSC's 29.97fps).
<p></p>
<h3>Storing the Data</h3>
<p>Now we need a place to actually store the frame:
</p><pre>AVFrame *pFrame;
// Allocate video frame
pFrame=avcodec_alloc_frame();
</pre>
Since we're planning to output PPM files, which are stored in 24-bit RGB, we're going to have to convert our frame from its native format to RGB. ffmpeg will do these conversions for us. For most projects (including ours) we're going to want to convert our initial frame to a specific format. Let's allocate a frame for the converted frame now.
<pre>// Allocate an AVFrame structure
pFrameRGB=avcodec_alloc_frame();
if(pFrameRGB==NULL)
return -1;
</pre>
Even though we've allocated the frame, we still need a place to put the raw data when we convert it. We use <tt>avpicture_get_size</tt> to get the size we need, and allocate the space manually:
<pre>uint8_t *buffer;
int numBytes;
// Determine required buffer size and allocate buffer
numBytes=avpicture_get_size(PIX_FMT_RGB24, pCodecCtx->width,
pCodecCtx->height);
buffer=(uint8_t *)av_malloc(numBytes*sizeof(uint8_t));
</pre>
<tt>av_malloc</tt> is ffmpeg's malloc that is just a simple wrapper around malloc that makes sure the memory addresses are aligned and such. It will <i>not</i> protect you from memory leaks, double freeing, or other malloc problems.
<p></p>
<p>
Now we use avpicture_fill to associate the frame with our newly allocated buffer. About the AVPicture cast: the AVPicture struct is a subset of the AVFrame struct - the beginning of the AVFrame struct is identical to the AVPicture struct.
</p><pre>// Assign appropriate parts of buffer to image planes in pFrameRGB
// Note that pFrameRGB is an AVFrame, but AVFrame is a superset
// of AVPicture
avpicture_fill((AVPicture *)pFrameRGB, buffer, PIX_FMT_RGB24,
pCodecCtx->width, pCodecCtx->height);
</pre>
Finally! Now we're ready to read from the stream!
<p></p>
<h3>Reading the Data</h3>
<p>
What we're going to do is read through the entire video stream by reading in the packet, decoding it into our frame, and once our frame is complete, we will convert and save it.
</p><pre>int frameFinished;
AVPacket packet;
i=0;
while(av_read_frame(pFormatCtx, &packet)>=0) {
// Is this a packet from the video stream?
if(packet.stream_index==videoStream) {
// Decode video frame
avcodec_decode_video(pCodecCtx, pFrame, &frameFinished,
packet.data, packet.size);
// Did we get a video frame?
if(frameFinished) {
// Convert the image from its native format to RGB
img_convert((AVPicture *)pFrameRGB, PIX_FMT_RGB24,
(AVPicture*)pFrame, pCodecCtx->pix_fmt,
pCodecCtx->width, pCodecCtx->height);
// Save the frame to disk
if(++i<=5)
SaveFrame(pFrameRGB, pCodecCtx->width,
pCodecCtx->height, i);
}
}
// Free the packet that was allocated by av_read_frame
av_free_packet(&packet);
}
</pre>
<span class="sidenote">
<b>A note on packets</b>
<p>
Technically a packet can contain partial frames or other bits of data, but ffmpeg's parser ensures that the packets we get contain either complete or multiple frames.
</p>
</span>
The process, again, is simple: <tt>av_read_frame()</tt> reads in a packet and stores it in the <tt>AVPacket</tt> struct. Note that we've only allocated the packet structure - ffmpeg allocates the internal data for us, which is pointed to by <tt>packet.data</tt>. This is freed by the <tt>av_free_packet()</tt> later. <tt>avcodec_decode_video()</tt> converts the packet to a frame for us. However, we might not have all the information we need for a frame after decoding a packet, so <tt>avcodec_decode_video()</tt> sets frameFinished for us when we have the next frame. Finally, we use <tt>img_convert()</tt> to convert from the native format (<tt>pCodecCtx->pix_fmt</tt>) to RGB. Remember that you can cast an AVFrame pointer to an AVPicture pointer. Finally, we pass the frame and height and width information to our SaveFrame function.
<p></p>
<p>
Now all we need to do is make the SaveFrame function to write the RGB information to a file in PPM format. We're going to be kind of sketchy on the PPM format itself; trust us, it works.
</p><pre>void SaveFrame(AVFrame *pFrame, int width, int height, int iFrame) {
FILE *pFile;
char szFilename[32];
int y;
// Open file
sprintf(szFilename, "frame%d.ppm", iFrame);
pFile=fopen(szFilename, "wb");
if(pFile==NULL)
return;
// Write header
fprintf(pFile, "P6\n%d %d\n255\n", width, height);
// Write pixel data
for(y=0; y<height; y++)
fwrite(pFrame->data[0]+y*pFrame->linesize[0], 1, width*3, pFile);
// Close file
fclose(pFile);
}
</pre>
We do a bit of standard file opening, etc., and then write the RGB data. We write the file one line at a time. A PPM file is simply a file that has RGB information laid out in a long string. If you know HTML colors, it would be like laying out the color of each pixel end to end like <tt>#ff0000#ff0000</tt>.... would be a red screen. (It's stored in binary and without the separator, but you get the idea.) The header indicated how wide and tall the image is, and the max size of the RGB values.
<p></p>
<p>
Now, going back to our main() function. Once we're done reading from the video stream, we just have to clean everything up:
</p><pre>// Free the RGB image
av_free(buffer);
av_free(pFrameRGB);
// Free the YUV frame
av_free(pFrame);
// Close the codec
avcodec_close(pCodecCtx);
// Close the video file
av_close_input_file(pFormatCtx);
return 0;
</pre>
You'll notice we use av_free for the memory we allocated with avcode_alloc_frame and av_malloc.
<p></p>
<p>
That's it for the code! Now, if you're on Linux or a similar platform, you'll run:
</p><pre>gcc -o tutorial01 tutorial01.c -lavutil -lavformat -lavcodec -lz -lavutil -lm
</pre>
If you have an older version of ffmpeg, you may need to drop -lavutil:
<pre>gcc -o tutorial01 tutorial01.c -lavformat -lavcodec -lz -lm
</pre>
Most image programs should be able to open PPM files. Test it on some movie files.
<p></p>
<p>
<em><b>>></b> Tutorial 2: Outputting to the Screen</em>
<a name="tutorial02.html"></a>
</p><h2>Tutorial 02: Outputting to the Screen</h2>
<span class="codelink">Code: tutorial02.c</span>
<h3>SDL and Video</h3>
<p>
To draw to the screen, we're going to use SDL. SDL stands for Simple Direct Layer, and is an excellent library for multimedia, is cross-platform, and is used in several projects. You can get the library at the official website or you can download the development package for your operating system if there is one. You'll need the libraries to compile the code for this tutorial (and for the rest of them, too).
</p>
<p>
SDL has many methods for drawing images to the screen, and it has one in particular that is meant for displaying movies on the screen - what it calls a YUV overlay. YUV (technically not YUV but YCbCr)
<span class="sidenote"><b>* A note: </b>There is a great deal of annoyance from some people at the convention of calling "YCbCr" "YUV". Generally speaking, YUV is an analog format and YCbCr is a digital format. ffmpeg and SDL both refer to YCbCr as YUV in their code and macros.</span>
is a way of storing raw image data like RGB. Roughly speaking, Y is the brightness (or "luma") component, and U and V are the color components. (It's more complicated than RGB because some of the color information is discarded, and you might have only 1 U and V sample for every 2 Y samples.) SDL's YUV overlay takes in a raw array of YUV data and displays it. It accepts 4 different kinds of YUV formats, but YV12 is the fastest. There is another YUV format called YUV420P that is the same as YV12, except the U and V arrays are switched. The 420 means it is subsampled at a ratio of 4:2:0, basically meaning there is 1 color sample for every 4 luma samples, so the color information is quartered. This is a good way of saving bandwidth, as the human eye does not percieve this change. The "P" in the name means that the format is "planar" simply meaning that the Y, U, and V components are in separate arrays. ffmpeg can convert images to YUV420P, with the added bonus that many video streams are in that format already, or are easily converted to that format.
</p>
<p>
So our current plan is to replace the <tt>SaveFrame()</tt> function from Tutorial 1, and instead output our frame to the screen. But first we have to start by seeing how to use the SDL Library. First we have to include the libraries and initalize SDL:
</p><pre>#include <SDL.h>
#include <SDL_thread.h>
if(SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER)) {
fprintf(stderr, "Could not initialize SDL - %s\n", SDL_GetError());
exit(1);
}
</pre>
<tt>SDL_Init()</tt> essentially tells the library what features we're going to use. <tt>SDL_GetError()</tt>, of course, is a handy debugging function.
<p></p>
<h3>Creating a Display</h3>
<p>
Now we need a place on the screen to put stuff. The basic area for displaying images with SDL is called a <b>surface</b>:
</p><pre>SDL_Surface *screen;
screen = SDL_SetVideoMode(pCodecCtx->width, pCodecCtx->height, 0, 0);
if(!screen) {
fprintf(stderr, "SDL: could not set video mode - exiting\n");
exit(1);
}
</pre>
This sets up a screen with the given width and height. The next option is the bit depth of the screen - 0 is a special value that means "same as the current display". (This does not work on OS X; see source.)
<p></p>
<p>
Now we create a YUV overlay on that screen so we can input video to it:
</p><pre>SDL_Overlay *bmp;
bmp = SDL_CreateYUVOverlay(pCodecCtx->width, pCodecCtx->height,
SDL_YV12_OVERLAY, screen);
</pre>
As we said before, we are using YV12 to display the image.
<p></p>
<h3>Displaying the Image</h3>
<p>
Well that was simple enough! Now we just need to display the image. Let's go all the way down to where we had our finished frame. We can get rid of all that stuff we had for the RGB frame, and we're going to replace the <tt>SaveFrame()</tt> with our display code. To display the image, we're going to make an AVPicture struct and set its data pointers and linesize to our YUV overlay:
</p><pre> if(frameFinished) {
SDL_LockYUVOverlay(bmp);
AVPicture pict;
pict.data[0] = bmp->pixels[0];
pict.data[1] = bmp->pixels[2];
pict.data[2] = bmp->pixels[1];
pict.linesize[0] = bmp->pitches[0];
pict.linesize[1] = bmp->pitches[2];
pict.linesize[2] = bmp->pitches[1];
// Convert the image into YUV format that SDL uses
img_convert(&pict, PIX_FMT_YUV420P,
(AVPicture *)pFrame, pCodecCtx->pix_fmt,
pCodecCtx->width, pCodecCtx->height);
SDL_UnlockYUVOverlay(bmp);
}
</pre>
First, we lock the overlay because we are going to be writing to it. This is a good habit to get into so you don't have problems later. The AVPicture struct, as shown before, has a <tt>data</tt> pointer that is an array of 4 pointers. Since we are dealing with YUV420P here, we only have 3 channels, and therefore only 3 sets of data. Other formats might have a fourth pointer for an alpha channel or something. <tt>linesize</tt> is what it sounds like. The analogous structures in our YUV overlay are the <tt>pixels</tt> and <tt>pitches</tt> variables. ("pitches" is the term SDL uses to refer to the width of a given line of data.) So what we do is point the three arrays of <tt>pict.data</tt> at our overlay, so when we write to pict, we're actually writing into our overlay, which of course already has the necessary space allocated. Similarly, we get the linesize information directly from our overlay. We change the conversion format to <tt>PIX_FMT_YUV420P</tt>, and we use <tt>img_convert</tt> just like before.
<p></p>
<h3>Drawing the Image</h3>
<p>
But we still need to tell SDL to actually show the data we've given it. We also pass this function a rectangle that says where the movie should go and what width and height it should be scaled to. This way, SDL does the scaling for us, and it can be assisted by your graphics processor for faster scaling:
</p><pre>SDL_Rect rect;
if(frameFinished) {
/* ... code ... */
// Convert the image into YUV format that SDL uses
img_convert(&pict, PIX_FMT_YUV420P,
(AVPicture *)pFrame, pCodecCtx->pix_fmt,
pCodecCtx->width, pCodecCtx->height);
SDL_UnlockYUVOverlay(bmp);
rect.x = 0;
rect.y = 0;
rect.w = pCodecCtx->width;
rect.h = pCodecCtx->height;
SDL_DisplayYUVOverlay(bmp, &rect);
}
</pre>
Now our video is displayed!
