Objective C Generating (empty) WAV file - objective-c

I'm trying to save some audio data to a WAV file -- I have audio data that normally I've been using in RemoteIO but I'm now trying to implement a function to save the data. I know the audio data is valid, so that's not a concern -- if I can just get an empty WAV file set up of the correct length, I can fill it with data later.
Right now, the code creates the file and it looks to be the right length in bytes, but apparently it's not formatted correctly, because OSX, QuickTime, iTunes, etc can't recognize it (they see the file, can't determine a length, or play it)
NSURL * tvarFilename = [savePanel URL];
NSLog(#"doSaveAs filename = %#",tvarFilename);
//try to create an audio file there
AudioFileID mRecordFile;
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = 44100.00;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 2;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerPacket = 4;
audioFormat.mBytesPerFrame = 4;
OSStatus status = AudioFileCreateWithURL((CFURLRef)tvarFilename, kAudioFileWAVEType, &audioFormat, kAudioFileFlags_EraseFile, &mRecordFile);
int beatsToRecord = 4; //temporary
int bpm = 120;
double intervalInSamples = (double) 60 / bpm;
intervalInSamples *= (double)44100;
int inNumberFrames = (intervalInSamples * beatsToRecord);
UInt32 frameBuffer[inNumberFrames];
int sampleTime = 0;
UInt32 thisSubBuffer[inNumberFrames];
for (int i = 0; i < inNumberFrames; i++) { frameBuffer[i] = 0; }
UInt32 bytesToWrite = inNumberFrames * sizeof(UInt32);
status = AudioFileWriteBytes(mRecordFile, false, 0, &bytesToWrite, &frameBuffer);

A WAV file is really simple: it only consists of a (usually 44-byte long) header section, then the raw PCM data. I've written a library which needs to record WAV files, and I'm pretty sure you'll understand how I accomplish it. For clarifying:
/**
* CD quality: 44100 Hz sample rate, 16 bit per sample, 2 channels (stereo):
**/
struct sprec_wav_header *hdr = sprec_wav_header_from_params(44100, 16, 2);
int filesize = (obtain the filesize somehow here);
/**
* -8 bytes for the first part of the header, see the WAV specification
**/
hdr->filesize = filesize - 8;
int filedesc = open("/tmp/dummy.wav", O_WRONLY | O_CREAT, 0644);
if (sprec_wav_header_write(filedesc, hdr))
{
printf("Error writing WAV header!\n");
}
close(filedesc);
free(hdr);
And the library I've written: https://github.com/H2CO3/libsprec/
Hope this helps.

The problem in code in initial question was in missing AudioFileClose(mRecordFile); line in the very end.
For those who still search for working sample without using 3rd party libraries, here is slightly modified code snippet:
- (void)createSilentWAVFileAtURL:(NSURL *)fileURL {
AudioFileID mRecordFile;
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = 44100.00;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 2;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerPacket = 4;
audioFormat.mBytesPerFrame = 4;
OSStatus status = AudioFileCreateWithURL((__bridge CFURLRef)fileURL, kAudioFileWAVEType, &audioFormat, kAudioFileFlags_EraseFile, &mRecordFile);
double intervalInSamples = 0.5;
intervalInSamples *= audioFormat.mSampleRate * audioFormat.mChannelsPerFrame;
int beatsToRecord = 4; //seconds of silence
int inNumberFrames = (intervalInSamples * beatsToRecord);
UInt32 frameBuffer[inNumberFrames];
for (int i = 0; i < inNumberFrames; i++) { frameBuffer[i] = 0; }
UInt32 bytesToWrite = inNumberFrames * sizeof(uint32_t);
status = AudioFileWriteBytes(mRecordFile, false, 0, &bytesToWrite, &frameBuffer);
status = AudioFileClose(mRecordFile);
NSAssert(status == noErr, #"");
}
P.S.: To decrease final file size reduce mChannelsPerFrame from 2 to 1 and mSampleRate (e.g. to 11000)

