How to write bitmaps as frames to Ogg Theora in C\C++? - c++

How to write bitmaps as frames to Ogg Theora in C\C++?
Some Examples with source would be grate!)

The entire solution is a little lengthy to post on here as a code sample, but if you download libtheora from Xiph.org, there is an example png2theora. All of the library functions I am about to mention can be found in the documentation on Xiph.org for theora and ogg.
Call th_info_init() to initialise a th_info structure, then set up you output parameters by assigning the appropriate members in that.
Use that structure in a call to th_encode_alloc() to get an encoder context
Initialise an ogg stream, with ogg_stream_init()
Initialise a blank th_comment structure using th_comment_init
Iterate through the following:
Call th_encode_flushheader with the the encoder context, the blank comment structure and an ogg_packet.
Send the resulting packet to the ogg stream with ogg_stream_packetin()
Until th_encode_flushheader returns 0 (or an error code)
Now, repeatedly call ogg_stream_pageout(), every time writing the page.header and then page.body to an output file, until it returns 0. Now call ogg_stream_flush and write the resulting page to the file.
You can now write frames to the encoder. Here is how I did it:
int theora_write_frame(int outputFd, unsigned long w, unsigned long h, unsigned char *yuv_y, unsigned char *yuv_u, unsigned char *yuv_v, int last)
{
th_ycbcr_buffer ycbcr;
ogg_packet op;
ogg_page og;
unsigned long yuv_w;
unsigned long yuv_h;
/* Must hold: yuv_w >= w */
yuv_w = (w + 15) & ~15;
/* Must hold: yuv_h >= h */
yuv_h = (h + 15) & ~15;
//Fill out the ycbcr buffer
ycbcr[0].width = yuv_w;
ycbcr[0].height = yuv_h;
ycbcr[0].stride = yuv_w;
ycbcr[1].width = yuv_w;
ycbcr[1].stride = ycbcr[1].width;
ycbcr[1].height = yuv_h;
ycbcr[2].width = ycbcr[1].width;
ycbcr[2].stride = ycbcr[1].stride;
ycbcr[2].height = ycbcr[1].height;
if(encoderInfo->pixel_fmt == TH_PF_420)
{
//Chroma is decimated by 2 in both directions
ycbcr[1].width = yuv_w >> 1;
ycbcr[2].width = yuv_w >> 1;
ycbcr[1].height = yuv_h >> 1;
ycbcr[2].height = yuv_h >> 1;
}else if(encoderInfo->pixel_fmt == TH_PF_422)
{
ycbcr[1].width = yuv_w >> 1;
ycbcr[2].width = yuv_w >> 1;
}else if(encoderInfo->pixel_fmt != TH_PF_422)
{
//Then we have an unknown pixel format
//We don't know how long the arrays are!
fprintf(stderr, "[theora_write_frame] Unknown pixel format in writeFrame!\n");
return -1;
}
ycbcr[0].data = yuv_y;
ycbcr[1].data = yuv_u;
ycbcr[2].data = yuv_v;
/* Theora is a one-frame-in,one-frame-out system; submit a frame
for compression and pull out the packet */
if(th_encode_ycbcr_in(encoderContext, ycbcr)) {
fprintf(stderr, "[theora_write_frame] Error: could not encode frame\n");
return -1;
}
if(!th_encode_packetout(encoderContext, last, &op)) {
fprintf(stderr, "[theora_write_frame] Error: could not read packets\n");
return -1;
}
ogg_stream_packetin(&theoraStreamState, &op);
ssize_t bytesWritten = 0;
int pagesOut = 0;
while(ogg_stream_pageout(&theoraStreamState, &og)) {
pagesOut ++;
bytesWritten = write(outputFd, og.header, og.header_len);
if(bytesWritten != og.header_len)
{
fprintf(stderr, "[theora_write_frame] Error: Could not write to file\n");
return -1;
}
bytesWritten = write(outputFd, og.body, og.body_len);
if(bytesWritten != og.body_len)
{
bytesWritten = fprintf(stderr, "[theora_write_frame] Error: Could not write to file\n");
return -1;
}
}
return pagesOut;
}
Where encoderInfo is the th_info structure used to initialise the encoder (static in the data section for me).