<p></p>
<p>
Let's take this time to show you another feature of SDL: its event system. SDL is set up so that when you type, or move the mouse in the SDL application, or send it a signal, it generates an <b>event</b>. Your program then checks for these events if it wants to handle user input. Your program can also make up events to send the SDL event system. This is especially useful when multithread programming with SDL, which we'll see in Tutorial 4. In our program, we're going to poll for events right after we finish processing a packet. For now, we're just going to handle the <tt>SDL_QUIT</tt> event so we can exit:
</p><pre>SDL_Event event;
av_free_packet(&packet);
SDL_PollEvent(&event);
switch(event.type) {
case SDL_QUIT:
SDL_Quit();
exit(0);
break;
default:
break;
}
</pre>
And there we go! Get rid of all the old cruft, and you're ready to compile. If you are using Linux or a variant, the best way to compile using the SDL libs is this:
<pre>gcc -o tutorial02 tutorial02.c -lavutil -lavformat -lavcodec -lz -lm \
`sdl-config --cflags --libs`
</pre>
sdl-config just prints out the proper flags for gcc to include the SDL libraries properly. You may need to do something different to get it to compile on your system; please check the SDL documentation for your system. Once it compiles, go ahead and run it.
<p></p>
<p>
What happens when you run this program? The video is going crazy! In fact, we're just displaying all the video frames as fast as we can extract them from the movie file. We don't have any code right now for figuring out <i>when</i> we need to display video. Eventually (in Tutorial 5), we'll get around to syncing the video. But first we're missing something even more important: sound!
</p>
<p>
<em><b>>></b> Playing Sound</em>
</p>
<a name="tutorial03.html"></a>
<h2>Tutorial 03: Playing Sound</h2>
<span class="codelink">Code: tutorial03.c</span>
<h3>Audio</h3>
<p>
So now we want to play sound. SDL also gives us methods for outputting sound. The <tt>SDL_OpenAudio()</tt> function is used to open the audio device itself. It takes as arguments an <tt>SDL_AudioSpec</tt> struct, which contains all the information about the audio we are going to output.
</p>
<p>
Before we show how you set this up, let's explain first about how audio is handled by computers. Digital audio consists of a long stream of <b>samples</b>. Each sample represents a value of the audio waveform. Sounds are recorded at a certain <b>sample rate</b>, which simply says how fast to play each sample, and is measured in number of samples per second. Example sample rates are 22,050 and 44,100 samples per second, which are the rates used for radio and CD respectively. In addition, most audio can have more than one channel for stereo or surround, so for example, if the sample is in stereo, the samples will come 2 at a time. When we get data from a movie file, we don't know how many samples we will get, but ffmpeg will not give us partial samples - that also means that it will not split a stereo sample up, either.
</p>
<p>
SDL's method for playing audio is this: you set up your audio options: the sample rate (called "freq" for <b>frequency</b> in the SDL struct), number of channels, and so forth, and we also set a callback function and userdata. When we begin playing audio, SDL will continually call this callback function and ask it to fill the audio buffer with a certain number of bytes. After we put this information in the <tt>SDL_AudioSpec</tt> struct, we call <tt>SDL_OpenAudio()</tt>, which will open the audio device and give us back <i>another</i> AudioSpec struct. These are the specs we will <i>actually</i> be using we are not guaranteed to get what we asked for!
</p>
<h3>Setting Up the Audio</h3>
<p>Keep that all in your head for the moment, because we don't actually have any information yet about the audio streams yet! Let's go back to the place in our code where we found the video stream and find which stream is the audio stream.
</p><pre>// Find the first video stream
videoStream=-1;
audioStream=-1;
for(i=0; i < pFormatCtx->nb_streams; i++) {
if(pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_VIDEO
&&
videoStream < 0) {
videoStream=i;
}
if(pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_AUDIO &&
audioStream < 0) {
audioStream=i;
}
}
if(videoStream==-1)
return -1; // Didn't find a video stream
if(audioStream==-1)
return -1;
</pre>
From here we can get all the info we want from the <tt>AVCodecContext</tt> from the stream, just like we did with the video stream:
<pre>AVCodecContext *aCodecCtx;
aCodecCtx=pFormatCtx->streams[audioStream]->codec;
</pre>
<p></p>
<p>
Contained within this codec context is all the information we need to set up our audio:
</p><pre>wanted_spec.freq = aCodecCtx->sample_rate;
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.channels = aCodecCtx->channels;
wanted_spec.silence = 0;
wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
wanted_spec.callback = audio_callback;
wanted_spec.userdata = aCodecCtx;
if(SDL_OpenAudio(&wanted_spec, &spec) < 0) {
fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError());
return -1;
}
</pre>
Let's go through these:
<ul>
<li><tt>freq</tt>: The sample rate, as explained earlier.</li>
<li><tt>format</tt>: This tells SDL what format we will be giving it. The "S" in "S16SYS" stands for "signed", the 16 says that each sample is 16 bits long, and "SYS" means that the endian-order will depend on the system you are on. This is the format that <tt>avcodec_decode_audio2</tt> will give us the audio in.</li>
<li><tt>channels</tt>: Number of audio channels.</li>
<li><tt>silence</tt>: This is the value that indicated silence. Since the audio is signed, 0 is of course the usual value.</li>
<li><tt>samples</tt>: This is the size of the audio buffer that we would like SDL to give us when it asks for more audio. A good value here is between 512 and 8192; ffplay uses 1024.</li>
<li><tt>callback</tt>: Here's where we pass the actual callback function. We'll talk more about the callback function later.</li>
<li><tt>userdata</tt>: SDL will give our callback a void pointer to any user data that we want our callback function to have. We want to let it know about our codec context; you'll see why.</li>
</ul>
Finally, we open the audio with <tt>SDL_OpenAudio</tt>.
<p></p>
<p>
If you remember from the previous tutorials, we still need to open the audio codec itself. This is straightforward:
</p><pre>AVCodec *aCodec;
aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
if(!aCodec) {
fprintf(stderr, "Unsupported codec!\n");
return -1;
}
avcodec_open(aCodecCtx, aCodec);
</pre>
<p></p>
<h3>Queues</h3>
<p>
There! Now we're ready to start pulling audio information from the stream. But what do we do with that information? We are going to be continuously getting packets from the movie file, but at the same time SDL is going to call the callback function! The solution is going to be to create some kind of global structure that we can stuff audio packets in so our <tt>audio_callback</tt> has something to get audio data from! So what we're going to do is to create a <b>queue</b> of packets. ffmpeg even comes with a structure to help us with this: <tt>AVPacketList</tt>, which is just a linked list for packets. Here's our queue structure:
</p>
<pre>typedef struct PacketQueue {
AVPacketList *first_pkt, *last_pkt;
int nb_packets;
int size;
SDL_mutex *mutex;
SDL_cond *cond;
} PacketQueue;
</pre>
First, we should point out that <tt>nb_packets</tt> is not the same as <tt>size</tt> <tt>size</tt> refers to a byte size that we get from <tt>packet->size</tt>. You'll notice that we have a mutex and a condtion variable in there. This is because SDL is running the audio process as a separate thread. If we don't lock the queue properly, we could really mess up our data. We'll see how in the implementation of the queue. Every programmer should know how to make a queue, but we're including this so you can learn the SDL functions.
<p></p>
<p>
First we make a function to initialize the queue:
</p><pre>void packet_queue_init(PacketQueue *q) {
memset(q, 0, sizeof(PacketQueue));
q->mutex = SDL_CreateMutex();
q->cond = SDL_CreateCond();
}
</pre>
Then we will make a function to put stuff in our queue:
<pre>int packet_queue_put(PacketQueue *q, AVPacket *pkt) {
AVPacketList *pkt1;
if(av_dup_packet(pkt) < 0) {
return -1;
}
pkt1 = av_malloc(sizeof(AVPacketList));
if (!pkt1)
return -1;
pkt1->pkt = *pkt;
pkt1->next = NULL;
SDL_LockMutex(q->mutex);
if (!q->last_pkt)
q->first_pkt = pkt1;
else
q->last_pkt->next = pkt1;
q->last_pkt = pkt1;
q->nb_packets++;
q->size += pkt1->pkt.size;
SDL_CondSignal(q->cond);
SDL_UnlockMutex(q->mutex);
return 0;
}
</pre>
<tt>SDL_LockMutex()</tt> locks the mutex in the queue so we can add something to it, and then <tt>SDL_CondSignal()</tt> sends a signal to our get function (if it is waiting) through our condition variable to tell it that there is data and it can proceed, then unlocks the mutex to let it go.
<p></p>
<p>
Here's the corresponding get function. Notice how <tt>SDL_CondWait()</tt> makes the function <b>block</b> (i.e. pause until we get data) if we tell it to.
</p><pre>int quit = 0;
static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block) {
AVPacketList *pkt1;
int ret;
SDL_LockMutex(q->mutex);
for(;;) {
if(quit) {
ret = -1;
break;
}
pkt1 = q->first_pkt;
if (pkt1) {
q->first_pkt = pkt1->next;
if (!q->first_pkt)
q->last_pkt = NULL;
q->nb_packets--;
q->size -= pkt1->pkt.size;
*pkt = pkt1->pkt;
av_free(pkt1);
ret = 1;
break;
} else if (!block) {
ret = 0;
break;
} else {
SDL_CondWait(q->cond, q->mutex);
}
}
SDL_UnlockMutex(q->mutex);
return ret;
}
</pre>
As you can see, we've wrapped the function in a forever loop so we will be sure to get some data if we want to block. We avoid looping forever by making use of SDL's <t>SDL_CondWait() function. Basically, all CondWait does is wait for a signal from <tt>SDL_CondSignal()</tt> (or <tt>SDL_CondBroadcast()</tt>) and then continue. However, it looks as though we've trapped it within our mutex if we hold the lock, our put function can't put anything in the queue! However, what <tt>SDL_CondWait()</tt> also does for us is to unlock the mutex we give it and then attempt to lock it again once we get the signal.
<p></p>
<h3>In Case of Fire</h3>
<p>
You'll also notice that we have a global <tt>quit</tt> variable that we check to make sure that we haven't set the program a quit signal (SDL automatically handles TERM signals and the like). Otherwise, the thread will continue forever and we'll have to <tt>kill -9</tt> the program. ffmpeg also has a function for providing a callback to check and see if we need to quit some blocking function: <tt>url_set_interrupt_cb</tt>.
</p><pre>int decode_interrupt_cb(void) {
return quit;
}
...
main() {
...
url_set_interrupt_cb(decode_interrupt_cb);
...
SDL_PollEvent(&event);
switch(event.type) {
case SDL_QUIT:
quit = 1;
...
</pre>
This only applies for ffmpeg functions that block, of course, not SDL ones. We make sure to set the <tt>quit</tt> flag to 1.
<p></p>
<h3>Feeding Packets</h3>
<p>
The only thing left is to set up our queue:
</p><pre>PacketQueue audioq;
main() {
...
avcodec_open(aCodecCtx, aCodec);
packet_queue_init(&audioq);
SDL_PauseAudio(0);
</pre>
<tt>SDL_PauseAudio()</tt> finally starts the audio device. It plays silence if it doesn't get data; which it won't right away.
<p></p>
<p>
So, we've got our queue set up, now we're ready to start feeding it packets. We go to our packet-reading loop:
</p><pre>while(av_read_frame(pFormatCtx, &packet)>=0) {
// Is this a packet from the video stream?
if(packet.stream_index==videoStream) {
// Decode video frame
....