Related

Vulkan: Loading floating point cubemap textures distorted

I am using vulkan-tutorial codes and i made modify for cubemap.
when i use VK_FORMAT_R8G8B8A8_UNORM is working with this code:
unsigned char* pixelsArray[6];
for (int i = 0; i < 6; ++i)
{
pixelsArray[i] = stbi_load(imageFileArray[i].c_str(), &texWidth, &texHeight, &texChannels, STBI_rgb_alpha);
}
VkDeviceSize allSize = texWidth * texHeight * 4 * 6;
VkDeviceSize size = texWidth * texHeight * 4 ;
VkBufferCreateInfo bufferInfo{};
...
bufferInfo.size = allSize ;
vkMapMemory(device, stagingBufferMemory, 0, AllSize, 0, &data);
for(int i = 0; i < 6; ++i)
{
memcpy( (char*) data + (size*i) , pixelsArray[i], static_cast<size_t>(size));
}
vkUnmapMemory(device, stagingBufferMemory);
VkImageCreateInfo imageInfo{};
...
imageInfo.arrayLayers = 6;
imageInfo.format = VK_FORMAT_R8G8B8A8_UNORM;
imageInfo.flags = VK_IMAGE_CREATE_CUBE_COMPATIBLE_BIT;
VkImageViewCreateInfo viewInfo{};
...
viewInfo.viewType = VK_IMAGE_VIEW_TYPE_CUBE;
viewInfo.format = VK_FORMAT_R8G8B8A8_UNORM;
viewInfo.subresourceRange.layerCount = 6;
but when i try VK_FORMAT_R16G16B16A16_SFLOAT is giving distorted display and no validation error with this code:
float* pixelsArray[6];
for (int i = 0; i < 6; ++i)
{
pixelsArray[i] = stbi_loadf(imageFileArray[i].c_str(), &texWidth, &texHeight, &texChannels, STBI_rgb_alpha);
}
VkDeviceSize allSize = texWidth * texHeight * 4 * 6 * 2;// I added *2
VkDeviceSize size = texWidth * texHeight * 4 * 2;// I added *2
VkBufferCreateInfo bufferInfo{};
...
bufferInfo.size = allSize ;
vkMapMemory(device, stagingBufferMemory, 0, AllSize, 0, &data);
for(int i = 0; i < 6; ++i)
{
memcpy( (char*) data + (size*i) , pixelsArray[i], static_cast<size_t>(size));
}
vkUnmapMemory(device, stagingBufferMemory);
VkImageCreateInfo imageInfo{};
...
imageInfo.arrayLayers = 6;
imageInfo.format = VK_FORMAT_R16G16B16A16_SFLOAT;
imageInfo.flags = VK_IMAGE_CREATE_CUBE_COMPATIBLE_BIT;
VkImageViewCreateInfo viewInfo{};
...
viewInfo.viewType = VK_IMAGE_VIEW_TYPE_CUBE;
viewInfo.format = VK_FORMAT_R16G16B16A16_SFLOAT;
viewInfo.subresourceRange.layerCount = 6;
when VK_FORMAT_R8G8B8A8_UNORM :
when VK_FORMAT_R16G16B16A16_SFLOAT :
i fixed the problem. problem was that i want to use half float but i was sending float to memcpy function.i searched how can i use half float and i found a solution without using extra library.
what i did add helper functions :
typedef unsigned int uint;
typedef unsigned short ushort;
uint as_uint(const float x)
{
return *(uint*)&x;
}
ushort float_to_half(const float x)
{
// IEEE-754 16-bit floating-point format (without infinity): 1-5-10, exp-15, +-131008.0, +-6.1035156E-5, +-5.9604645E-8, 3.311 digits
const uint b = as_uint(x)+0x00001000; // round-to-nearest-even: add last bit after truncated mantissa
const uint e = (b&0x7F800000)>>23; // exponent
const uint m = b&0x007FFFFF; // mantissa; in line below: 0x007FF000 = 0x00800000-0x00001000 = decimal indicator flag - initial rounding
return (b&0x80000000)>>16 | (e>112)*((((e-112)<<10)&0x7C00)|m>>13) | ((e<113)&(e>101))*((((0x007FF000+m)>>(125-e))+1)>>1) | (e>143)*0x7FFF; // sign : normalized : denormalized : saturate
}
and fix problem with this helper functions :
VkDeviceSize size_2 = texWidth * texHeight * 4;// different from the above variables in question : allSize or size
//create half float for cubemap
void* half_pixelsArray[6];
half_pixelsArray[0] = new ushort[size_2];
half_pixelsArray[1] = new ushort[size_2];
half_pixelsArray[2] = new ushort[size_2];
half_pixelsArray[3] = new ushort[size_2];
half_pixelsArray[4] = new ushort[size_2];
half_pixelsArray[5] = new ushort[size_2];
//copy from float to half float
for (int i = 0; i < 6; ++i)
{
for (int j = 0; j < size_2; ++j)
{
((ushort*)half_pixelsArray[i])[j] = float_to_half( pixelsArray[i][j] );
}
}
// and change float to half flaot in memcpy
memcpy( (char*) data + (layerSize*i) , half_pixelsArray[i], static_cast<size_t>(layerSize));