On your last frame, setting the last frame on th_encode_packetout() will make sure the stream terminates properly.
Once your done, just make sure to clean up (closing fds mainly). th_info_clear() will clear the th_info structure, and th_encode_free() will free your encoder context.
Obviously, you'll need to convert your bitmap into YUV planes before you can pass them to theora_write_frame().
Hope this is of some help. Good luck!

Here's the libtheora API and example code.
Here's a micro howto that shows how to use the theora binaries. As the encoder reads raw, uncompressed 'yuv4mpeg' data for video you could use that from your app, too by piping the video frames to the encoder.

Related

Decoding with OGG/Vorbis gives no sound

I'd like to play an Ogg/Vorbis audio/video file, but right now I can't get to read audio from a file.
My algorithm to read audio is:
Initialize required structures:
vorbis_info info;
vorbis_comment comment;
vorbis_dsp_state dsp;
vorbis_block block;
vorbis_info_init(&info);
vorbis_comment_init(&comment);
Read headers:
Call vorbis_synthesis_headerin(&info, &comment, packet); until it returns OV_ENOTVORBIS
vorbis_synthesis_init(&dsp, &info);
vorbis_block_init(&dsp, &block);
Pass the first non-header packet to function below
Parse packets, do it until audioReady == READY
putPacket(ogg_packet *packet) {
int ret;
ret = vorbis_synthesis(&block, packet);
if( ret == 0 ) {
ret = vorbis_synthesis_blockin(&dsp, &block);
audioReady = (ret == 0) ? READY : NOT_READY;
} else {
audioReady = NOT_READY;
}
}
Read audio data:
float** rawData = nullptr;
readSamples = vorbis_synthesis_pcmout(&dsp, &rawData);
if( readSamples == 0 ) {
audioReady = NOT_READY;
return;
}
int16_t* newData = new int16_t[readSamples * getChannels()];
int16_t* dst = newData;
for(unsigned int i=0; i<readSamples; ++i) {
for(unsigned char ch=0; ch<getChannels(); ++ch) {
*(dst++) = math::clamp<int16_t>(rawData[ch][i]*32767 + 0.5f, -32767, 32767);
}
}
audioData.push_back({readSamples * getChannels() , newData});
vorbis_synthesis_read(&dsp, static_cast<int>(readSamples));
audioReady = NOT_READY;
This is where it gets wrong: after examining the newData contents it is revealed that it contains a very silent sound. I doubt if it is the right data which means somewhere along my algorithm I did something wrong.
I tried to find some examples of similar programs, but all I got are sources with very spaghetti-like code, which seems to do the same algorithm like mine, yet they do their job. (There is one off such library: https://github.com/icculus/theoraplay )
Is there any reason why I'm getting (almost) silence in my application?
PS: If you are wondering if I might getting OGG packets wrong, then I assure you this part of my code is working right, as I'm also reading video data from the same file, using the same code and it shows the video right.
I've found it: during reading packets I assumed that one Ogg Page = one Ogg packet. I's wrong: for audio one page can contain many packets. To read it properly one has to make a code like:
do{
putPacket(&packet);
}while( ogg_stream_packetout(&state, &packet) == 1 );
I did this mistake because for video packets (which I did first) a page contains only one packet.

Sound playback using FFmpeg and libsoundio in c++

I am trying to make a video player desktop application in c++ using primarily FFmpeg and Qt6. As of for now, I can decode and play video frames correctly at the right speed, that is not a problem. I am now trying to get to playback audio, which is much harder than I expected it to be. I am using libsoundio for my audio library but the documentation is really poor and there are not many examples/tutorials on it. I am also a beginner when it comes to audio programming, although I understand the basics. First off, if anyone can recommend an audio library for this type of job let me know, but I would like to use open source libraries. Anyways, here is how I decode my audio data with FFmpeg. I'm not sure if I am doing it correctly as I could barely find documentation on that as well...
