Develop Reference

Android encoded stream (H264/AAC) audio does not play in Flash Player

I'm working on adding a live broadcasting feature to an Android app. I do so through RTMP and make use of the DailyMotion Android SDK, which in turn makes use of Kickflip.
Everything works perfect, except for the playback of the audio on the website (which makes use of Flash). The audio does work in VLC, so it seems to be an issue with Flash being unable to decode the AAC audio.
For the audio I instantiate an encoder with the "audio/mp4a-latm" mime type. The Android developer docs state the following about this mime type: "audio/mp4a-latm" - AAC audio (note, this is raw AAC packets, not packaged in LATM!). I expect that my problem lies here, but yet I have not been able to find a solution for it.
Pretty much all my research, including this SO question about the matter pointed me in the direction of adding an ADTS header to the audio byte array. That results in the following code in the writeSampleData method:
boolean isHeader = false;
if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != 0) {
isHeader = true;
} else {
pts = bufferInfo.presentationTimeUs - mFirstPts;
}
if (mFirstPts != -1 && pts >= 0) {
pts /= 1000;
byte data[] = new byte[bufferInfo.size + 7];
addADTStoPacket(data, bufferInfo.size + 7);
encodedData.position(bufferInfo.offset);
encodedData.get(data, 7, bufferInfo.size);
addDataPacket(new AudioPacket(data, isHeader, pts, mAudioFirstByte));
}
The addADTStoPacket method is identical to the one in the above mentioned SO post, but I will show it here regardless:
private void addADTStoPacket(byte[] packet, int packetLen) {
int profile = 2; //AAC LC
//39=MediaCodecInfo.CodecProfileLevel.AACObjectELD;
int freqIdx = 4; //44.1KHz
int chanCfg = 1; //CPE
// fill in ADTS data
packet[0] = (byte)0xFF;
packet[1] = (byte)0xF9;
packet[2] = (byte)(((profile-1)<<6) + (freqIdx<<2) +(chanCfg>>2));
packet[3] = (byte)(((chanCfg&3)<<6) + (packetLen>>11));
packet[4] = (byte)((packetLen&0x7FF) >> 3);
packet[5] = (byte)(((packetLen&7)<<5) + 0x1F)
packet[6] = (byte)0xFC;
}
The variables in the above method match the settings I have configured in the application, so I'm pretty sure that's fine.
The data is written to the output stream in the following method of the AudioPacket class:
#Override
public void writePayload(OutputStream outputStream) throws IOException {
outputStream.write(mFirstByte);
outputStream.write(mIsAudioSpecificConfic ? 0 : 1);
outputStream.write(mData);
}
Am I missing something here? I could present more code if necessary, but I think this covers the most related parts. Thanks in advance and I really hope someone is able to help, I've been stuck for a couple of days now...

OpenSL: What's causing static when loading wav files through AAsset_read?