}
} else if(packet.stream_index==audioStream) {
packet_queue_put(&audioq, &packet);
} else {
av_free_packet(&packet);
}
</pre>
Note that we don't free the packet after we put it in the queue. We'll free it later when we decode it.
<p></p>
<h3>Fetching Packets</h3>
<p>
Now let's finally make our <tt>audio_callback</tt> function to fetch the packets on the queue. The callback has to be of the form <tt>void callback(void *userdata, Uint8 *stream, int len)</tt>, where <tt>userdata</tt> of course is the pointer we gave to SDL, <tt>stream</tt> is the buffer we will be writing audio data to, and <tt>len</tt> is the size of that buffer. Here's the code:
</p><pre>void audio_callback(void *userdata, Uint8 *stream, int len) {
AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
int len1, audio_size;
static uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = 0;
static unsigned int audio_buf_index = 0;
while(len > 0) {
if(audio_buf_index >= audio_buf_size) {
/* We have already sent all our data; get more */
audio_size = audio_decode_frame(aCodecCtx, audio_buf,
sizeof(audio_buf));
if(audio_size < 0) {
/* If error, output silence */
audio_buf_size = 1024;
memset(audio_buf, 0, audio_buf_size);
} else {
audio_buf_size = audio_size;
}
audio_buf_index = 0;
}
len1 = audio_buf_size - audio_buf_index;
if(len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
len -= len1;
stream += len1;
audio_buf_index += len1;
}
}
</pre>
This is basically a simple loop that will pull in data from another function we will write, <tt>audio_decode_frame()</tt>, store the result in an intermediary buffer, attempt to write <tt>len</tt> bytes to <tt>stream</tt>, and get more data if we don't have enough yet, or save it for later if we have some left over. The size of <tt>audio_buf</tt> is 1.5 times the size of the largest audio frame that ffmpeg will give us, which gives us a nice cushion.
<p></p>
<h3>Finally Decoding the Audio</h3>
<p>
Let's get to the real meat of the decoder, <tt>audio_decode_frame</tt>:
</p><pre>int audio_decode_frame(AVCodecContext *aCodecCtx, uint8_t *audio_buf,
int buf_size) {
static AVPacket pkt;
static uint8_t *audio_pkt_data = NULL;
static int audio_pkt_size = 0;
int len1, data_size;
for(;;) {
while(audio_pkt_size > 0) {
data_size = buf_size;
len1 = avcodec_decode_audio2(aCodecCtx, (int16_t *)audio_buf, &data_size,
audio_pkt_data, audio_pkt_size);
if(len1 < 0) {
/* if error, skip frame */
audio_pkt_size = 0;
break;
}
audio_pkt_data += len1;
audio_pkt_size -= len1;
if(data_size <= 0) {
/* No data yet, get more frames */
continue;
}
/* We have data, return it and come back for more later */
return data_size;
}
if(pkt.data)
av_free_packet(&pkt);
if(quit) {
return -1;
}
if(packet_queue_get(&audioq, &pkt, 1) < 0) {
return -1;
}
audio_pkt_data = pkt.data;
audio_pkt_size = pkt.size;
}
}
</pre>
This whole process actually starts towards the end of the function, where we call <tt>packet_queue_get()</tt>. We pick the packet up off the queue, and save its information. Then, once we have a packet to work with, we call <tt>avcodec_decode_audio2()</tt>, which acts a lot like its sister function, <tt>avcodec_decode_video()</tt>, except in this case, a packet might have more than one frame, so you may have to call it several times to get all the data out of the packet.
<span class="sidenote"><b>* A note: </b>Why <tt>avcodec_decode_audio<b>2</b></tt>? Because there used to be an <tt>avcodec_decode_audio</tt>, but it's now deprecated. The new function reads from the <tt>data_size</tt> variable to figure out how big <tt>audio_buf</tt> is.</span>
Also, remember the cast to audio_buf, because SDL gives an 8 bit int buffer, and ffmpeg gives us data in a 16 bit int buffer. You should also notice the difference between <tt>len1</tt> and <tt>data_size</tt>. <tt>len1</tt> is how much of the packet we've used, and <tt>data_size</tt> is the amount of raw data returned.
<p></p>
<p>
When we've got some data, we immediately return to see if we still need to get more data from the queue, or if we are done. If we still had more of the packet to process, we save it for later. If we finish up a packet, we finally get to free that packet.
</p>
<p>
So that's it! We've got audio being carried from the main read loop to the queue, which is then read by the <tt>audio_callback</tt> function, which hands that data to SDL, which SDL beams to your sound card. Go ahead and compile:
</p><pre>gcc -o tutorial03 tutorial03.c -lavutil -lavformat -lavcodec -lz -lm \
`sdl-config --cflags --libs`
</pre>
Hooray! The video is still going as fast as possible, but the audio is playing in time. Why is this? That's because the audio information has a sample rate we're pumping out audio information as fast as we can, but the audio simply plays from that stream at its leisure according to the sample rate.
<p></p>
<p>
We're almost ready to start syncing video and audio ourselves, but first we need to do a little program reorganization. The method of queueing up audio and playing it using a separate thread worked very well: it made the code more managable and more modular. Before we start syncing the video to the audio, we need to make our code easier to deal with. Next time: Spawning Threads!
</p>
<p>
<em><b>>></b> Spawning Threads</em>
</p>
<a name="tutorial04.html"></a>
<h2>Tutorial 04: Spawning Threads</h2>
<span class="codelink">Code: tutorial04.c</span>
<h3>Overview</h3>
<p>
Last time we added audio support by taking advantage of SDL's audio functions. SDL started a thread that made callbacks to a function we defined every time it needed audio. Now we're going to do the same sort of thing with the video display. This makes the code more modular and easier to work with - especially when we want to add syncing. So where do we start?
</p>
<p>
First we notice that our main function is handling an awful lot: it's running through the event loop, reading in packets, and decoding the video. So what we're going to do is split all those apart: we're going to have a thread that will be responsible for decoding the packets; these packets will then be added to the queue and read by the corresponding audio and video threads. The audio thread we have already set up the way we want it; the video thread will be a little more complicated since we have to display the video ourselves. We will add the actual display code to the main loop. But instead of just displaying video every time we loop, we will integrate the video display into the event loop. The idea is to decode the video, save the resulting frame in <i>another</i> queue, then create a custom event (<tt>FF_REFRESH_EVENT</tt>) that we add to the event system, then when our event loop sees this event, it will display the next frame in the queue. Here's a handy ASCII art illustration of what is going on:
</p><pre> ________ audio _______ _____
| | pkts | | | | to spkr
| DECODE |----->| AUDIO |--->| SDL |-->
|________| |_______| |_____|
| video _______
| pkts | |
+---------->| VIDEO |
________ |_______| _______
| | | | |
| EVENT | +------>| VIDEO | to mon.
| LOOP |----------------->| DISP. |-->
|_______|<---FF_REFRESH----|_______|
</pre>
The main purpose of moving controlling the video display via the event loop is that using an SDL_Delay thread, we can control exactly when the next video frame shows up on the screen. When we finally sync the video in the next tutorial, it will be a simple matter to add the code that will schedule the next video refresh so the right picture is being shown on the screen at the right time.
<p></p>
<h3>Simplifying Code</h3>
<p>
We're also going to clean up the code a bit. We have all this audio and video codec information, and we're going to be adding queues and buffers and who knows what else. All this stuff is for one logical unit, <i>viz.</i> the movie. So we're going to make a large struct that will hold all that information called the <tt>VideoState</tt>.
</p><pre>typedef struct VideoState {
AVFormatContext *pFormatCtx;
int videoStream, audioStream;
AVStream *audio_st;
PacketQueue audioq;
uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
unsigned int audio_buf_size;
unsigned int audio_buf_index;
AVPacket audio_pkt;
uint8_t *audio_pkt_data;
int audio_pkt_size;
AVStream *video_st;
PacketQueue videoq;
VideoPicture pictq[VIDEO_PICTURE_QUEUE_SIZE];
int pictq_size, pictq_rindex, pictq_windex;
SDL_mutex *pictq_mutex;
SDL_cond *pictq_cond;
SDL_Thread *parse_tid;
SDL_Thread *video_tid;
char filename[1024];
int quit;
} VideoState;
</pre>
Here we see a glimpse of what we're going to get to. First we see the basic information - the format context and the indices of the audio and video stream, and the corresponding AVStream objects. Then we can see that we've moved some of those audio buffers into this structure. These (audio_buf, audio_buf_size, etc.) were all for information about audio that was still lying around (or the lack thereof). We've added another queue for the video, and a buffer (which will be used as a queue; we don't need any fancy queueing stuff for this) for the decoded frames (saved as an overlay). The VideoPicture struct is of our own creations (we'll see what's in it when we come to it). We also notice that we've allocated pointers for the two extra threads we will create, and the quit flag and the filename of the movie.
<p></p>
<p>
So now we take it all the way back to the main function to see how this changes our program. Let's set up our <tt>VideoState</tt> struct:
</p><pre>int main(int argc, char *argv[]) {
SDL_Event event;
VideoState *is;
is = av_mallocz(sizeof(VideoState));
</pre>
<tt>av_mallocz()</tt> is a nice function that will allocate memory for us and zero it out.
<p></p>
<p>
Then we'll initialize our locks for the display buffer (<tt>pictq</tt>), because since the event loop calls our display function - the display function, remember, will be pulling pre-decoded frames from <tt>pictq</tt>. At the same time, our video decoder will be putting information into it - we don't know who will get there first. Hopefully you recognize that this is a classic <b>race condition</b>. So we allocate it now before we start any threads. Let's also copy the filename of our movie into our <tt>VideoState</tt>.
</p><pre>pstrcpy(is->filename, sizeof(is->filename), argv[1]);
is->pictq_mutex = SDL_CreateMutex();
is->pictq_cond = SDL_CreateCond();
</pre>
<tt>pstrcpy</tt> is a function from ffmpeg that does some extra bounds checking beyond strncpy.
<p></p>
<h3>Our First Thread</h3>
<p>
Now let's finally launch our threads and get the real work done:
</p><pre>schedule_refresh(is, 40);
is->parse_tid = SDL_CreateThread(decode_thread, is);
if(!is->parse_tid) {
av_free(is);
return -1;
}
</pre>
<tt>schedule_refresh</tt> is a function we will define later. What it basically does is tell the system to push a <tt>FF_REFRESH_EVENT</tt> after the specified number of milliseconds. This will in turn call the video refresh function when we see it in the event queue. But for now, let's look at <tt>SDL_CreateThread()</tt>.
<p></p>
<p>
<tt>SDL_CreateThread()</tt> does just that - it spawns a new thread that has complete access to all the memory of the original process, and starts the thread running on the function we give it. It will also pass that function user-defined data. In this case, we're calling <tt>decode_thread()</tt> and with our <tt>VideoState</tt> struct attached. The first half of the function has nothing new; it simply does the work of opening the file and finding the index of the audio and video streams. The only thing we do different is save the format context in our big struct. After we've found our stream indices, we call another function that we will define, <tt>stream_component_open()</tt>. This is a pretty natural way to split things up, and since we do a lot of similar things to set up the video and audio codec, we reuse some code by making this a function.