I have a struct that contains all the information which is initiated through a function:
struct VideoReader
{
bool valid;
int width, height;
int video_stream_index;
int audio_stream_index;
AVRational time_base;
AVFormatContext* av_format_ctx;
AVCodecContext* av_vi_codec_ctx;
AVCodecContext* av_au_codec_ctx;
AVPacket* packet;
AVFrame* frame;
SwsContext* sws_ctx;
SwrContext* swr_ctx;
};
The function that initiates it is quite long and is not necessary to share but it populates all those values except for the sws_ctx and the swr_ctx.
Here is how I decode packets, this function is simplified, I left the video decoding out of it, ill take care of syncing once I can properly playback audio:
bool video_reader_read_au_frame(VideoReader *video_reader, unsigned char **frame_buffer)
{
// Unpack video_reader
auto& av_format_ctx = video_reader->av_format_ctx;
auto& av_codec_ctx = video_reader->av_au_codec_ctx;
auto& av_packet = video_reader->packet;
auto& av_frame = video_reader->frame;
auto& swr_ctx = video_reader->swr_ctx;
int& audio_stream_index = video_reader->audio_stream_index;
// Decode the video frame data
int response;
while (av_read_frame(av_format_ctx, av_packet) >= 0)
{
last_frame = false;
if (av_packet->stream_index != audio_stream_index)
{
av_packet_unref(av_packet);
continue;
}
response = avcodec_send_packet(av_codec_ctx, av_packet);
if (response < 0)
{
Logger::error("Could not decode packet.");
return false;
}
response = avcodec_receive_frame(av_codec_ctx, av_frame);
if (response == AVERROR(EAGAIN) || response == AVERROR_EOF)
{
av_packet_unref(av_packet);
continue;
}
else if (response < 0)
{
Logger::error("Could not decode packet.");
return false;
}
av_packet_unref(av_packet);
break;
}
// Initialize SwrContext
if (!swr_ctx) {
swr_ctx = swr_alloc_set_opts(nullptr,
av_codec_ctx->channel_layout, AV_SAMPLE_FMT_FLT,
av_codec_ctx->sample_rate, av_codec_ctx->channel_layout,
av_codec_ctx->sample_fmt, av_codec_ctx->sample_rate,
0, nullptr);
if (!swr_ctx)
{
Logger::error("Could not create SwrContext.");
return false;
}
if (swr_init(swr_ctx) < 0)
{
Logger::error("Could not initialize SwrContext.");
return false;
}
}
const int MAX_BUFFER_SIZE = av_samples_get_buffer_size(nullptr, av_frame->channels, av_frame->nb_samples, AV_SAMPLE_FMT_FLT, 1);
*frame_buffer = (unsigned char*)av_malloc(MAX_BUFFER_SIZE);
swr_convert(swr_ctx, frame_buffer, av_frame->nb_samples,
(const unsigned char**)av_frame->data, av_frame->nb_samples);
av_frame_unref(av_frame);
return true;
}
Here is how I would normally call this function:
VideoReader vr{};
if(!video_reader_open(&vr, "C:/Path/to/file.mp4"))
{
Logger::error("Could not initialize VideoReader.");
return 1;
}
unsigned char* buffer;
if(!video_reader_read_au_frame(&vr, &buffer))
{
Logger::error("Could not read audio data.");
return 1;
}
play_audio(&buffer); <-- Find a way to play audio once buffer has data in it
video_reader_close(&vr);
return 0;
Obviously I will loop over video_reader_read_au_frame(&vr, &buffer) to playback the whole video.
I believe my code puts the samples from the decoded frame in buffer, but I am really not sure.. I am unsure as well if I need to convert to AV_SAMPLE_FMT_FLT audio format or something else or just leave it as it is. For libsoundio, I kind of understand this example: http://libsound.io/ but I'm not sure I fully understand how this library works, especially the callback function. I know I have to pass buffer in outstream->userdata as a void pointer, but I don't know how to use it in the callback function. Any help or guidance would be greatly appreciated. Note that later on in this project I might want to send this data over a network to play the video on another computer in sync.