I'm working on a native android project and trying to use OpenSL to play some audio effects. Working from the native audio sample project VisualGDB provides, I've written the code posted below.
Near the end, you can see I have commented a line that stores the contents of a variable called hello in the buffer to the destination. hello comes from the sample project, and contains about 700 lines of character bytes like this:
"\x02\x00\x01\x00\xff\xff\x09\x00\x0c\x00\x10\x00\x07\x00\x07\x00"
which make an audio file of someone saying "hello". When reading that byte data into the stream, my code works fine and I hear "hello" when I run the application. When I read from wav file to play the asset I want, however, I only hear static. The size of the data buffer is the same as the size of the file, so it appears it's being read in properly. The static plays for the duration of the wav file (or very close to it).
I really know nothing about data formats or audio programming. I've tried tweaking the format_pcm variables some with different enum values, but had no success. Using a tool called GSpot I found on The Internet, I know the following about the audio file I'm trying to play:
File Size: 557 KB (570,503 bytes) (this is the same size as the data buffer
AAsset_read returns
Codec: PCM Audio
Sample rate: 48000Hz
Bit rate: 1152 kb/s
Channels: 1
Any help or direction would be greatly appreciated.
SLDataLocator_AndroidSimpleBufferQueue loc_bufq = { SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 1 };
SLDataFormat_PCM format_pcm;
format_pcm.formatType = SL_DATAFORMAT_PCM;
format_pcm.numChannels = 1;
format_pcm.samplesPerSec = SL_SAMPLINGRATE_48;// SL_SAMPLINGRATE_8;
format_pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_8; // SL_PCMSAMPLEFORMAT_FIXED_16;
format_pcm.containerSize = 16;
format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
SLDataSource audioSrc = { &loc_bufq, &format_pcm };
// configure audio sink
SLDataLocator_OutputMix loc_outmix = { SL_DATALOCATOR_OUTPUTMIX, manager->GetOutputMixObject() };
SLDataSink audioSnk = { &loc_outmix, NULL };
//create audio player
const SLInterfaceID ids[3] = { SL_IID_BUFFERQUEUE, SL_IID_EFFECTSEND, SL_IID_VOLUME };
const SLboolean req[3] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE };
SLEngineItf engineEngine = manager->GetEngine();
result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk,
3, ids, req);
// realize the player
result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
// get the play interface
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay);
// get the buffer queue interface
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_BUFFERQUEUE,
&bqPlayerBufferQueue);
// register callback on the buffer queue
result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, NULL);
// get the effect send interface
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_EFFECTSEND,
&bqPlayerEffectSend);
// get the volume interface
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_VOLUME, &bqPlayerVolume);
// set the player's state to playing
result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
uint8* pOutBytes = nullptr;
uint32 outSize = 0;
result = MyFileManager::GetInstance()->OpenFile(m_strAbsolutePath, (void**)&pOutBytes, &outSize, true);
const char* filename = m_strAbsolutePath->GetUTF8String();
result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, pOutBytes, outSize);
// result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, hello, sizeof(hello));
if (SL_RESULT_SUCCESS != result) {
return JNI_FALSE;
}

Several things were to blame. The format of the wave files I was testing with was not what the specification described. There seemed to be a lot of empty data after the first chunk of header data. Also, the buffer that needs to be passed to the queue needs to be a char* of just the wav data, not the header. I'd wrongly assumed the queue parsed the header out.

FFMPEG x264 encoder Android

I've compiled an FFMPEG library for use on Android with libx264 and using the NDK.
I want to encode an MPEG video file however the application is failing when opening the encoder codec, in avcodec_open2.
The FFMPEG logs I receive from avcodec_open2 are below with the function returning -22.
Picture size %ux%u is invalid.
ignoring invalid width/height values
Specified pix_fmt is not supported
On windows this code works fine, it's only on Android that there is a failure. Any ides why this would fail on Android?
if (!(codec = avcodec_find_encoder(AV_CODEC_ID_MPEG1VIDEO)))
{
return -1;
}
//Allocate context based on codec
if (!(context = avcodec_alloc_context3(codec)))
{
return -2;
}
//Setup Context
// put sample parameters
context->bit_rate = 4000000;
// resolution must be a multiple of two
context->width = 1280;
context->height = 720;
// frames per second
context->time_base = (AVRational){1,25};
context->inter_quant_bias = 96;
context->gop_size = 10;
context->max_b_frames = 1;
//IDs
context->pix_fmt = AV_PIX_FMT_YUV420P;
context->codec_id = AV_CODEC_ID_MPEG1VIDEO;
context->codec_type = AVMEDIA_TYPE_VIDEO;
if (AV_CODEC_ID_MPEG1VIDEO == AV_CODEC_ID_H264)
{
av_opt_set(context->priv_data, "preset", "slow", 0);
}
if ((result = avcodec_open2(context, codec, NULL)) < 0)
{
//Failed opening Codec!
}

This problem was caused by building FFMPEG with outdated source code.
I got the most recent source from https://www.ffmpeg.org/ and compiled it in the same way and the new library works fine.
Note: I hadn't considered the full implications regarding licenses of using libx264. I've since dropped it.

