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-rw-r--r--rotord/1wave.mp3bin0 -> 8575 bytes
-rw-r--r--rotord/1wave.wavbin0 -> 88878 bytes
-rw-r--r--rotord/rotor.cpp274
-rwxr-xr-xrotord/rotor.h8
-rw-r--r--rotord/silence.mp3bin481070 -> 8575 bytes
-rw-r--r--rotord/silence.wavbin2646044 -> 88244 bytes
6 files changed, 42 insertions, 240 deletions
diff --git a/rotord/1wave.mp3 b/rotord/1wave.mp3
new file mode 100644
index 0000000..040b65e
--- /dev/null
+++ b/rotord/1wave.mp3
Binary files differ
diff --git a/rotord/1wave.wav b/rotord/1wave.wav
new file mode 100644
index 0000000..0ae9d9b
--- /dev/null
+++ b/rotord/1wave.wav
Binary files differ
diff --git a/rotord/rotor.cpp b/rotord/rotor.cpp
index 53ba33b..f7c017d 100644
--- a/rotord/rotor.cpp
+++ b/rotord/rotor.cpp
@@ -10,14 +10,6 @@ bool fequal(const float u,const float v){
using namespace Rotor;
-/*
- string soname="qm-vamp-plugins";
- string id="qm-tempotracker";
- string myname="";
- string output="";
- int outputNo=0;
- */
-
void Render_context::runTask() {
while (!isCancelled()) {
int cmd=0;
@@ -29,13 +21,10 @@ void Render_context::runTask() {
mutex.unlock();
if (cmd==ANALYSE_AUDIO) {
state=ANALYSING_AUDIO;
- //audio_analyser.process(audio_filename);
- //vampHost::runPlugin("","qm-vamp-plugins","qm-tempotracker", "",0, audio_filename, cerr,true);
vector<base_audio_processor*> proc;
proc.push_back(audio_thumb);
if (load_audio(audio_filename,proc)) {
state=AUDIO_READY;
- //set the response
}
else state=IDLE;
}
@@ -72,8 +61,8 @@ bool Signal_output::render(const float duration, const float framerate,string &x
}
Command_response Render_context::session_command(const std::vector<std::string>& command){
- //method,id,command1,{command2,}{body}
- //here we allow the controlling server to communicate with running tasks
+ //method,id,command1,{command2,}{body}
+ //here we allow the controlling server to communicate with running tasks
Command_response response;
response.status=HTTPResponse::HTTP_BAD_REQUEST;
if (command[2]=="audio") {
@@ -82,9 +71,6 @@ Command_response Render_context::session_command(const std::vector<std::string>&
if (state==IDLE) {
//check file exists
Poco::File f=Poco::File(command[3]);
- //std::auto_ptr<std::istream> pStr(URIStreamOpener::defaultOpener().open(command[3]));
-
-
if (f.exists()) {
//pass to worker thread ??if engine is ready?? ??what if engine has finished but results aren't read??
audio_filename=command[3]; //for now, store session variables in memory
@@ -269,193 +255,8 @@ Command_response Render_context::session_command(const std::vector<std::string>&
//great to use c++11 features
bool Render_context::load_audio(const string &filename,vector<base_audio_processor*> processors){
- //load audio data from file
- //what's the best way to use this? the model is background processing, and we want to update a progress bar
- //could pass a function pointer to call when each chunk of data becomes available?
- //should the data processing be the responsibility of an object or a function?
- //in the case of the audio thumbnail, there will be just one place where its kept
- //in the case of audio analysis, the daemon will pass each audio analysis object each chunk of data as it gets it
- //there could even be an array of audio analysis functions to perform simultaneously?
- //how about a vector of objects that subclass the base audio processor class?
-
- //1st get parameters and initialise the processors
- //then begin data loop locking progress variable after each frame
-
- //
- //
- //the example in ffmpeg works, but it isn't one that identifies a codec- it is hard coded to look for a codec for AV_CODEC_ID_MP2
- //it also doesn't load through libavformat - which opens containers- it just loads a naked .mp2 stream
- //
- //
-
- /*
- av_register_all();
-
- std::shared_ptr<AVFormatContext> avFormat(avformat_alloc_context(), &avformat_free_context);
-
- auto avFormatPtr = avFormat.get();
- if (avformat_open_input(&avFormatPtr,filename.c_str(),nullptr, nullptr) != 0) {
- cerr <<"Rotor: Error while calling avformat_open_input (probably invalid file format)" << endl;
- return false;
- }
-
- if (avformat_find_stream_info(avFormat.get(), nullptr) < 0) {
- cerr << "Rotor: Error while calling avformat_find_stream_info" << endl;
- return false;
- }
-
- av_dump_format(avFormat.get(), 0, 0, false); //avformat.h line 1256
-
-
-
- AVStream* stream = nullptr;
- for (unsigned int i = 0; i < avFormat->nb_streams; ++i) {
- if (avFormat->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
- // we've found a stream!