Oboe Async Audio Extraction

I am trying to build a NDK based c++ low latancy audio player which will encounter three operations for multiple audios.
Play from assets.
Stream from an online source.
Play from local device storage.
From one of the Oboe samples provided by Google, I added another function to the class NDKExtractor.cpp to extract a URL based audio and render it to audio device while reading from source at the same time.
int32_t NDKExtractor::decode(char *file, uint8_t *targetData, AudioProperties targetProperties) {
LOGD("Using NDK decoder: %s",file);
// Extract the audio frames
AMediaExtractor *extractor = AMediaExtractor_new();
//using this method instead of AMediaExtractor_setDataSourceFd() as used for asset files in the rythem game example
media_status_t amresult = AMediaExtractor_setDataSource(extractor, file);
if (amresult != AMEDIA_OK) {
LOGE("Error setting extractor data source, err %d", amresult);
return 0;
}
// Specify our desired output format by creating it from our source
AMediaFormat *format = AMediaExtractor_getTrackFormat(extractor, 0);
int32_t sampleRate;
if (AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_SAMPLE_RATE, &sampleRate)) {
LOGD("Source sample rate %d", sampleRate);
if (sampleRate != targetProperties.sampleRate) {
LOGE("Input (%d) and output (%d) sample rates do not match. "
"NDK decoder does not support resampling.",
sampleRate,
targetProperties.sampleRate);
return 0;
}
} else {
LOGE("Failed to get sample rate");
return 0;
};
int32_t channelCount;
if (AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_CHANNEL_COUNT, &channelCount)) {
LOGD("Got channel count %d", channelCount);
if (channelCount != targetProperties.channelCount) {
LOGE("NDK decoder does not support different "
"input (%d) and output (%d) channel counts",
channelCount,
targetProperties.channelCount);
}
} else {
LOGE("Failed to get channel count");
return 0;
}
const char *formatStr = AMediaFormat_toString(format);
LOGD("Output format %s", formatStr);
const char *mimeType;
if (AMediaFormat_getString(format, AMEDIAFORMAT_KEY_MIME, &mimeType)) {
LOGD("Got mime type %s", mimeType);
} else {
LOGE("Failed to get mime type");
return 0;
}
// Obtain the correct decoder
AMediaCodec *codec = nullptr;
AMediaExtractor_selectTrack(extractor, 0);
codec = AMediaCodec_createDecoderByType(mimeType);
AMediaCodec_configure(codec, format, nullptr, nullptr, 0);
AMediaCodec_start(codec);
// DECODE
bool isExtracting = true;
bool isDecoding = true;
int32_t bytesWritten = 0;
while (isExtracting || isDecoding) {
if (isExtracting) {
// Obtain the index of the next available input buffer
ssize_t inputIndex = AMediaCodec_dequeueInputBuffer(codec, 2000);
//LOGV("Got input buffer %d", inputIndex);
// The input index acts as a status if its negative
if (inputIndex < 0) {
if (inputIndex == AMEDIACODEC_INFO_TRY_AGAIN_LATER) {
// LOGV("Codec.dequeueInputBuffer try again later");
} else {
LOGE("Codec.dequeueInputBuffer unknown error status");
}
} else {
// Obtain the actual buffer and read the encoded data into it
size_t inputSize;
uint8_t *inputBuffer = AMediaCodec_getInputBuffer(codec, inputIndex,
&inputSize);
//LOGV("Sample size is: %d", inputSize);
ssize_t sampleSize = AMediaExtractor_readSampleData(extractor, inputBuffer,
inputSize);
auto presentationTimeUs = AMediaExtractor_getSampleTime(extractor);
if (sampleSize > 0) {
// Enqueue the encoded data
AMediaCodec_queueInputBuffer(codec, inputIndex, 0, sampleSize,
presentationTimeUs,
0);
AMediaExtractor_advance(extractor);
} else {
LOGD("End of extractor data stream");
isExtracting = false;
// We need to tell the codec that we've reached the end of the stream
AMediaCodec_queueInputBuffer(codec, inputIndex, 0, 0,
presentationTimeUs,
AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM);
}
}
}
if (isDecoding) {
// Dequeue the decoded data
AMediaCodecBufferInfo info;
ssize_t outputIndex = AMediaCodec_dequeueOutputBuffer(codec, &info, 0);
if (outputIndex >= 0) {