</p>
<p>
The <tt>stream_component_open()</tt> function is where we will find our codec decoder, set up our audio options, save important information to our big struct, and launch our audio and video threads. This is where we would also insert other options, such as forcing the codec instead of autodetecting it and so forth. Here it is:
</p><pre>int stream_component_open(VideoState *is, int stream_index) {
AVFormatContext *pFormatCtx = is->pFormatCtx;
AVCodecContext *codecCtx;
AVCodec *codec;
SDL_AudioSpec wanted_spec, spec;
if(stream_index < 0 || stream_index >= pFormatCtx->nb_streams) {
return -1;
}
// Get a pointer to the codec context for the video stream
codecCtx = pFormatCtx->streams[stream_index]->codec;
if(codecCtx->codec_type == CODEC_TYPE_AUDIO) {
// Set audio settings from codec info
wanted_spec.freq = codecCtx->sample_rate;
/* .... */
wanted_spec.callback = audio_callback;
wanted_spec.userdata = is;
if(SDL_OpenAudio(&wanted_spec, &spec) < 0) {
fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError());
return -1;
}
}
codec = avcodec_find_decoder(codecCtx->codec_id);
if(!codec || (avcodec_open(codecCtx, codec) < 0)) {
fprintf(stderr, "Unsupported codec!\n");
return -1;
}
switch(codecCtx->codec_type) {
case CODEC_TYPE_AUDIO:
is->audioStream = stream_index;
is->audio_st = pFormatCtx->streams[stream_index];
is->audio_buf_size = 0;
is->audio_buf_index = 0;
memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
packet_queue_init(&is->audioq);
SDL_PauseAudio(0);
break;
case CODEC_TYPE_VIDEO:
is->videoStream = stream_index;
is->video_st = pFormatCtx->streams[stream_index];
packet_queue_init(&is->videoq);
is->video_tid = SDL_CreateThread(video_thread, is);
break;
default:
break;
}
}
</pre>
This is pretty much the same as the code we had before, except now it's generalized for audio and video. Notice that instead of aCodecCtx, we've set up our big struct as the userdata for our audio callback. We've also saved the streams themselves as <tt>audio_st</tt> and <tt>video_st</tt>. We also have added our video queue and set it up in the same way we set up our audio queue. Most of the point is to launch the video and audio threads. These bits do it:
<pre> SDL_PauseAudio(0);
break;
/* ...... */
is->video_tid = SDL_CreateThread(video_thread, is);
</pre>
We remember <tt>SDL_PauseAudio()</tt> from last time, and <tt>SDL_CreateThread()</tt> is used as in the exact same way as before. We'll get back to our <tt>video_thread()</tt> function.
<p></p>
<p>
Before that, let's go back to the second half of our <tt>decode_thread()</tt> function. It's basically just a for loop that will read in a packet and put it on the right queue:
</p><pre> for(;;) {
if(is->quit) {
break;
}
// seek stuff goes here
if(is->audioq.size > MAX_AUDIOQ_SIZE ||
is->videoq.size > MAX_VIDEOQ_SIZE) {
SDL_Delay(10);
continue;
}
if(av_read_frame(is->pFormatCtx, packet) < 0) {
if(url_ferror(&pFormatCtx->pb) == 0) {
SDL_Delay(100); /* no error; wait for user input */
continue;
} else {
break;
}
}
// Is this a packet from the video stream?
if(packet->stream_index == is->videoStream) {
packet_queue_put(&is->videoq, packet);
} else if(packet->stream_index == is->audioStream) {
packet_queue_put(&is->audioq, packet);
} else {
av_free_packet(packet);
}
}
</pre>
Nothing really new here, except that we now have a max size for our audio and video queue, and we've added a function that will check for read errors. The format context has a <tt>ByteIOContext</tt> struct inside it called <tt>pb</tt>. <tt>ByteIOContext</tt> is the structure that basically keeps all the low-level file information in it. <tt>url_ferror</tt> checks that structure to see if there was some kind of error reading from our file.
<p></p>
<p>
After our for loop, we have all the code for waiting for the rest of the program to end or informing it that we've ended. This code is instructive because it shows us how we push events - something we'll have to later to display the video.
</p><pre> while(!is->quit) {
SDL_Delay(100);
}
fail:
if(1){
SDL_Event event;
event.type = FF_QUIT_EVENT;
event.user.data1 = is;
SDL_PushEvent(&event);
}
return 0;
</pre>
We get values for user events by using the SDL constant <tt>SDL_USEREVENT</tt>. The first user event should be assigned the value <tt>SDL_USEREVENT</tt>, the next <tt>SDL_USEREVENT + 1</tt>, and so on. <tt>FF_QUIT_EVENT</tt> is defined in our program as <tt>SDL_USEREVENT + 2</tt>. We can also pass user data if we like, too, and here we pass our pointer to the big struct. Finally we call <tt>SDL_PushEvent()</tt>. In our event loop switch, we just put this by the <tt>SDL_QUIT_EVENT</tt> section we had before. We'll see our event loop in more detail; for now, just be assured that when we push the <tt>FF_QUIT_EVENT</tt>, we'll catch it later and raise our <tt>quit</tt> flag.
<p></p>
<h3>Getting the Frame: <tt>video_thread</tt></h3>
<p>
After we have our codec prepared, we start our video thread. This thread reads in packets from the video queue, decodes the video into frames, and then calls a <tt>queue_picture</tt> function to put the processed frame onto a picture queue:
</p><pre>int video_thread(void *arg) {
VideoState *is = (VideoState *)arg;
AVPacket pkt1, *packet = &pkt1;
int len1, frameFinished;
AVFrame *pFrame;
pFrame = avcodec_alloc_frame();
for(;;) {
if(packet_queue_get(&is->videoq, packet, 1) < 0) {
// means we quit getting packets
break;
}
// Decode video frame
len1 = avcodec_decode_video(is->video_st->codec, pFrame, &frameFinished,
packet->data, packet->size);
// Did we get a video frame?
if(frameFinished) {
if(queue_picture(is, pFrame) < 0) {
break;
}
}
av_free_packet(packet);
}
av_free(pFrame);
return 0;
}
</pre>
Most of this function should be familiar by this point. We've moved our <tt>avcodec_decode_video</tt> function here, just replaced some of the arguments; for example, we have the AVStream stored in our big struct, so we get our codec from there. We just keep getting packets from our video queue until someone tells us to quit or we encounter an error.
<p></p>
<h3>Queueing the Frame</h3>
<p>
Let's look at the function that stores our decoded frame, <tt>pFrame</tt> in our picture queue. Since our picture queue is an SDL overlay (presumably to allow the video display function to have as little calculation as possible), we need to convert our frame into that. The data we store in the picture queue is a struct of our making:
</p><pre>typedef struct VideoPicture {
SDL_Overlay *bmp;
int width, height; /* source height & width */
int allocated;
} VideoPicture;
</pre>
Our big struct has a buffer of these in it where we can store them. However, we need to allocate the <tt>SDL_Overlay</tt> ourselves (notice the <tt>allocated</tt> flag that will indicate whether we have done so or not).
<p></p>
<p>
To use this queue, we have two pointers - the writing index and the reading index. We also keep track of how many actual pictures are in the buffer. To write to the queue, we're going to first wait for our buffer to clear out so we have space to store our <tt>VideoPicture</tt>. Then we check and see if we have already allocated the overlay at our writing index. If not, we'll have to allocate some space. We also have to reallocate the buffer if the size of the window has changed! However, instead of allocating it here, to avoid locking issues. (I'm still not quite sure why; I believe it's to avoid calling the SDL overlay functions in different threads.)
</p><pre>int queue_picture(VideoState *is, AVFrame *pFrame) {
VideoPicture *vp;
int dst_pix_fmt;
AVPicture pict;
/* wait until we have space for a new pic */
SDL_LockMutex(is->pictq_mutex);
while(is->pictq_size >= VIDEO_PICTURE_QUEUE_SIZE &&
!is->quit) {
SDL_CondWait(is->pictq_cond, is->pictq_mutex);
}
SDL_UnlockMutex(is->pictq_mutex);
if(is->quit)
return -1;
// windex is set to 0 initially
vp = &is->pictq[is->pictq_windex];
/* allocate or resize the buffer! */
if(!vp->bmp ||
vp->width != is->video_st->codec->width ||
vp->height != is->video_st->codec->height) {
SDL_Event event;
vp->allocated = 0;
/* we have to do it in the main thread */
event.type = FF_ALLOC_EVENT;
event.user.data1 = is;
SDL_PushEvent(&event);
/* wait until we have a picture allocated */
SDL_LockMutex(is->pictq_mutex);
while(!vp->allocated && !is->quit) {
SDL_CondWait(is->pictq_cond, is->pictq_mutex);
}
SDL_UnlockMutex(is->pictq_mutex);
if(is->quit) {
return -1;
}
}
</pre>
The event mechanism here is the same one we saw earlier when we wanted to quit. We've defined <tt>FF_ALLOC_EVENT</tt> as <tt>SDL_USEREVENT</tt>. We push the event and then wait on the conditional variable for the allocation function to run.
<p></p>
<p>
Let's look at how we change our event loop:
</p><pre>for(;;) {
SDL_WaitEvent(&event);
switch(event.type) {
/* ... */
case FF_ALLOC_EVENT:
alloc_picture(event.user.data1);
break;
</pre>
Remember that <tt>event.user.data1</tt> is our big struct. That was simple enough. Let's look at the <tt>alloc_picture()</tt> function:
<pre>void alloc_picture(void *userdata) {
VideoState *is = (VideoState *)userdata;
VideoPicture *vp;
vp = &is->pictq[is->pictq_windex];
if(vp->bmp) {
// we already have one make another, bigger/smaller
SDL_FreeYUVOverlay(vp->bmp);
}
// Allocate a place to put our YUV image on that screen
vp->bmp = SDL_CreateYUVOverlay(is->video_st->codec->width,
is->video_st->codec->height,
SDL_YV12_OVERLAY,
screen);
vp->width = is->video_st->codec->width;
vp->height = is->video_st->codec->height;
SDL_LockMutex(is->pictq_mutex);
vp->allocated = 1;
SDL_CondSignal(is->pictq_cond);
SDL_UnlockMutex(is->pictq_mutex);
}
</pre>
You should recognize the <tt>SDL_CreateYUVOverlay</tt> function that we've moved from our main loop to this section. This code should be fairly self-explanitory by now. Remember that we save the width and height in the <tt>VideoPicture</tt> structure because we need to make sure that our video size doesn't change for some reason.
<p></p>
<p>
Okay, we're all settled and we have our YUV overlay allocated and ready to receive a picture. Let's go back to <tt>queue_picture</tt> and look at the code to copy the frame into the overlay. You should recognize that part of it:
</p><pre>int queue_picture(VideoState *is, AVFrame *pFrame) {
/* Allocate a frame if we need it... */
/* ... */
/* We have a place to put our picture on the queue */
if(vp->bmp) {
SDL_LockYUVOverlay(vp->bmp);
dst_pix_fmt = PIX_FMT_YUV420P;
/* point pict at the queue */
pict.data[0] = vp->bmp->pixels[0];
pict.data[1] = vp->bmp->pixels[2];
pict.data[2] = vp->bmp->pixels[1];
pict.linesize[0] = vp->bmp->pitches[0];
pict.linesize[1] = vp->bmp->pitches[2];
pict.linesize[2] = vp->bmp->pitches[1];
// Convert the image into YUV format that SDL uses
img_convert(&pict, dst_pix_fmt,
(AVPicture *)pFrame, is->video_st->codec->pix_fmt,
is->video_st->codec->width, is->video_st->codec->height);
SDL_UnlockYUVOverlay(vp->bmp);
/* now we inform our display thread that we have a pic ready */
if(++is->pictq_windex == VIDEO_PICTURE_QUEUE_SIZE) {
is->pictq_windex = 0;
}
SDL_LockMutex(is->pictq_mutex);
is->pictq_size++;
SDL_UnlockMutex(is->pictq_mutex);
}
return 0;
}
</pre>
The majority of this part is simply the code we used earlier to fill the YUV overlay with our frame. The last bit is simply "adding" our value onto the queue. The queue works by adding onto it until it is full, and reading from it as long as there is something on it. Therefore everything depends upon the <tt>is->pictq_size</tt> value, requiring us to lock it. So what we do here is increment the write pointer (and rollover if necessary), then lock the queue and increase its size. Now our reader will know there is more information on the queue, and if this makes our queue full, our writer will know about it.
<p></p>
<h3>Displaying the Video</h3>
<p>
That's it for our video thread! Now we've wrapped up all the loose threads except for one remember that we called the <tt>schedule_refresh()</tt> function way back? Let's see what that actually did:
</p><pre>/* schedule a video refresh in 'delay' ms */
static void schedule_refresh(VideoState *is, int delay) {
SDL_AddTimer(delay, sdl_refresh_timer_cb, is);
}
</pre>
<tt>SDL_AddTimer()</tt> is an SDL function that simply makes a callback to the user-specfied function after a certain number of milliseconds (and optionally carrying some user data). We're going to use this function to schedule video updates - every time we call this function, it will set the timer, which will trigger an event, which will have our <tt>main()</tt> function in turn call a function that pulls a frame from our picture queue and displays it! Phew!