- stream = avFormat->streams[i];
- break;
- }
- }
- if (!stream) {
- cerr <<"Rotor: Didn't find any audio stream in the file"<< endl;
- return false;
- }
-
- // getting the required codec structure
- const auto codec = avcodec_find_decoder(stream->codec->codec_id); //returns AVCodec*
- if (codec == nullptr) {
- cerr <<"Rotor: Audio codec not available"<< endl;
- return false;
- }
-
- //AVCodecContext?? avFormat->streams[i]->codec
-
- // allocating a structure
- std::shared_ptr<AVCodecContext> audioCodec(avcodec_alloc_context3(codec), [](AVCodecContext* c) { avcodec_close(c); av_free(c); });
-
- // extradata???
- // we need to make a copy of videoStream->codec->extradata and give it to the context
- // make sure that this vector exists as long as the avVideoCodec exists
- std::vector<uint8_t> codecContextExtraData;
- codecContextExtraData.assign(stream->codec->extradata, stream->codec->extradata + stream->codec->extradata_size);
-
- audioCodec->extradata = reinterpret_cast<uint8_t*>(codecContextExtraData.data());
- audioCodec->extradata_size = codecContextExtraData.size();
-
- // initializing the structure by opening the codec
- if (avcodec_open2(audioCodec.get(), codec, nullptr) < 0) {
- cerr <<"Rotor: Could not open codec"<< endl;
- return false;
- }
-
-
- //avcodec.h line 1026
-
-
- Packet packet;
-
-// decoding code here
- //cerr << "audio codec context - sample rate: "<< audioCodec->sample_rate <<", channels: "<<audioCodec->channels<<", sample format: "<<audioCodec->sample_fmt<<endl;
-
- // suspect why was this in twice 210313
- //do {
- // if (packet.packet.stream_index != stream->index)
- /// continue; // the packet is not about the stream we want, let's jump again the start of the loop
- //} while(0);
- //
-
- // allocating an AVFrame
- std::shared_ptr<AVFrame> avFrame(avcodec_alloc_frame(), &av_free);
-
- // the current packet of data
- //Packet packet;
- // data in the packet of data already processed
- size_t offsetInData = 0;
-
-
- bool foundPacket=false;
-
- // the decoding loop, running until EOF
- while (true) {
- // reading a packet using libavformat
- if (offsetInData >= packet.packet.size) {
- do {
- packet.reset(avFormat.get());
- if (packet.packet.stream_index != stream->index)
- continue;
- } while(0);
- if (!foundPacket){
- cerr << "audio codec context - sample rate: "<< audioCodec->sample_rate <<", channels: "<<audioCodec->channels<<", sample format: "<<audioCodec->sample_fmt<<endl;
- foundPacket=true;
- }
-
- }
-
-
-
- // preparing the packet that we will send to libavcodec
- AVPacket packetToSend;
- packetToSend.data = packet.packet.data + offsetInData;
- packetToSend.size = packet.packet.size - offsetInData;
-
- // sending data to libavcodec
- int isFrameAvailable = 0;
- //const auto processedLength = avcodec_decode_video2(avVideoCodec.get(), avFrame.get(), &isFrameAvailable, &packetToSend);
-
- const auto processedLength = avcodec_decode_audio4(audioCodec.get(), avFrame.get(), &isFrameAvailable, &packetToSend);
- //utils.c line 2018
-
- //unsigned int bitsPerSample = av_get_bytes_per_sample(stream.codecContext->sample_fmt) * CHAR_BIT; // 16 or 32
- //unsigned int numberOfChannels = stream.codecContext->channels; // 1 for mono, 2 for stereo, or more
- //unsigned int numberOfSamplesContainedInTheFrame = stream.frame->nb_samples * stream.codecContext->channels;
- //unsigned int numberOfSamplesToPlayPerSecond = stream.codecContext->sample_rate; // usually 44100 or 48000
- //const void* data = stream.frame->data[0];
-
-
- if (processedLength < 0) {
- //av_free_packet(&packet); shouldn't have to because of type safe wrapper
- cerr <<"Error "<<processedLength<< " while processing the data"<< endl;
- return false;
- }
- offsetInData += processedLength;
-
-
- // processing the image if available
- if (isFrameAvailable) {
- // display image on the screen
-
- // sleeping until next frame
- const auto msToWait = avVideoContext->ticks_per_frame * 1000 * avVideoContext->time_base.num / avVideoContext->time_base.den;
- std::this_thread::sleep(std::chrono::milliseconds(msToWait));
- }
- */
-
-
- //http://www.gamedev.net/topic/624876-how-to-read-an-audio-file-with-ffmpeg-in-c/
- // Initialize FFmpeg
-
- //this seems to only read a certain percentage of the sound file? - different for wav and mp3
- //peculiar, because the data comes in as frames and we just read them until all are done
- //is it possible that thhe problem lies with the drawing process only?