// Check whether this is set earlier
if (info.flags & AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM) {
LOGD("Reached end of decoding stream");
isDecoding = false;
} else {
// Valid index, acquire buffer
size_t outputSize;
uint8_t *outputBuffer = AMediaCodec_getOutputBuffer(codec, outputIndex,
&outputSize);
/*LOGV("Got output buffer index %d, buffer size: %d, info size: %d writing to pcm index %d",
outputIndex,
outputSize,
info.size,
m_writeIndex);*/
// copy the data out of the buffer
memcpy(targetData + bytesWritten, outputBuffer, info.size);
bytesWritten += info.size;
AMediaCodec_releaseOutputBuffer(codec, outputIndex, false);
}
} else {
// The outputIndex doubles as a status return if its value is < 0
switch (outputIndex) {
case AMEDIACODEC_INFO_TRY_AGAIN_LATER:
LOGD("dequeueOutputBuffer: try again later");
break;
case AMEDIACODEC_INFO_OUTPUT_BUFFERS_CHANGED:
LOGD("dequeueOutputBuffer: output buffers changed");
break;
case AMEDIACODEC_INFO_OUTPUT_FORMAT_CHANGED:
LOGD("dequeueOutputBuffer: output outputFormat changed");
format = AMediaCodec_getOutputFormat(codec);
LOGD("outputFormat changed to: %s", AMediaFormat_toString(format));
break;
}
}
}
}
// Clean up
AMediaFormat_delete(format);
AMediaCodec_delete(codec);
AMediaExtractor_delete(extractor);
return bytesWritten;
}
Now the problem i am facing is that this code it first extracts all the audio data saves it into a buffer which then becomes part of AFileDataSource which i derived from DataSource class in the same sample.
And after its done extracting the whole file it plays by calling the onAudioReady() for Oboe AudioStreamBuilder.
What I need is to play as it streams the chunk of audio buffer.
Optional Query: Also aside from the question it blocks the UI even though i created a foreground service to communicate with the NDK functions to execute this code. Any thoughts on this?
You probably solved this already, but for future readers...
You need a FIFO buffer to store the decoded audio. You can use the Oboe's FIFO buffer e.g. oboe::FifoBuffer.
You can have a low/high watermark for the buffer and a state machine, so you start decoding when the buffer is almost empty and you stop decoding when it's full (you'll figure out the other states that you need).
As a side note, I implemented such player only to find at some later time, that the AAC codec is broken on some devices (Xiaomi and Amazon come to mind), so I had to throw away the AMediaCodec/AMediaExtractor parts and use an AAC library instead.
You have to implement a ringBuffer (or use the one implemented in the oboe example LockFreeQueue.h) and copy the data on buffers that you send on the ringbuffer from the extracting thread. On the other end of the RingBuffer, the audio thread will get that data from the queue and copy it to the audio buffer. This will happen on onAudioReady(oboe::AudioStream *oboeStream, void *audioData, int32_t numFrames) callback that you have to implement in your class (look oboe docs). Be sure to follow all the good practices on the Audio thread (don't allocate/deallocate memory there, no mutexes and no file I/O etc.)
Optional query: A service doesn't run in a separate thread, so obviously if you call it from UI thread it blocks the UI. Look at other types of services, there you can have IntentService or a service with a Messenger that will launch a separate thread on Java, or you can create threads in C++ side using std::thread

how to use self-defined inputSamples for trasforming pcm to aac with facc

I'm trying to transform a live stream with g726 and h264 to mp4. I decode g726 to pcm then use faac to encode pcm to aac. Every g726 audio packet I receive is 320 bytes. After decoding, the pcm size is 1280 bytes, so the sample number is 640. But the inputSamples which faacEncOpen gives me is 1024, and my inputFormat is FAAC_INPUT_16BIT. When I pass 640 to faacEncEncode, the sound is not good at all. Does anyone know how to fix this. Thanks in advance!