Android MediaCodec decoder input/output frame count

I'm working on video transcoding in Android, and using the standard method as these samples to extract/decode a video. I test the same process on different devices with different video devices, and I found a problem on the frame count of decoder input/output.
For some timecode issues as in this question, I use a queue to record the extracted video samples, and check the queue when I got a decoder frame output, like the following codes:
(I omit the encoding-related codes to make it clearer)
Queue<Long> sample_time_queue = new LinkedList<Long>();
....
// in transcoding loop
if (is_decode_input_done == false)
{
int decode_input_index = decoder.dequeueInputBuffer(TIMEOUT_USEC);
if (decode_input_index >= 0)
{
ByteBuffer decoder_input_buffer = decode_input_buffers[decode_input_index];
int sample_size = extractor.readSampleData(decoder_input_buffer, 0);
if (sample_size < 0)
{
decoder.queueInputBuffer(decode_input_index, 0, 0, 0, MediaCodec.BUFFER_FLAG_END_OF_STREAM);
is_decode_input_done = true;
}
else
{
long sample_time = extractor.getSampleTime();
decoder.queueInputBuffer(decode_input_index, 0, sample_size, sample_time, 0);
sample_time_queue.offer(sample_time);
extractor.advance();
}
}
else
{
DumpLog(TAG, "Decoder dequeueInputBuffer timed out! Try again later");
}
}
....
if (is_decode_output_done == false)
{
int decode_output_index = decoder.dequeueOutputBuffer(decode_buffer_info, TIMEOUT_USEC);
switch (decode_output_index)
{
case MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED:
{
....
break;
}
case MediaCodec.INFO_OUTPUT_FORMAT_CHANGED:
{
....
break;
}
case MediaCodec.INFO_TRY_AGAIN_LATER:
{
DumpLog(TAG, "Decoder dequeueOutputBuffer timed out! Try again later");
break;
}
default:
{
ByteBuffer decode_output_buffer = decode_output_buffers[decode_output_index];
long ptime_us = decode_buffer_info.presentationTimeUs;
boolean is_decode_EOS = ((decode_buffer_info.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0);
if (is_decode_EOS)
{
// Decoder gives an EOS output.
is_decode_output_done = true;
....
}
else
{
// The frame time may not be consistent for some videos.
// As a workaround, we use a frame time queue to guard this.
long sample_time = sample_time_queue.poll();
if (sample_time == ptime_us)
{
// Very good, the decoder input/output time is consistent.
}
else
{
// If the decoder input/output frame count is consistent, we can trust the sample time.
ptime_us = sample_time;
}
// process this frame
....
}
decoder.releaseOutputBuffer(decode_output_index, false);
}
}
}
In some cases, the queue can "correct" the PTS if the decoder gives error value (e.g. a lot of 0s). However, there are still some issues about the frame count of decoder input/output.
On an HTC One 801e device, I use the codec OMX.qcom.video.decoder.avc to decode the video (with MIME types video/avc). The sample time and PTS is matched well for the frames, except the last one.
For example, if the extractor feeds 100 frames and then EOS to the decoder, the first 99 decoded frames has the exactly same time values, but the last frame is missing and I get output EOS from the decoder. I test different videos encoded by the built-in camera, by ffmpeg muxer, or by a video processing AP on Windows. All of them have the last one frame disappeared.
On some pads with OMX.MTK.VIDEO.DECODER.AVC codec, things becomes more confused. Some videos has good PTS from the decoder and the input/output frame count is correct (i.e. the queue is empty when the decoding is done.). Some videos has consistent input/output frame count with bad PTS in decoder output (and I can still correct them by the queue). For some videos, a lot of frames are missing during the decoding. For example, the extractor get 210 frames in a 7 second video, but the decoder only output the last 180 frames. It is impossible to recover the PTS using the same workaround.
Is there any way to expect the input/output frame count for a MediaCodec decoder? Or more accurately, to know which frame(s) are dropped by the decoder while the extractor gives it video samples with correct sample time?