<p></p>
<p>
But first thing's first. Let's trigger that event. That sends us over to:
</p><pre>static Uint32 sdl_refresh_timer_cb(Uint32 interval, void *opaque) {
SDL_Event event;
event.type = FF_REFRESH_EVENT;
event.user.data1 = opaque;
SDL_PushEvent(&event);
return 0; /* 0 means stop timer */
}
</pre>
Here is the now-familiar event push. <tt>FF_REFRESH_EVENT</tt> is defined here as <tt>SDL_USEREVENT + 1</tt>. One thing to notice is that when we <tt>return 0</tt>, SDL stops the timer so the callback is not made again.
<p></p>
<p>
Now we've pushed an <tt>FF_REFRESH_EVENT</tt>, we need to handle it in our event loop:
</p><pre>for(;;) {
SDL_WaitEvent(&event);
switch(event.type) {
/* ... */
case FF_REFRESH_EVENT:
video_refresh_timer(event.user.data1);
break;
</pre>
and <i>that</i> sends us to this function, which will actually pull the data from our picture queue:
<pre>void video_refresh_timer(void *userdata) {
VideoState *is = (VideoState *)userdata;
VideoPicture *vp;
if(is->video_st) {
if(is->pictq_size == 0) {
schedule_refresh(is, 1);
} else {
vp = &is->pictq[is->pictq_rindex];
/* Timing code goes here */
schedule_refresh(is, 80);
/* show the picture! */
video_display(is);
/* update queue for next picture! */
if(++is->pictq_rindex == VIDEO_PICTURE_QUEUE_SIZE) {
is->pictq_rindex = 0;
}
SDL_LockMutex(is->pictq_mutex);
is->pictq_size--;
SDL_CondSignal(is->pictq_cond);
SDL_UnlockMutex(is->pictq_mutex);
}
} else {
schedule_refresh(is, 100);
}
}
</pre>
For now, this is a pretty simple function: it pulls from the queue when we have something, sets our timer for when the next video frame should be shown, calls video_display to actually show the video on the screen, then increments the counter on the queue, and decreases its size. You may notice that we don't actually do anything with <tt>vp</tt> in this function, and here's why: we will. Later. We're going to use it to access timing information when we start syncing the video to the audio. See where it says "timing code here"? In that section, we're going to figure out how soon we should show the next video frame, and then input that value into the <tt>schedule_refresh()</tt> function. For now we're just putting in a dummy value of 80. Technically, you could guess and check this value, and recompile it for every movie you watch, but 1) it would drift after a while and 2) it's quite silly. We'll come back to it later, though.
<p></p>
<p>
We're almost done; we just have one last thing to do: display the video! Here's that <tt>video_display</tt> function:
</p><pre>
<p></p>void video_display(VideoState *is) {
SDL_Rect rect;
VideoPicture *vp;
AVPicture pict;
float aspect_ratio;
int w, h, x, y;
int i;
vp = &is->pictq[is->pictq_rindex];
if(vp->bmp) {
if(is->video_st->codec->sample_aspect_ratio.num == 0) {
aspect_ratio = 0;
} else {
aspect_ratio = av_q2d(is->video_st->codec->sample_aspect_ratio) *
is->video_st->codec->width / is->video_st->codec->height;
}
if(aspect_ratio <= 0.0) {
aspect_ratio = (float)is->video_st->codec->width /
(float)is->video_st->codec->height;
}
h = screen->h;
w = ((int)rint(h * aspect_ratio)) & -3;
if(w > screen->w) {
w = screen->w;
h = ((int)rint(w / aspect_ratio)) & -3;
}
x = (screen->w - w) / 2;
y = (screen->h - h) / 2;
rect.x = x;
rect.y = y;
rect.w = w;
rect.h = h;
SDL_DisplayYUVOverlay(vp->bmp, &rect);
}
}
</pre>
Since our screen can be of any size (we set ours to 640x480 and there are ways to set it so it is resizable by the user), we need to dynamically figure out how big we want our movie rectangle to be. So first we need to figure out our movie's <b>aspect ratio</b>, which is just the width divided by the height. Some codecs will have an odd <b>sample aspect ratio</b>, which is simply the width/height radio of a single pixel, or sample. Since the height and width values in our codec context are measured in pixels, the actual aspect ratio is equal to the aspect ratio times the sample aspect ratio. Some codecs will show an aspect ratio of 0, and this indicates that each pixel is simply of size 1x1. Then we scale the movie to fit as big in our <tt>screen</tt> as we can. The <tt>& -3</tt> bit-twiddling in there simply rounds the value to the nearest multiple of 4. Then we center the movie, and call <tt>SDL_DisplayYUVOverlay()</tt>.
<p></p>
<p>
So is that it? Are we done? Well, we still have to rewrite the audio code to use the new <tt>VideoStruct</tt>, but those are trivial changes, and you can look at those in the sample code. The last thing we have to do is to change our callback for ffmpeg's internal "quit" callback function:
</p><pre>VideoState *global_video_state;
int decode_interrupt_cb(void) {
return (global_video_state && global_video_state->quit);
}
</pre>
We set <tt>global_video_state</tt> to the big struct in <tt>main()</tt>.
<p></p>
<p>
So that's it! Go ahead and compile it:
</p><pre>gcc -o tutorial04 tutorial04.c -lavutil -lavformat -lavcodec -lz -lm \
`sdl-config --cflags --libs`
</pre>
and enjoy your unsynced movie! Next time we'll finally build a video player that actually works!
<p></p>
<p>
<em><b>>></b> Syncing Video</em>
</p>
<a name="tutorial05.html"></a>
<h2>Tutorial 05: Synching Video</h2>
<span class="codelink">Code: tutorial05.c</span>
<h3>How Video Syncs</h3>
<p>
So this whole time, we've had an essentially useless movie player. It plays the video, yeah, and it plays the audio, yeah, but it's not quite yet what we would call a <i>movie</i>. So what do we do?
</p>
<h4>PTS and DTS</h4>
<p>
Fortunately, both the audio and video streams have the information about how fast and when you are supposed to play them inside of them. Audio streams have a sample rate, and the video streams have a frames per second value. However, if we simply synced the video by just counting frames and multiplying by frame rate, there is a chance that it will go out of sync with the audio. Instead, packets from the stream might have what is called a <b>decoding time stamp (DTS)</b> and a <b>presentation time stamp (PTS)</b>. To understand these two values, you need to know about the way movies are stored. Some formats, like MPEG, use what they call "B" frames (B stands for "bidirectional"). The two other kinds of frames are called "I" frames and "P" frames ("I" for "intra" and "P" for "predicted"). I frames contain a full image. P frames depend upon previous I and P frames and are like diffs or deltas. B frames are the same as P frames, but depend upon information found in frames that are displayed both <i>before</i> and <i>after</i> them! This explains why we might not have a finished frame after we call <tt>avcodec_decode_video</tt>.
</p>
<p>
So let's say we had a movie, and the frames were displayed like: I B B P. Now, we need to know the information in P before we can display either B frame. Because of this, the frames might be stored like this: I P B B. This is why we have a decoding timestamp and a presentation timestamp on each frame. The decoding timestamp tells us when we need to decode something, and the presentation time stamp tells us when we need to display something. So, in this case, our stream might look like this:
</p><pre> PTS: 1 4 2 3
DTS: 1 2 3 4
Stream: I P B B
</pre>
Generally the PTS and DTS will only differ when the stream we are playing has B frames in it.
<p></p>
<p>
When we get a packet from <tt>av_read_frame()</tt>, it will contain the PTS and DTS values for the information inside that packet. But what we really want is the PTS of our newly decoded raw frame, so we know when to display it. However, the frame we get from <tt>avcodec_decode_video()</tt> gives us an AVFrame, which doesn't contain a useful PTS value. (Warning: AVFrame does contain a <tt>pts</tt> variable, but this will not always contain what we want when we get a frame.) However, ffmpeg reorders the packets so that the DTS of the packet being processed by <tt>avcodec_decode_video()</tt> will <i>always be the same</i> as the PTS of the frame it returns. But, another warning: we won't always get this information, either.
</p>
<p>
Not to worry, because there's another way to find out the PTS of a frame, and we can have our program reorder the packets by itself. We save the PTS of the first packet of a frame: this will be the PTS of the finished frame. So when the stream doesn't give us a DTS, we just use this saved PTS. We can figure out which packet is the first packet of a frame by letting <tt>avcodec_decode_video()</tt> tell us. How? Whenever a packet starts a frame, the <tt>avcodec_decode_video()</tt> will call a function to allocate a buffer in our frame. And of course, ffmpeg allows us to redefine what that allocation function is. So we'll make a new function that saves the pts of the packet.
</p>
<p>Of course, even then, we might not get a proper pts. We'll deal with that later.
</p>
<h4>Synching</h4>
<p>
Now, while it's all well and good to know when we're supposed to show a particular video frame, but how do we actually do so? Here's the idea: after we show a frame, we figure out when the next frame should be shown. Then we simply set a new timeout to refresh the video again after that amount of time. As you might expect, we check the value of the PTS of the next frame against the system clock to see how long our timeout should be. This approach works, but there are two issues that need to be dealt with.
</p>
<p>
First is the issue of knowing when the next PTS will be. Now, you might think that we can just add the video rate to the current PTS and you'd be mostly right. However, some kinds of video call for frames to be repeated. This means that we're supposed to repeat the current frame a certain number of times. This could cause the program to display the next frame too soon. So we need to account for that.
</p>
<p>
The second issue is that as the program stands now, the video and the audio chugging away happily, not bothering to sync at all. We wouldn't have to worry about that if everything worked perfectly. But your computer isn't perfect, and a lot of video files aren't, either. So we have three choices: sync the audio to the video, sync the video to the audio, or sync both to an external clock (like your computer). For now, we're going to sync the video to the audio.
</p>
<h3>Coding it: getting the frame PTS</h3>
<p>
Now let's get into the code to do all this. We're going to need to add some more members to our big struct, but we'll do this as we need to. First let's look at our video thread. Remember, this is where we pick up the packets that were put on the queue by our decode thread. What we need to do in this part of the code is get the PTS of the frame given to us by <tt>avcodec_decode_video</tt>. The first way we talked about was getting the DTS of the last packet processed, which is pretty easy:
</p><pre> double pts;
for(;;) {
if(packet_queue_get(&is->videoq, packet, 1) < 0) {
// means we quit getting packets
break;
}
pts = 0;
// Decode video frame
len1 = avcodec_decode_video(is->video_st->codec,
pFrame, &frameFinished,
packet->data, packet->size);
if(packet->dts != AV_NOPTS_VALUE) {
pts = packet->dts;
} else {
pts = 0;
}
pts *= av_q2d(is->video_st->time_base);
</pre>
We set the PTS to 0 if we can't figure out what it is.
<p></p>
<p>
Well, that was easy. But we said before that if the packet's DTS doesn't help us, we need to use the PTS of the first packet that was decoded for the frame. We do this by telling ffmpeg to use our custom functions to allocate a frame. Here's the types of the functions:
</p><pre>int get_buffer(struct AVCodecContext *c, AVFrame *pic);
void release_buffer(struct AVCodecContext *c, AVFrame *pic);
</pre>
The get function doesn't tell us anything about packets, so what we'll do is save the PTS to a global variable each time we get a packet, and our get function can just read that. Then, we can store the value in the AVFrame structure's opaque variable. This is a user-defined variable, so we can use it for whatever we want. So first, here are our functions:
<pre>uint64_t global_video_pkt_pts = AV_NOPTS_VALUE;
/* These are called whenever we allocate a frame
* buffer. We use this to store the global_pts in
* a frame at the time it is allocated.