- //wav sees 6/10 waves and mp3 sees 3.5/10
- //could this be an aliasing effect i.e. from reading the sample format wrong?
- //I think probably not?
-
- //every format combination seems to miss a different number of samples
- //I don't understand how different formats would make the samples seem to truncate?
- //unless its an aliasing type thing?
-
- //wave.mp3 reads 16.409% of 160913
- //wave.wav reads 29.706% of 882044
-
-
-
- av_register_all();
- //??
- //avcodec_init();
- avcodec_register_all();
- //??
+ av_register_all();
AVFrame* frame = avcodec_alloc_frame();
if (!frame)
@@ -464,8 +265,6 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
return false;
}
- // you can change the file name "01 Push Me to the Floor.wav" to whatever the file is you're reading, like "myFile.ogg" or
- // "someFile.webm" and this should still work
AVFormatContext* formatContext = NULL;
if (avformat_open_input(&formatContext, filename.c_str(), NULL, NULL) != 0)
{
@@ -484,7 +283,6 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
}
AVStream* audioStream = NULL;
- // Find the audio stream (some container files can have multiple streams in them)
for (unsigned int i = 0; i < formatContext->nb_streams; ++i)
{
if (formatContext->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
@@ -519,12 +317,6 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
std::cout << "Couldn't open the context with the decoder" << std::endl;
return false;
}
-
- //
- // why is the file truncated???
- //if(codecContext->codec->capabilities & CODEC_CAP_TRUNCATED) codecContext->codec->capabilities|=CODEC_FLAG_TRUNCATED;
- //
- //
av_dump_format(formatContext, 0, 0, false); //avformat.h line 1256
int samples = ((formatContext->duration + 5000)*codecContext->sample_rate)/AV_TIME_BASE;
@@ -532,9 +324,6 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
std::cout << "This stream has " << codecContext->channels << " channels, a sample rate of " << codecContext->sample_rate << "Hz and "<<samples <<" samples" << std::endl;
std::cout << "The data is in format " <<codecContext->sample_fmt<< " (aka "<< av_get_sample_fmt_name(codecContext->sample_fmt) << ") "<<std::endl;
- //we can now tell the processors the format
- //we can work out the number of samples at this point
-
for (auto p: processors) {
p->init(codecContext->channels,16,samples);
}
@@ -545,16 +334,13 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
bool diag=true;
- // Read the packets in a loop
while (true)
- //while(sample_processed<samples)
{
int ret=av_read_frame(formatContext, &packet);
if (ret<0) {
cerr << "finished with code "<<ret <<(ret==AVERROR_EOF?" ,EOF":"")<<endl;
break;
}
- //av_read_frame(formatContext, &packet); //hangs once the packets have been read
if (packet.stream_index == audioStream->index)
{
// Try to decode the packet into a frame
@@ -580,7 +366,6 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
diag=false;
}
- //std::cout << "Got a frame: bytes " << bytes << ", "<<frame->nb_samples<<" samples"<<std::endl;
//now we can pass the data to the processor(s)
for (auto p: processors) {
sample_processed=p->process_frame(frame->data[0],frame->nb_samples);
@@ -592,7 +377,7 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
}
}
// You *must* call av_free_packet() after each call to av_read_frame() or else you'll leak memory
- av_free_packet(&packet); //crashes here!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! SIGSEV In _int_free (av=0xb4600010, p=0xb46025c8, have_lock=0) at malloc.c:4085 ()
+ av_free_packet(&packet);
}
// Some codecs will cause frames to be buffered up in the decoding process. If the CODEC_CAP_DELAY flag
@@ -614,9 +399,8 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
}
}
- cerr << "finished processed: "<<sample_processed << " samples of "<<samples<<" , "<<((double)sample_processed*100)/samples<<"%"<< std::endl;
+ cerr << "finished processing: "<<sample_processed << " samples of "<<samples<<", "<<((double)sample_processed*100)/samples<<"%"<< std::endl;
-
// Clean up!