// (1) Open FAAC engine
hEncoder = faacEncOpen(nSampleRate, nChannels, &nInputSamples, &nMaxOutputBytes); // nInputSamples the function returns is 1024
if(hEncoder == NULL)
{
printf("[ERROR] Failed to call faacEncOpen()\n");
return -1;
}
nInputSamples = 640;// here overwrites the input samples returned from faacEncOpen
nPCMBufferSize = nInputSamples * nPCMBitSize / 8; // nPCMBitSize is 16
pbPCMBuffer = new BYTE [nPCMBufferSize];
pbAACBuffer = new BYTE [nMaxOutputBytes];
// (2.1) Get current encoding configuration
pConfiguration = faacEncGetCurrentConfiguration(hEncoder);
pConfiguration->inputFormat = FAAC_INPUT_16BIT;
// (2.2) Set encoding configuration
nRet = faacEncSetConfiguration(hEncoder, pConfiguration);
for(int i = 0; 1; i++)
{
nBytesRead = fread(pbPCMBuffer, 1, nPCMBufferSize, fpIn);
nInputSamples = nBytesRead * 8 / nPCMBitSize;
// (3) Encode
nRet = faacEncEncode(
hEncoder, (int*) pbPCMBuffer, nInputSamples, pbAACBuffer, nMaxOutputBytes);
fwrite(pbAACBuffer, 1, nRet, fpOut);
printf("%d: faacEncEncode returns %d\n", i, nRet);
if(nBytesRead <= 0)
{
break;
}
}

FFmpeg + OpenAL - playback streaming sound from video won't work

I am decoding an OGG video (theora & vorbis as codecs) and want to show it on the screen (using Ogre 3D) while playing its sound. I can decode the image stream just fine and the video plays perfectly with the correct frame rate, etc.
However, I cannot get the sound to play at all with OpenAL.
Edit: I managed to make the playing sound resemble the actual audio in the video at least somewhat. Updated sample code.
Edit 2: I was able to get "almost" correct sound now. I had to set OpenAL to use AL_FORMAT_STEREO_FLOAT32 (after initializing the extension) instead of just STEREO16. Now the sound is "only" extremely high pitched and stuttering, but at the correct speed.
Here is how I decode audio packets (in a background thread, the equivalent works just fine for the image stream of the video file):
//------------------------------------------------------------------------------
int decodeAudioPacket( AVPacket& p_packet, AVCodecContext* p_audioCodecContext, AVFrame* p_frame,
FFmpegVideoPlayer* p_player, VideoInfo& p_videoInfo)
{
// Decode audio frame
int got_frame = 0;
int decoded = avcodec_decode_audio4(p_audioCodecContext, p_frame, &got_frame, &p_packet);
if (decoded < 0)
{
p_videoInfo.error = "Error decoding audio frame.";
return decoded;
}
// Frame is complete, store it in audio frame queue
if (got_frame)
{
int bufferSize = av_samples_get_buffer_size(NULL, p_audioCodecContext->channels, p_frame->nb_samples,
p_audioCodecContext->sample_fmt, 0);
int64_t duration = p_frame->pkt_duration;
int64_t dts = p_frame->pkt_dts;
if (staticOgreLog)
{
staticOgreLog->logMessage("Audio frame bufferSize / duration / dts: "
+ boost::lexical_cast<std::string>(bufferSize) + " / "
+ boost::lexical_cast<std::string>(duration) + " / "
+ boost::lexical_cast<std::string>(dts), Ogre::LML_NORMAL);
}
// Create the audio frame
AudioFrame* frame = new AudioFrame();
frame->dataSize = bufferSize;
frame->data = new uint8_t[bufferSize];
if (p_frame->channels == 2)
{
memcpy(frame->data, p_frame->data[0], bufferSize >> 1);
memcpy(frame->data + (bufferSize >> 1), p_frame->data[1], bufferSize >> 1);
}
else
{
memcpy(frame->data, p_frame->data, bufferSize);
}
double timeBase = ((double)p_audioCodecContext->time_base.num) / (double)p_audioCodecContext->time_base.den;
frame->lifeTime = duration * timeBase;
p_player->addAudioFrame(frame);
}
return decoded;
}
So, as you can see, I decode the frame, memcpy it to my own struct, AudioFrame. Now, when the sound is played, I use these audio frame like this:
int numBuffers = 4;
ALuint buffers[4];
alGenBuffers(numBuffers, buffers);