Same basic story as in the other question. Pre-4.3, there were no tests confirming that every frame fed to an encoder or decoder came out the other side. I recall that some devices would reliably drop the last frame in certain tests until the codecs were fixed in 4.3.
I didn't search for a workaround at the time, so I don't know if one exists. Delaying before sending EOS might help if it's causing something to shut down early.
I don't believe I ever saw a device drop large numbers of frames. This seems like an unusual case, as it would have been noticeable in any apps that exercised MediaCodec in similar ways even without careful testing.

How to get MJPG stream video from android IPWebcam using opencv

I am using the IP Webcam program on android and receiving it on my PC by WiFi. What I want is to use opencv in Visual Studio, C++, to get that video stream, there is an option to get MJPG stream by the following URL: http://MyIP:port/videofeed
How to get it using opencv?

Old question, but I hope this can help someone (same as my answer here)
OpenCV expects a filename extension for its VideoCapture argument,
even though one isn't always necessary (like in your case).
You can "trick" it by passing in a dummy parameter which ends in the
mjpg extension:
So perhaps try:
VideoCapture vc;
ipCam.open("http://MyIP:port/videofeed/?dummy=param.mjpg")

Install IP Camera Adapter and configure it to capture the videostream. Then install ManyCam and you'll see "MPEG Camera" in the camera section.(you'll see the same instructions if you go to the link on how to setup IPWebCam for skype)
Now you can access your MJPG stream just like a webcam through openCV. I tried this with OpenCV 2.2 + QT and works well.
Think this helps.

I did a dirty patch to make openCV working with android ipWebcam:
In the file OpenCV-2.3.1/modules/highgui/src/cap_ffmpeg_impl.hpp
In the function bool CvCapture_FFMPEG::open( const char* _filename )
replace:
int err = av_open_input_file(&ic, _filename, NULL, 0, NULL);
by
AVInputFormat* iformat = av_find_input_format("mjpeg");
int err = av_open_input_file(&ic, _filename, iformat, 0, NULL);
ic->iformat = iformat;
and comment:
err = av_seek_frame(ic, video_stream, 10, 0);
if (err < 0)
{
filename=(char*)malloc(strlen(_filename)+1);
strcpy(filename, _filename);
// reopen videofile to 'seek' back to first frame
reopen();
}
else
{
// seek seems to work, so we don't need the filename,
// but we still need to seek back to filestart
filename=NULL;
int64_t ts = video_st->first_dts;
int flags = AVSEEK_FLAG_FRAME | AVSEEK_FLAG_BACKWARD;
av_seek_frame(ic, video_stream, ts, flags);
}
That should work. Hope it helps.

This is the solution (im using IP Webcam on android):
CvCapture* capture = 0;
capture = cvCaptureFromFile("http://IP:Port/videofeed?dummy=param.mjpg");
I am not able to comment, so im posting new post. In original answer is an error - used / before dummy. THX for solution.

Working example for me
// OpenCVTest.cpp : Defines the entry point for the console application.
//
#include "stdafx.h"
#include "opencv2/highgui/highgui.hpp"
/**
* #function main
*/
int main( int argc, const char** argv )
{
CvCapture* capture;
IplImage* frame = 0;
while (true)
{
//Read the video stream
capture = cvCaptureFromFile("http://192.168.1.129:8080/webcam.mjpeg");
frame = cvQueryFrame( capture );
// create a window to display detected faces
cvNamedWindow("Sample Program", CV_WINDOW_AUTOSIZE);
// display face detections
cvShowImage("Sample Program", frame);
int c = cvWaitKey(10);
if( (char)c == 27 ) { exit(0); }
}
// clean up and release resources
cvReleaseImage(&frame);
return 0;
}
Broadcast mjpeg from a webcam with vlc, how described at http://tumblr.martinml.com/post/2108887785/how-to-broadcast-a-mjpeg-stream-from-your-webcam-with