*/
int our_get_buffer(struct AVCodecContext *c, AVFrame *pic) {
int ret = avcodec_default_get_buffer(c, pic);
uint64_t *pts = av_malloc(sizeof(uint64_t));
*pts = global_video_pkt_pts;
pic->opaque = pts;
return ret;
}
void our_release_buffer(struct AVCodecContext *c, AVFrame *pic) {
if(pic) av_freep(&pic->opaque);
avcodec_default_release_buffer(c, pic);
}
</pre>
<tt>avcodec_default_get_buffer</tt> and <tt>avcodec_default_release_buffer</tt> are the default functions that ffmpeg uses for allocation. <tt>av_freep</tt> is a memory management function that not only frees the memory pointed to, but sets the pointer to <tt>NULL</tt>.
<p></p>
<p>
Now going way down to our stream open function (<tt>stream_component_open</tt>), we add these lines to tell ffmpeg what to do:
</p><pre> codecCtx->get_buffer = our_get_buffer;
codecCtx->release_buffer = our_release_buffer;
</pre>
Now we have to add code to save the PTS to the global variable and then use the stored PTS when necessary. Our code now looks like this:
<pre> for(;;) {
if(packet_queue_get(&is->videoq, packet, 1) < 0) {
// means we quit getting packets
break;
}
pts = 0;
// Save global pts to be stored in pFrame in first call
global_video_pkt_pts = packet->pts;
// Decode video frame
len1 = avcodec_decode_video(is->video_st->codec, pFrame, &frameFinished,
packet->data, packet->size);
if(packet->dts == AV_NOPTS_VALUE
&& pFrame->opaque && *(uint64_t*)pFrame->opaque != AV_NOPTS_VALUE) {
pts = *(uint64_t *)pFrame->opaque;
} else if(packet->dts != AV_NOPTS_VALUE) {
pts = packet->dts;
} else {
pts = 0;
}
pts *= av_q2d(is->video_st->time_base);
</pre>
A technical note: You may have noticed we're using <tt>int64</tt> for the PTS. This is because the PTS is stored as an integer. This value is a timestamp that corresponds to a measurement of time in that stream's time_base unit. For example, if a stream has 24 frames per second, a PTS of 42 is going to indicate that the frame should go where the 42nd frame would be if there we had a frame every 1/24 of a second (certainly not necessarily true). <p></p>
<p>
We can convert this value to seconds by dividing by the framerate. The <tt>time_base</tt> value of the stream is going to be 1/framerate (for fixed-fps content), so to get the PTS in seconds, we multiply by the <tt>time_base</tt>.
</p>
<h3>Coding: Synching and using the PTS</h3>
<p>
So now we've got our PTS all set. Now we've got to take care of the two synchronization problems we talked about above. We're going to define a function called <tt>synchronize_video</tt> that will update the PTS to be in sync with everything. This function will also finally deal with cases where we don't get a PTS value for our frame. At the same time we need to keep track of when the next frame is expected so we can set our refresh rate properly. We can accomplish this by using an internal video_clock value which keeps track of how much time has passed according to the video. We add this value to our big struct.
</p><pre>typedef struct VideoState {
double video_clock; ///<pts of="" last="" decoded="" frame="" predicted="" pts="" next="" <="" pre="">
Here's the synchronize_video function, which is pretty self-explanatory:
<pre>double synchronize_video(VideoState *is, AVFrame *src_frame, double pts) {
double frame_delay;
if(pts != 0) {
/* if we have pts, set video clock to it */
is->video_clock = pts;
} else {
/* if we aren't given a pts, set it to the clock */
pts = is->video_clock;
}
/* update the video clock */
frame_delay = av_q2d(is->video_st->codec->time_base);
/* if we are repeating a frame, adjust clock accordingly */
frame_delay += src_frame->repeat_pict * (frame_delay * 0.5);
is->video_clock += frame_delay;
return pts;
}
</pre>
You'll notice we account for repeated frames in this function, too.
<p></p>
<p>
Now let's get our proper PTS and queue up the frame using <tt>queue_picture</tt>, adding a new pts argument:
</p><pre> // Did we get a video frame?
if(frameFinished) {
pts = synchronize_video(is, pFrame, pts);
if(queue_picture(is, pFrame, pts) < 0) {
break;
}
}
</pre>
The only thing that changes about <tt>queue_picture</tt> is that we save that pts value to the VideoPicture structure that we queue up. So we have to add a pts variable to the struct and add a line of code:
<pre>typedef struct VideoPicture {
...
double pts;
}
int queue_picture(VideoState *is, AVFrame *pFrame, double pts) {
... stuff ...
if(vp->bmp) {
... convert picture ...
vp->pts = pts;
... alert queue ...
}
</pre>
So now we've got pictures lining up onto our picture queue with proper PTS values, so let's take a look at our video refreshing function. You may recall from last time that we just faked it and put a refresh of 80ms. Well, now we're going to find out how to actually figure it out.
<p></p>
<p>
Our strategy is going to be to predict the time of the next PTS by simply measuring the time between the previous pts and this one. At the same time, we need to sync the video to the audio. We're going to make an <b>audio clock</b>: an internal value thatkeeps track of what position the audio we're playing is at. It's like the digital readout on any mp3 player. Since we're synching the video to the audio, the video thread uses this value to figure out if it's too far ahead or too far behind.
</p>
<p>
We'll get to the implementation later; for now let's assume we have a <tt>get_audio_clock</tt> function that will give us the time on the audio clock. Once we have that value, though, what do we do if the video and audio are out of sync? It would silly to simply try and leap to the correct packet through seeking or something. Instead, we're just going to adjust the value we've calculated for the next refresh: if the PTS is too far behind the audio time, we double our calculated delay. if the PTS is too far ahead of the audio time, we simply refresh as quickly as possible. Now that we have our adjusted refresh time, or <b>delay</b>, we're going to compare that with our computer's clock by keeping a running <tt>frame_timer</tt>. This frame timer will sum up all of our calculated delays while playing the movie. In other words, this frame_timer is <i>what time it should be when we display the next frame.</i> We simply add the new delay to the frame timer, compare it to the time on our computer's clock, and use that value to schedule the next refresh. This might be a bit confusing, so study the code carefully:
</p><pre>void video_refresh_timer(void *userdata) {
VideoState *is = (VideoState *)userdata;
VideoPicture *vp;
double actual_delay, delay, sync_threshold, ref_clock, diff;
if(is->video_st) {
if(is->pictq_size == 0) {
schedule_refresh(is, 1);
} else {
vp = &is->pictq[is->pictq_rindex];
delay = vp->pts - is->frame_last_pts; /* the pts from last time */
if(delay <= 0 || delay >= 1.0) {
/* if incorrect delay, use previous one */
delay = is->frame_last_delay;
}
/* save for next time */
is->frame_last_delay = delay;
is->frame_last_pts = vp->pts;
/* update delay to sync to audio */
ref_clock = get_audio_clock(is);
diff = vp->pts - ref_clock;
/* Skip or repeat the frame. Take delay into account
FFPlay still doesn't "know if this is the best guess." */
sync_threshold = (delay > AV_SYNC_THRESHOLD) ? delay : AV_SYNC_THRESHOLD;
if(fabs(diff) < AV_NOSYNC_THRESHOLD) {
if(diff <= -sync_threshold) {
delay = 0;
} else if(diff >= sync_threshold) {
delay = 2 * delay;
}
}
is->frame_timer += delay;
/* computer the REAL delay */
actual_delay = is->frame_timer - (av_gettime() / 1000000.0);
if(actual_delay < 0.010) {
/* Really it should skip the picture instead */
actual_delay = 0.010;
}
schedule_refresh(is, (int)(actual_delay * 1000 + 0.5));
/* show the picture! */
video_display(is);
/* update queue for next picture! */
if(++is->pictq_rindex == VIDEO_PICTURE_QUEUE_SIZE) {
is->pictq_rindex = 0;
}
SDL_LockMutex(is->pictq_mutex);
is->pictq_size--;
SDL_CondSignal(is->pictq_cond);
SDL_UnlockMutex(is->pictq_mutex);
}
} else {
schedule_refresh(is, 100);
}
}
</pre>
There are a few checks we make: first, we make sure that the delay between the PTS and the previous PTS make sense. If it doesn't we just guess and use the last delay. Next, we make sure we have a synch threshold because things are never going to be perfectly in synch. ffplay uses 0.01 for its value. We also make sure that the synch threshold is never smaller than the gaps in between PTS values. Finally, we make the minimum refresh value 10 milliseconds*.<span class="sidenote"><b>*</b> Really here we should skip the frame, but we're not going to bother.</span>
<p></p>
<p>
We added a bunch of variables to the big struct so don't forget to check the code. Also, don't forget to initialize the frame timer and the initial previous frame delay in <tt>stream_component_open</tt>:
</p><pre> is->frame_timer = (double)av_gettime() / 1000000.0;
is->frame_last_delay = 40e-3;
</pre>
<p></p>
<h3>Synching: The Audio Clock</h3>
<p>
Now it's time for us to implement the audio clock. We can update the clock time in our <tt>audio_decode_frame</tt> function, which is where we decode the audio. Now, remember that we don't always process a new packet every time we call this function, so there are two places we have to update the clock at. The first place is where we get the new packet: we simply set the audio clock to the packet's PTS. Then if a packet has multiple frames, we keep time the audio play by counting the number of samples and multiplying them by the given samples-per-second rate. So once we have the packet:
</p><pre> /* if update, update the audio clock w/pts */
if(pkt->pts != AV_NOPTS_VALUE) {
is->audio_clock = av_q2d(is->audio_st->time_base)*pkt->pts;
}
</pre>
And once we are processing the packet:
<pre> /* Keep audio_clock up-to-date */
pts = is->audio_clock;
*pts_ptr = pts;
n = 2 * is->audio_st->codec->channels;
is->audio_clock += (double)data_size /
(double)(n * is->audio_st->codec->sample_rate);
</pre>
A few fine details: the template of the function has changed to include <tt>pts_ptr</tt>, so make sure you change that. <tt>pts_ptr</tt> is a pointer we use to inform <tt>audio_callback</tt> the pts of the audio packet. This will be used next time for synchronizing the audio with the video.
<p></p>
<p>
Now we can finally implement our <tt>get_audio_clock</tt> function. It's not as simple as getting the <tt>is->audio_clock</tt> value, thought. Notice that we set the audio PTS every time we process it, but if you look at the <tt>audio_callback</tt> function, it takes time to move all the data from our audio packet into our output buffer. That means that the value in our audio clock could be too far ahead. So we have to check how much we have left to write. Here's the complete code:
</p><pre>double get_audio_clock(VideoState *is) {
double pts;
int hw_buf_size, bytes_per_sec, n;
pts = is->audio_clock; /* maintained in the audio thread */
hw_buf_size = is->audio_buf_size - is->audio_buf_index;
bytes_per_sec = 0;
n = is->audio_st->codec->channels * 2;
if(is->audio_st) {
bytes_per_sec = is->audio_st->codec->sample_rate * n;
}
if(bytes_per_sec) {
pts -= (double)hw_buf_size / bytes_per_sec;
}
return pts;
}
</pre>
You should be able to tell why this function works by now ;)
<p></p>
<p>
So that's it! Go ahead and compile it:
</p><pre>gcc -o tutorial05 tutorial05.c -lavutil -lavformat -lavcodec -lz -lm`sdl-config --cflags --libs`
</pre>
and <b>finally!</b> you can watch a movie on your own movie player. Next time we'll look at audio synching, and then the tutorial after that we'll talk about seeking.
<p></p>
<p>
<em><b>>></b> Synching Audio</em>
</p>
<a name="tutorial06.html"></a>
<h2>Tutorial 06: Synching Audio</h2>
<span class="codelink">Code: tutorial06.c</span>
<h3>Synching Audio</h3>
<p>So now we have a decent enough player to watch a movie, so let's see what kind of loose ends we have lying around. Last time, we glossed over synchronization a little bit, namely sychronizing audio to a video clock rather than the other way around. We're going to do this the same way as with the video: make an internal video clock to keep track of how far along the video thread is and sync the audio to that. Later we'll look at how to generalize things to sync both audio and video to an external clock, too.