ALenum success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error on alGenBuffers : " + Ogre::StringConverter::toString(success) + alGetString(success));
return;
}
// Fill a number of data buffers with audio from the stream
std::vector<AudioFrame*> audioBuffers;
std::vector<unsigned int> audioBufferSizes;
unsigned int numReturned = FFMPEG_PLAYER->getDecodedAudioFrames(numBuffers, audioBuffers, audioBufferSizes);
// Assign the data buffers to the OpenAL buffers
for (unsigned int i = 0; i < numReturned; ++i)
{
alBufferData(buffers[i], _streamingFormat, audioBuffers[i]->data, audioBufferSizes[i], _streamingFrequency);
success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error on alBufferData : " + Ogre::StringConverter::toString(success) + alGetString(success)
+ " size: " + Ogre::StringConverter::toString(audioBufferSizes[i]));
return;
}
}
// Queue the buffers into OpenAL
alSourceQueueBuffers(_source, numReturned, buffers);
success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error queuing streaming buffers: " + Ogre::StringConverter::toString(success) + alGetString(success));
return;
}
}
alSourcePlay(_source);
The format and frequency I give to OpenAL are AL_FORMAT_STEREO_FLOAT32 (it is a stereo sound stream, and I did initialize the FLOAT32 extension) and 48000 (which is the sample rate of the AVCodecContext of the audio stream).
And during playback, I do the following to refill OpenAL's buffers:
ALint numBuffersProcessed;
// Check if OpenAL is done with any of the queued buffers
alGetSourcei(_source, AL_BUFFERS_PROCESSED, &numBuffersProcessed);
if(numBuffersProcessed <= 0)
return;
// Fill a number of data buffers with audio from the stream
std::vector<AudiFrame*> audioBuffers;
std::vector<unsigned int> audioBufferSizes;
unsigned int numFilled = FFMPEG_PLAYER->getDecodedAudioFrames(numBuffersProcessed, audioBuffers, audioBufferSizes);
// Assign the data buffers to the OpenAL buffers
ALuint buffer;
for (unsigned int i = 0; i < numFilled; ++i)
{
// Pop the oldest queued buffer from the source,
// fill it with the new data, then re-queue it
alSourceUnqueueBuffers(_source, 1, &buffer);
ALenum success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error Unqueuing streaming buffers: " + Ogre::StringConverter::toString(success));
return;
}
alBufferData(buffer, _streamingFormat, audioBuffers[i]->data, audioBufferSizes[i], _streamingFrequency);
success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error on re- alBufferData: " + Ogre::StringConverter::toString(success));
return;
}
alSourceQueueBuffers(_source, 1, &buffer);
success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error re-queuing streaming buffers: " + Ogre::StringConverter::toString(success) + " "
+ alGetString(success));
return;
}
}
// Make sure the source is still playing,
// and restart it if needed.
ALint playStatus;
alGetSourcei(_source, AL_SOURCE_STATE, &playStatus);
if(playStatus != AL_PLAYING)
alSourcePlay(_source);
As you can see, I do quite heavy error checking. But I do not get any errors, neither from OpenAL nor from FFmpeg.
Edit: What I hear somewhat resembles the actual audio from the video, but VERY high pitched and stuttering VERY much. Also, it seems to be playing on top of TV noise. Very strange. Plus, it is playing much slower than the correct audio would.
Edit: 2 After using AL_FORMAT_STEREO_FLOAT32, the sound plays at the correct speed, but is still very high pitched and stuttering (though less than before).
The video itself is not broken, it can be played fine on any player. OpenAL can also play *.way files just fine in the same application, so it is also working.
Any ideas what could be wrong here or how to do this correctly?