heap corruption when using pin_ptr to copy from native code to managed code

I am trying to copy unsigned short from native code to managed code, but I get a heap corruption when calling memcpy.
INPUT: unsigned short* input
OUTPUT: array<unsigned short> output
I have the following code and if I set testDataSize is 100 then I don't see corruption.
Could someone please shed some light ?
Thanks,
typedef unsigned short uns16;
// DLL Entry Point
void main()
{
int testDataSize = 600;
int frSize = testDataSize / 2;
for (int j = 0; j < 1; j++)
{
uns16* input;
array<uns16>^ output1;
array<uns16>^ output2;
input = new uns16(frSize);
output1 = gcnew array <uns16>(frSize);
output2 = gcnew array <uns16>(frSize);
// initialize
for (int i = 0; i < frSize; i++)
{
input[i] = i;
}
//test 1
Stopwatch^ sw1 = Stopwatch::StartNew();
//-------------------------------------------------------------------
array<short>^ frameDataSigned = gcnew array<short>(frSize);
Marshal::Copy(IntPtr((void*)(input)), frameDataSigned, 0, frameDataSigned->Length);
System::Buffer::BlockCopy(frameDataSigned, 0, output1, 0, (Int32)(frSize) * 2);
//-------------------------------------------------------------------
auto res1 = sw1->ElapsedTicks;
//test 2
Stopwatch^ sw2 = Stopwatch::StartNew();
//-------------------------------------------------------------------
cli::pin_ptr<uns16> pinnedManagedData = &output2[0];
memcpy(pinnedManagedData, (void*)(input), frSize * sizeof(uns16));
//-------------------------------------------------------------------
auto res2 = sw2->ElapsedTicks;
....
int frSize = 300;
input = new uns16(frSize);
This doesn't allocate an array. It allocates a single uint16_t, and sets its value to 300. You need to use square brackets to allocate an array.
input = new uns16[frSize];

220 hertz sine doesn't sound correct

I have the following Objective-C code that generates a 220hz tone, however, it doesn't sound like other 220hz tones, for example Wikipedia has a 220 tone here: https://en.wikipedia.org/wiki/File:220_Hz_sine_wave.ogg that sounds very low, mine sounds really high pitched in comparison...any ideas?
- (void)render:(AVAudioPCMBuffer *)bufferInput frequency:(float)frequency amplitude:(float)amplitude
{
const int channels = 2;
int inNumberFrames = bufferInput.frameLength;
float *const *floatChannelData = bufferInput.floatChannelData;
for (int i = 0; i < inNumberFrames ; i ++) {
float theta = frequency * i * 2.0 * M_PI / 44100;
float value = sinf(theta);
for (int channelNumber = 0; channelNumber < channels ; channelNumber++) {
float * const channelBuffer = floatChannelData[channelNumber];
channelBuffer[i] = value * amplitude;
}
}
}

How to write NALs produced by x264_encoder_encode() using ffmpeg av_interleaved_write_frame()