av_free(frame);
avcodec_close(codecContext);
@@ -695,6 +479,9 @@ void audio_thumbnailer::init(int _channels,int _bits,int _samples) {
out_sample=0; //sample in whole track
offset=0x1<<(bits-1); //signed audio
scale=1.0/offset;
+ sample=0;
+ samples=0;
+ accum=0.0;
}
int audio_thumbnailer::process_frame(uint8_t *_data,int samples_in_frame){
//begin by processing remaining samples
@@ -704,36 +491,51 @@ int audio_thumbnailer::process_frame(uint8_t *_data,int samples_in_frame){
int stride=channels*bytes;
int in_sample=0;
while (in_sample<samples_in_frame&&column<width) {
- int sample=0;
- int samples=0;
- double accum=0;
+ //start a new column
+
while (sample<samples_per_column&&in_sample<samples_in_frame) {
- //accumulate squares for this column
+ //accumulate samples for this column until we run out of samples
for (int i=0;i<channels;i++) {
- int this_val=0;
+ unsigned int this_val=0;
for (int j=0;j<bytes;j++) {
this_val+=_data[(sample*stride)+(i*bytes)+j]<<(j*8);
}
//convert from integer data format - i.e s16p - to audio signal in -1..1 range
//don't know how many bytes we are dealing with necessarily?
//double val=((double)(this_val-offset))*scale;
- if (this_val>offset) this_val=-(this_val-(offset*2));
- double val=((double)this_val)*scale;
- accum+=abs(val); //(val*val);
+
+ //
+ //cerr << this_val << " ";
+ //
+
+
+ //if (this_val>offset) this_val=-(this_val-(offset*2));
+
+ //this_val -=offset;
+ double val=((double)((int16_t)this_val))*scale;
+ accum+=val*val;
samples++;
}
in_sample++;
sample++;
out_sample++;
}
- //get root-mean
- double mean=accum/samples; //pow(accum/samples,0.5);
- int colheight=height*mean*0.5;
- int hh=height>>1;
- for (int i=0;i<height;i++) {
- data[i*width+column]=abs(i-hh)<colheight?0xff:0x00;
+ if (sample==samples_per_column) { //finished a column
+ //get root-mean
+ double mean=pow(accum/samples,0.5);
+ if (column==0) {
+ cerr << "first column total: "<< accum << " in " << samples << " samples, average " << (accum/samples)<<endl;
+ }
+ int colheight=height*mean*0.5;
+ int hh=height>>1;
+ for (int i=0;i<height;i++) {
+ data[i*width+column]=abs(i-hh)<colheight?0xff:0x00;
+ }
+ column++;
+ sample=0;
+ samples=0;
+ accum=0.0;
}
- column++;
}
return out_sample;
}
diff --git a/rotord/rotor.h b/rotord/rotor.h
index b7ffc25..436de10 100755
--- a/rotord/rotor.h
+++ b/rotord/rotor.h
@@ -345,8 +345,8 @@ namespace Rotor {
//draw pixels based on rms value
public:
audio_thumbnailer(){
- height=16;
- width=64; //fit
+ height=48;
+ width=128; //fit
data=new uint8_t[height*width];
memset(data,0,height*width);
}
@@ -358,10 +358,10 @@ namespace Rotor {
string print();
uint8_t *data;
int height,width,samples_per_column;
- int column,out_sample;
+ int column,out_sample,sample,samples;
//for drawing graph
int offset;
- double scale;
+ double scale,accum;
};
class Render_context: public Poco::Task { //Poco task object
//manages a 'patchbay'
diff --git a/rotord/silence.mp3 b/rotord/silence.mp3
index 32fbbb4..888e669 100644
--- a/rotord/silence.mp3
+++ b/rotord/silence.mp3
Binary files differ
diff --git a/rotord/silence.wav b/rotord/silence.wav
index 3c98ffc..24d262a 100644
--- a/rotord/silence.wav
+++ b/rotord/silence.wav
Binary files differ
generated by cgit v1.2.3 (git 2.39.1) at 2025-12-07 01:49:24 +0000