My only guess is that somehow, FFmpeg's decode function does not produce data OpenGL can read. But this is as far as the FFmpeg decode example goes, so I don't know what's missing. As I understand it, the decode_audio4 function decodes the frame to raw data. And OpenAL should be able to work with RAW data (or rather, doesn't work with anything else).
So, I finally figured out how to do it. Gee, what a mess. It was a hint from a user on the libav-users mailing list that put me on the correct path.
Here are my mistakes:
Using the wrong format in the alBufferData function. I used AL_FORMAT_STEREO16 (as that is what every single streaming example with OpenAL uses). I should have used AL_FORMAT_STEREO_FLOAT32, as the video I stream is Ogg and vorbis is stored in floating points. And using swr_convert to convert from AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_S16 just crashes. No idea why.
Not using swr_convert to convert the decoded audio frame to the target format. After I was trying to use swr_convert to convert from FLTP to S16, and it would simply crash without a reason given, I assumed it was broken. But after figuring out my first mistake, I tried again, converting from FLTP to FLT (non-planar) and then it worked! So OpenAL uses interleaved format, not planar. Good to know.
So here is the decodeAudioPacket function that is working for me with Ogg video, vorbis audio stream:
int decodeAudioPacket( AVPacket& p_packet, AVCodecContext* p_audioCodecContext, AVFrame* p_frame,
SwrContext* p_swrContext, uint8_t** p_destBuffer, int p_destLinesize,
FFmpegVideoPlayer* p_player, VideoInfo& p_videoInfo)
{
// Decode audio frame
int got_frame = 0;
int decoded = avcodec_decode_audio4(p_audioCodecContext, p_frame, &got_frame, &p_packet);
if (decoded < 0)
{
p_videoInfo.error = "Error decoding audio frame.";
return decoded;
}
if(decoded <= p_packet.size)
{
/* Move the unread data to the front and clear the end bits */
int remaining = p_packet.size - decoded;
memmove(p_packet.data, &p_packet.data[decoded], remaining);
av_shrink_packet(&p_packet, remaining);
}
// Frame is complete, store it in audio frame queue
if (got_frame)
{
int outputSamples = swr_convert(p_swrContext,
p_destBuffer, p_destLinesize,
(const uint8_t**)p_frame->extended_data, p_frame->nb_samples);
int bufferSize = av_get_bytes_per_sample(AV_SAMPLE_FMT_FLT) * p_videoInfo.audioNumChannels
* outputSamples;
int64_t duration = p_frame->pkt_duration;
int64_t dts = p_frame->pkt_dts;
if (staticOgreLog)
{
staticOgreLog->logMessage("Audio frame bufferSize / duration / dts: "
+ boost::lexical_cast<std::string>(bufferSize) + " / "
+ boost::lexical_cast<std::string>(duration) + " / "
+ boost::lexical_cast<std::string>(dts), Ogre::LML_NORMAL);
}
// Create the audio frame
AudioFrame* frame = new AudioFrame();
frame->dataSize = bufferSize;
frame->data = new uint8_t[bufferSize];
memcpy(frame->data, p_destBuffer[0], bufferSize);
double timeBase = ((double)p_audioCodecContext->time_base.num) / (double)p_audioCodecContext->time_base.den;
frame->lifeTime = duration * timeBase;
p_player->addAudioFrame(frame);
}
return decoded;
}
And here is how I initialize the context and the destination buffer:
// Initialize SWR context
SwrContext* swrContext = swr_alloc_set_opts(NULL,
audioCodecContext->channel_layout, AV_SAMPLE_FMT_FLT, audioCodecContext->sample_rate,
audioCodecContext->channel_layout, audioCodecContext->sample_fmt, audioCodecContext->sample_rate,
0, NULL);
int result = swr_init(swrContext);
// Create destination sample buffer
uint8_t** destBuffer = NULL;
int destBufferLinesize;
av_samples_alloc_array_and_samples( &destBuffer,
&destBufferLinesize,
videoInfo.audioNumChannels,
2048,
AV_SAMPLE_FMT_FLT,
0);