</p>
<h3>Implementing the video clock</h3>
<p>
Now we want to implement a video clock similar to the audio clock we had last time: an internal value that gives the current time offset of the video currently being played. At first, you would think that this would be as simple as updating the timer with the current PTS of the last frame to be shown. However, don't forget that the time between video frames can be pretty long when we get down to the millisecond level. The solution is to keep track of another value, the time at which we set the video clock to the PTS of the last frame. That way the current value of the video clock will be <tt>PTS_of_last_frame + (current_time - time_elapsed_since_PTS_value_was_set)</tt>. This solution is very similar to what we did with <tt>get_audio_clock</tt>.
</p>
<p>
So, in our big struct, we're going to put a <tt>double video_current_pts</tt> and a <tt>int64_t video_current_pts_time</tt>. The clock updating is going to take place in the <tt>video_refresh_timer</tt> function:
</p><pre>void video_refresh_timer(void *userdata) {
/* ... */
if(is->video_st) {
if(is->pictq_size == 0) {
schedule_refresh(is, 1);
} else {
vp = &is->pictq[is->pictq_rindex];
is->video_current_pts = vp->pts;
is->video_current_pts_time = av_gettime();
</pre>
Don't forget to initialize it in <tt>stream_component_open</tt>:
<pre> is->video_current_pts_time = av_gettime();
</pre>
And now all we need is a way to get the information:
<pre>double get_video_clock(VideoState *is) {
double delta;
delta = (av_gettime() - is->video_current_pts_time) / 1000000.0;
return is->video_current_pts + delta;
}
</pre>
<p></p>
<h3>Abstracting the clock</h3>
<p>
But why force ourselves to use the video clock? We'd have to go and alter our video sync code so that the audio and video aren't trying to sync to each other. Imagine the mess if we tried to make it a command line option like it is in ffplay. So let's abstract things: we're going to make a new wrapper function, <tt>get_master_clock</tt> that checks an av_sync_type variable and then call <tt>get_audio_clock</tt>, <tt>get_video_clock</tt>, or whatever other clock we want to use. We could even use the computer clock, which we'll call <tt>get_external_clock</tt>:
</p><pre>enum {
AV_SYNC_AUDIO_MASTER,
AV_SYNC_VIDEO_MASTER,
AV_SYNC_EXTERNAL_MASTER,
};
#define DEFAULT_AV_SYNC_TYPE AV_SYNC_VIDEO_MASTER
double get_master_clock(VideoState *is) {
if(is->av_sync_type == AV_SYNC_VIDEO_MASTER) {
return get_video_clock(is);
} else if(is->av_sync_type == AV_SYNC_AUDIO_MASTER) {
return get_audio_clock(is);
} else {
return get_external_clock(is);
}
}
main() {
...
is->av_sync_type = DEFAULT_AV_SYNC_TYPE;
...
}
</pre>
<p></p>
<h3>Synchronizing the Audio</h3>
<p>Now the hard part: synching the audio to the video clock. Our strategy is going to be to measure where the audio is, compare it to the video clock, and then figure out how many samples we need to adjust by, that is, do we need to speed up by dropping samples or do we need to slow down by adding them?
</p>
We're going to run a <tt>synchronize_audio</tt> function each time we process each set of audio samples we get to shrink or expand them properly. However, we don't want to sync every single time it's off because process audio a lot more often than video packets. So we're going to set a minimum number of consecutive calls to the <tt>synchronize_audio</tt> function that have to be out of sync before we bother doing anything. Of course, just like last time, "out of sync" means that the audio clock and the video clock differ by more than our sync threshold.
<p></p>
<p>
<span class="sidenote"><b>Note:</b> What the heck is going on here? This equation looks like magic! Well, it's basically a weighted mean using a geometric series as weights. I don't know if there's a name for this (I even checked Wikipedia!) but for more info, here's an explanation (or at weightedmean.txt)</span>
So we're going to use a fractional coefficient, say <tt>c</tt>, and
So now let's say we've gotten N audio sample sets that have been out of sync. The amount we are out of sync can also vary a good deal, so we're going to take an average of how far each of those have been out of sync. So for example, the first call might have shown we were out of sync by 40ms, the next by 50ms, and so on. But we're not going to take a simple average because the most recent values are more important than the previous ones. So we're going to use a fractional coefficient, say <tt>c</tt>, and sum the differences like this: <tt>diff_sum = new_diff + diff_sum*c</tt>. When we are ready to find the average difference, we simply calculate <tt>avg_diff = diff_sum * (1-c)</tt>.
</p>
<p>
Here's what our function looks like so far:
</p><pre>/* Add or subtract samples to get a better sync, return new
audio buffer size */
int synchronize_audio(VideoState *is, short *samples,
int samples_size, double pts) {
int n;
double ref_clock;
n = 2 * is->audio_st->codec->channels;
if(is->av_sync_type != AV_SYNC_AUDIO_MASTER) {
double diff, avg_diff;
int wanted_size, min_size, max_size, nb_samples;
ref_clock = get_master_clock(is);
diff = get_audio_clock(is) - ref_clock;
if(diff < AV_NOSYNC_THRESHOLD) {
// accumulate the diffs
is->audio_diff_cum = diff + is->audio_diff_avg_coef
* is->audio_diff_cum;
if(is->audio_diff_avg_count < AUDIO_DIFF_AVG_NB) {
is->audio_diff_avg_count++;
} else {
avg_diff = is->audio_diff_cum * (1.0 - is->audio_diff_avg_coef);
/* Shrinking/expanding buffer code.... */
}
} else {
/* difference is TOO big; reset diff stuff */
is->audio_diff_avg_count = 0;
is->audio_diff_cum = 0;
}
}
return samples_size;
}
</pre>
<p></p>
<p>
So we're doing pretty well; we know approximately how off the audio is from the video or whatever we're using for a clock. So let's now calculate how many samples we need to add or lop off by putting this code where the "Shrinking/expanding buffer code" section is:
</p><pre>if(fabs(avg_diff) >= is->audio_diff_threshold) {
wanted_size = samples_size +
((int)(diff * is->audio_st->codec->sample_rate) * n);
min_size = samples_size * ((100 - SAMPLE_CORRECTION_PERCENT_MAX)
/ 100);
max_size = samples_size * ((100 + SAMPLE_CORRECTION_PERCENT_MAX)
/ 100);
if(wanted_size < min_size) {
wanted_size = min_size;
} else if (wanted_size > max_size) {
wanted_size = max_size;
}
</pre>
Remember that <tt>audio_length * (sample_rate * # of channels * 2)</tt> is the number of samples in <tt>audio_length</tt> seconds of audio. Therefore, number of samples we want is going to be the number of samples we already have plus or minus the number of samples that correspond to the amount of time the audio has drifted. We'll also set a limit on how big or small our correction can be because if we change our buffer too much, it'll be too jarring to the user.
<p></p>
<h3>Correcting the number of samples</h3>
<p>
Now we have to actually correct the audio. You may have noticed that our <tt>synchronize_audio</tt> function returns a sample size, which will then tell us how many bytes to send to the stream. So we just have to adjust the sample size to the <tt>wanted_size</tt>. This works for making the sample size smaller. But if we want to make it bigger, we can't just make the sample size larger because there's no more data in the buffer! So we have to add it. But what should we add? It would be foolish to try and extrapolate audio, so let's just use the audio we already have by padding out the buffer with the value of the last sample.
</p><pre>if(wanted_size < samples_size) {
/* remove samples */
samples_size = wanted_size;
} else if(wanted_size > samples_size) {
uint8_t *samples_end, *q;
int nb;
/* add samples by copying final samples */
nb = (samples_size - wanted_size);
samples_end = (uint8_t *)samples + samples_size - n;
q = samples_end + n;
while(nb > 0) {
memcpy(q, samples_end, n);
q += n;
nb -= n;
}
samples_size = wanted_size;
}
</pre>
Now we return the sample size, and we're done with that function. All we need to do now is use it:
<pre>void audio_callback(void *userdata, Uint8 *stream, int len) {
VideoState *is = (VideoState *)userdata;
int len1, audio_size;
double pts;
while(len > 0) {
if(is->audio_buf_index >= is->audio_buf_size) {
/* We have already sent all our data; get more */
audio_size = audio_decode_frame(is, is->audio_buf, sizeof(is->audio_buf), &pts);
if(audio_size < 0) {
/* If error, output silence */
is->audio_buf_size = 1024;
memset(is->audio_buf, 0, is->audio_buf_size);
} else {
audio_size = synchronize_audio(is, (int16_t *)is->audio_buf,
audio_size, pts);
is->audio_buf_size = audio_size;
</pre>
All we did is inserted the call to <tt>synchronize_audio</tt>. (Also, make sure to check the source code where we initalize the above variables I didn't bother to define.)
<p></p>
<p>
One last thing before we finish: we need to add an if clause to make sure we don't sync the video if it is the master clock:
</p><pre>if(is->av_sync_type != AV_SYNC_VIDEO_MASTER) {
ref_clock = get_master_clock(is);
diff = vp->pts - ref_clock;
/* Skip or repeat the frame. Take delay into account
FFPlay still doesn't "know if this is the best guess." */
sync_threshold = (delay > AV_SYNC_THRESHOLD) ? delay :
AV_SYNC_THRESHOLD;
if(fabs(diff) < AV_NOSYNC_THRESHOLD) {
if(diff <= -sync_threshold) {
delay = 0;
} else if(diff >= sync_threshold) {
delay = 2 * delay;
}
}
}
</pre>
And that does it! Make sure you check through the source file to initialize any variables that I didn't bother defining or initializing. Then compile it:
<pre>gcc -o tutorial06 tutorial06.c -lavutil -lavformat -lavcodec -lz -lm`sdl-config --cflags --libs`
</pre>
and you'll be good to go.
<p></p>
<p>
Next time we'll make it so you can rewind and fast forward your movie.
</p>
<p>
<em><b>>></b> Seeking</em>
</p>
<a name="tutorial07.html"></a>
<h2>Tutorial 07: Seeking</h2>
<span class="codelink">
Code: tutorial07.c
</span>
<h3>Handling the seek command</h3>
<p>
Now we're going to add some seeking capabilities to our player, because it's really annoying when you can't rewind a movie. Plus, this will show you how easy the <tt>av_seek_frame</tt> function is to use.
</p>
<p>
We're going to make the left and right arrows go back and forth in the movie by a little and the up and down arrows a lot, where "a little" is 10 seconds, and "a lot" is 60 seconds. So we need to set up our main loop so it catches the keystrokes. However, when we do get a keystroke, we can't call <tt>av_seek_frame</tt> directly. We have to do that in our main decode loop, the <tt>decode_thread</tt> loop. So instead, we're going to add some values to the big struct that will contain the new position to seek to and some seeking flags:
</p><pre> int seek_req;
int seek_flags;
int64_t seek_pos;
</pre>
<p></p>
<p>
Now we need to set up our main loop to catch the key presses:
</p><pre> for(;;) {
double incr, pos;
SDL_WaitEvent(&event);
switch(event.type) {
case SDL_KEYDOWN:
switch(event.key.keysym.sym) {
case SDLK_LEFT:
incr = -10.0;
goto do_seek;
case SDLK_RIGHT:
incr = 10.0;
goto do_seek;
case SDLK_UP:
incr = 60.0;
goto do_seek;
case SDLK_DOWN:
incr = -60.0;
goto do_seek;
do_seek:
if(global_video_state) {
pos = get_master_clock(global_video_state);
pos += incr;
stream_seek(global_video_state,
(int64_t)(pos * AV_TIME_BASE), incr);
}
break;
default:
break;
}
break;
</pre>
To detect keypresses, we first look and see if we get an <tt>SDL_KEYDOWN</tt> event. Then we check and see which key got hit using <tt>event.key.keysym.sym</tt>. Once we know which way we want to seek, we calculate the new time by adding the increment to the value from our new <tt>get_master_clock</tt> function. Then we call a <tt>stream_seek</tt> function to set the seek_pos, etc., values. We convert our new time to avcodec's internal timestamp unit. Recall that timestamps in streams are measured in frames rather than seconds, with the formula <tt>seconds = frames * time_base (fps)</tt>. avcodec defaults to a value of 1,000,000 fps (so a <tt>pos</tt> of 2 seconds will be timestamp of 2000000). We'll see why we need to convert this value later.
<p></p>
<p>
Here's our <tt>stream_seek</tt> function. Notice we set a flag if we are going backwards:
</p><pre>void stream_seek(VideoState *is, int64_t pos, int rel) {
if(!is->seek_req) {
is->seek_pos = pos;
is->seek_flags = rel < 0 ? AVSEEK_FLAG_BACKWARD : 0;
is->seek_req = 1;
}
}
</pre>
Now let's go over to our <tt>decode_thread</tt> where we will actually perform our seek. You'll notice in the source files that we've marked an area "seek stuff goes here". Well, we're going to put it there now. <p></p>
<p>
Seeking centers around the <tt>av_seek_frame</tt> function. This function takes a format context, a stream, a timestamp, and a set of flags as an argument. The function will seek to the timestamp you give it. The unit of the timestamp is the <tt>time_base</tt> of the stream you pass the function. <i>However</i>, you do not have to pass it a stream (indicated by passing a value of -1). If you do that, the <tt>time_base</tt> will be in avcodec's internal timestamp unit, or 1000000fps. This is why we multiplied our position by <tt>AV_TIME_BASE</tt> when we set <tt>seek_pos</tt>.
</p>
<p>
However, sometimes you can (rarely) run into problems with some files if you pass <tt>av_seek_frame</tt> -1 for a stream, so we're going to pick the first stream in our file and pass it to <tt>av_seek_frame</tt>. Don't forget we have to rescale our timestamp to be in the new unit too.
</p><pre>if(is->seek_req) {
int stream_index= -1;
int64_t seek_target = is->seek_pos;
if (is->videoStream >= 0) stream_index = is->videoStream;
else if(is->audioStream >= 0) stream_index = is->audioStream;
if(stream_index>=0){
seek_target= av_rescale_q(seek_target, AV_TIME_BASE_Q,
pFormatCtx->streams[stream_index]->time_base);
}
if(av_seek_frame(is->pFormatCtx, stream_index,
seek_target, is->seek_flags) < 0) {
fprintf(stderr, "%s: error while seeking\n",
is->pFormatCtx->filename);
} else {
/* handle packet queues... more later... */
</pre>
<tt>av_rescale_q(a,b,c)</tt> is a function that will rescale a timestamp from one base to another. It basically computes <tt>a*b/c</tt> but this function is required because that calculation could overflow. <tt>AV_TIME_BASE_Q</tt> is the fractional version of <tt>AV_TIME_BASE</tt>. They're quite different: <tt>AV_TIME_BASE * time_in_seconds = avcodec_timestamp</tt> and <tt>AV_TIME_BASE_Q * avcodec_timestamp = time_in_seconds</tt> (but note that <tt>AV_TIME_BASE_Q</tt> is actually an <tt>AVRational</tt> object, so you have to use special q functions in avcodec to handle it).
<p></p>
<h3>Flushing our buffers</h3>
<p>
So we've set our seek correctly, but we aren't finished quite yet. Remember that we have a queue set up to accumulate packets. Now that we're in a different place, we have to flush that queue or the movie ain't gonna seek! Not only that, but avcodec has its own internal buffers that need to be flushed too by each thread.
</p>
<p>
To do this, we need to first write a function to clear our packet queue. Then, we need to have some way of instructing the audio and video thread that they need to flush avcodec's internal buffers. We can do this by putting a special packet on the queue after we flush it, and when they detect that special packet, they'll just flush their buffers.
</p>
<p>
Let's start with the flush function. It's really quite simple, so I'll just show you the code:
</p><pre>static void packet_queue_flush(PacketQueue *q) {
AVPacketList *pkt, *pkt1;
SDL_LockMutex(q->mutex);
for(pkt = q->first_pkt; pkt != NULL; pkt = pkt1) {
pkt1 = pkt->next;
av_free_packet(&pkt->pkt);
av_freep(&pkt);
}
q->last_pkt = NULL;
q->first_pkt = NULL;
q->nb_packets = 0;
q->size = 0;
SDL_UnlockMutex(q->mutex);
}
</pre>
<p></p>
<p>
Now that the queue is flushed, let's put on our "flush packet." But first we're going to want to define what that is and create it:
</p><pre>AVPacket flush_pkt;
main() {
...
av_init_packet(&flush_pkt);
flush_pkt.data = "FLUSH";
...
}
</pre>
Now we put this packet on the queue:
<pre> } else {
if(is->audioStream >= 0) {
packet_queue_flush(&is->audioq);
packet_queue_put(&is->audioq, &flush_pkt);
}
if(is->videoStream >= 0) {
packet_queue_flush(&is->videoq);
packet_queue_put(&is->videoq, &flush_pkt);
}
}
is->seek_req = 0;
}
</pre>
(This code snippet also continues the code snippet above for <tt>decode_thread</tt>.) We also need to change <tt>packet_queue_put</tt> so that we don't duplicate the special flush packet:
<pre>int packet_queue_put(PacketQueue *q, AVPacket *pkt) {
AVPacketList *pkt1;
if(pkt != &flush_pkt && av_dup_packet(pkt) < 0) {
return -1;
}
</pre>
And then in the audio thread and the video thread, we put this call to <tt>avcodec_flush_buffers</tt> immediately after <tt>packet_queue_get</tt>:
<pre> if(packet_queue_get(&is->audioq, pkt, 1) < 0) {
return -1;
}
if(packet->data == flush_pkt.data) {
avcodec_flush_buffers(is->audio_st->codec);
continue;
}
</pre>
The above code snippet is exactly the same for the video thread, with "audio" being replaced by "video".
<p></p>
<p>
That's it! Go ahead and compile your player:
</p><pre>gcc -o tutorial07 tutorial07.c -lavutil -lavformat -lavcodec -lz -lm`sdl-config --cflags --libs`
</pre>
Try it out! We're almost done; next time we only have one little change to make, and that's to check out a small sampling of the software scaling that ffmpeg provides.
<p></p>
<p>
<em><b>>></b> Software scaling</em>
</p>
<a name="tutorial08.html"></a>
<h2>Tutorial 08: Software Scaling</h2>
<span class="codelink">
Code: tutorial08.c
</span>
<h3>libswscale</h3>
<p>
ffmpeg has recently added a new interface, libswscale to handle image scaling. Whereas before in our player we would use <tt>img_convert</tt> to go from RGB to YUV12, we now use the new interface. This new interface is more modular, faster, and I believe has MMX optimization stuff. In other words, it's the preferred way to do scaling.
</p>
<p>
The basic function we're going to use to scale is <tt>sws_scale</tt>. But first, we're going to have to set up what's called an <tt>SwsContext</tt>. This allows us to compile the conversion we want, and then pass that in later to <tt>sws_scale</tt>. It's kind of like a prepared statement in SQL or a compiled regexp in Python. To prepare this context, we use the <tt>sws_getContext</tt> function, which is going to want our source width and height, our desired width and height, the source format and desired format, along with some other options and flags. Then we use <tt>sws_scale</tt> the same way as <tt>img_convert</tt> except we pass it our SwsContext:
</p><pre>#include <ffmpeg/swscale.h> // include the header!
int queue_picture(VideoState *is, AVFrame *pFrame, double pts) {
static struct SwsContext *img_convert_ctx;
...
if(vp->bmp) {
SDL_LockYUVOverlay(vp->bmp);
dst_pix_fmt = PIX_FMT_YUV420P;
/* point pict at the queue */
pict.data[0] = vp->bmp->pixels[0];
pict.data[1] = vp->bmp->pixels[2];
pict.data[2] = vp->bmp->pixels[1];
pict.linesize[0] = vp->bmp->pitches[0];
pict.linesize[1] = vp->bmp->pitches[2];
pict.linesize[2] = vp->bmp->pitches[1];
// Convert the image into YUV format that SDL uses
if(img_convert_ctx == NULL) {
int w = is->video_st->codec->width;
int h = is->video_st->codec->height;
img_convert_ctx = sws_getContext(w, h,
is->video_st->codec->pix_fmt,
w, h, dst_pix_fmt, SWS_BICUBIC,
NULL, NULL, NULL);
if(img_convert_ctx == NULL) {
fprintf(stderr, "Cannot initialize the conversion context!\n");
exit(1);
}
}
sws_scale(img_convert_ctx, pFrame->data,
pFrame->linesize, 0,
is->video_st->codec->height,
pict.data, pict.linesize);
</pre>
and we have our new scaler in place. Hopefully this gives you a good idea of what libswscale can do.
<p></p>
<p>
That's it! We're done! Go ahead and compile your player:
</p><pre>gcc -o tutorial08 tutorial08.c -lavutil -lavformat -lavcodec -lz -lm `sdl-config --cflags --libs`
</pre>
and enjoy your movie player made in less than 1000 lines of C!
<p></p>
<p>
Of course, there's a lot of things we glanced over that we could add.
</p><p>
<em><b>>></b> What's left?</em>
</p>
<a name="end.html"></a>
<h3>What now?</h3>
<p>So we have a working player, but it's certainly not as nice as it could be. We did a lot of handwaving, and there are a lot of other features we could add:
</p><ul>
<li>Error handling. The error handling in our code is abysmal, and could be handled a lot better.</li>
<li>Pausing. We can't pause the movie, which is admittedly a useful feature. We can do this by making use of an internal paused variable in our big struct that we set when the user pauses. Then our audio, video, and decode threads check for it so they don't output anything. We also use <tt>av_read_play</tt> for network support. It's pretty simple to explain, but it's not obvious to figure out for yourself, <b>so consider this homework</b> if you want to try more. For hints, check out ffplay.c.</li>
<li>Support for video hardware. For a sample of setting this up, check out the Frame Grabbing section in Martin's old tutorial.</li>
<li>Seeking by bytes. If you calculate the seek position by bytes instead of seconds, it is more accurate on video files that have discontiguous timestamps, like VOB files.</li>
<li>Frame dropping. If the video falls too far behind, we should drop the next frame instead of setting a short refresh.</li>
<li>Network support. This video player can't play network streaming video.</li>
<li>Support for raw video like YUV files. There are some options we have to set if our player is to support raw video like YUV files, as we cannot guess the size or time_base.</li>
<li>Fullscreen</li>
<li>Various options, e.g. different pic formats; see ffplay.c for all the command line switches.</li>
<li>Other hand-wavy things; for example, the audio buffer in our struct should be declared aligned.</li>
</ul>
If you want to know more about ffmpeg, we've only covered a portion of it. The next step would be to study how to <i>encode</i> multimedia. A good place to start would be the output_example.c file in the ffmpeg distribution. I may write another tutorial on that, but I might not get around to it.
<p></p>
<p>
Well, I hope this tutorial was instructive and fun. If you have any suggestions, bugs, complaints, accolades, etc., please email me at dranger at gmail dot com.
</p>
<p>
Links:<br>
ffmpeg home page<br>
Martin Bohme's original tutorial<br>
libSDL<br>
SDL Documentation
</p>
<span class="fineprint">This work is licensed under the CreativeCommons Attribution-Share Alike 2.5 License. To view a copy of thislicense, visit http://creativecommons.org/licenses/by-sa/2.5/ or senda letter to Creative Commons, 543 Howard Street, 5th Floor, SanFrancisco, California, 94105, USA.
<br>
<br>
Code examples are based off of FFplay, Copyright (c) 2003 FabriceBellard, and a tutorial by Martin Bohme.
</span>
</pts></pre></t></body></html>
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简介
This is an updated version of Stephen Dranger tutorial on ffmpeg and SDL. http://dranger.com/ffmpeg/
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