I have been trying to produce a "flv" video file in the following sequence:
av_register_all();
// Open video file
if (avformat_open_input(&pFormatCtx, "6.mp4", NULL, NULL) != 0)
return -1; // Couldn't open file
// Retrieve stream information
if (avformat_find_stream_info(pFormatCtx, NULL) < 0)
return -1; // Couldn't find stream information
// Dump information about file onto standard error
av_dump_format(pFormatCtx, 0, "input_file.mp4", 0);
// Find the first video stream
videoStream = -1;
for (i = 0; i < pFormatCtx->nb_streams; i++)
if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_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;
// 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_open2(pCodecCtx, pCodec, NULL) < 0)
return -1; // Could not open codec
// Allocate video frame
pFrame = avcodec_alloc_frame();
// Allocate video frame
pFrame = avcodec_alloc_frame();
// Allocate an AVFrame structure
pFrameYUV420 = avcodec_alloc_frame();
if (pFrameYUV420 == NULL)
return -1;
// Determine required buffer size and allocate buffer
numBytes = avpicture_get_size(pCodecCtx->pix_fmt, pCodecCtx->width, pCodecCtx->height);
buffer = (uint8_t *) av_malloc(numBytes * sizeof(uint8_t));
// Assign appropriate parts of buffer to image planes in pFrameYUV420
// Note that pFrameYUV420 is an AVFrame, but AVFrame is a superset of AVPicture
avpicture_fill((AVPicture *) pFrameRGB, buffer, pCodecCtx->pix_fmt, pCodecCtx->width, pCodecCtx->height);
// Setup scaler
img_convert_ctx = sws_getContext(pCodecCtx->width, pCodecCtx->height, pCodecCtx->pix_fmt, pCodecCtx->width, pCodecCtx->height, pCodecCtx->pix_fmt, SWS_BILINEAR, 0, 0, 0);
if (img_convert_ctx == NULL) {
fprintf(stderr, "Cannot initialize the conversion context!\n");
exit(1);
}
// Setup encoder/muxing now
filename = "output_file.flv";
fmt = av_guess_format("flv", filename, NULL);
if (fmt == NULL) {
printf("Could not guess format.\n");
return -1;
}
/* allocate the output media context */
oc = avformat_alloc_context();
if (oc == NULL) {
printf("could not allocate context.\n");
return -1;
}
oc->oformat = fmt;
snprintf(oc->filename, sizeof(oc->filename), "%s", filename);
video_st = NULL;
if (fmt->video_codec != AV_CODEC_ID_NONE) {
video_st = add_stream(oc, &video_codec, fmt->video_codec);
}
// Let's see some information about our format
av_dump_format(oc, 0, filename, 1);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open '%s': %s\n", filename, av_err2str(ret));
return 1;
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n", av_err2str(ret));
return 1;
}
// Setup x264 params
x264_param_t param;
x264_param_default_preset(&param, "veryfast", "zerolatency");
param.i_threads = 1;
param.i_width = video_st->codec->width;
param.i_height = video_st->codec->height;
param.i_fps_num = STREAM_FRAME_RATE; // 30 fps, same as video
param.i_fps_den = 1;
// Intra refres:
param.i_keyint_max = STREAM_FRAME_RATE;
param.b_intra_refresh = 1;
// Rate control:
param.rc.i_rc_method = X264_RC_CRF;
param.rc.f_rf_constant = 25;
param.rc.f_rf_constant_max = 35;
// For streaming:
param.b_repeat_headers = 1;
param.b_annexb = 1;
x264_param_apply_profile(&param, "baseline");
x264_t* encoder = x264_encoder_open(&param);
x264_picture_t pic_in, pic_out;
x264_picture_alloc(&pic_in, X264_CSP_I420, video_st->codec->width, video_st->codec->height);
x264_nal_t* nals;
int i_nals;
// The loop:
// 1. Read frames
// 2. Decode the frame
// 3. Attempt to re-encode using x264
// 4. Write the x264 encoded frame using av_interleaved_write_frame
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_video2(pCodecCtx, pFrame, &frameFinished, &packet);
// Did we get a video frame?
if (frameFinished) {
sws_scale(img_convert_ctx, pFrame->data, pFrame->linesize, 0, pCodecCtx->height, pic_in.img.plane, pic_in.img.i_stride);
int frame_size = x264_encoder_encode(encoder, &nals, &i_nals, &pic_in, &pic_out);
if (frame_size >= 0) {
if (i_nals < 0)
printf("invalid frame size: %d\n", i_nals);
// write out NALs
for (i = 0; i < i_nals; i++) {
// initalize a packet
AVPacket p;
av_init_packet(&p);
p.data = nals[i].p_payload;
p.size = nals[i].i_payload;
p.stream_index = video_st->index;
p.flags = AV_PKT_FLAG_KEY;
p.pts = AV_NOPTS_VALUE;
p.dts = AV_NOPTS_VALUE;
ret = av_interleaved_write_frame(oc, &p);
}
}
printf("encoded frame #%d\n", frame_count);
frame_count++;
}
}
// Free the packet that was allocated by av_read_frame
av_free_packet(&packet);
}
// Now we free up resources used/close codecs, and finally close our program.
Here is the implementation for the add_stream() function:
/* Add an output stream. */
static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec, enum AVCodecID codec_id) {
AVCodecContext *c;
AVStream *st;
int r;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}
st = avformat_new_stream(oc, *codec);
if (!st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
st->id = oc->nb_streams - 1;
c = st->codec;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
st->id = 1;
c->sample_fmt = AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;
break;
case AVMEDIA_TYPE_VIDEO:
avcodec_get_context_defaults3(c, *codec);
c->codec_id = codec_id;
c->bit_rate = 500*1000;
//c->rc_min_rate = 500*1000;
//c->rc_max_rate = 500*1000;
//c->rc_buffer_size = 500*1000;
/* Resolution must be a multiple of two. */
c->width = 1280;
c->height = 720;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
c->time_base.den = STREAM_FRAME_RATE;
c->time_base.num = 1;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
default:
break;
}
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
return st;
}
After the encoding is complete, I check the output file output_file.flv. I notice it's size is very large: 101MB and it does not play. If I use ffmpeg to decode/encode the input file, then I get an output file about 83MB in size (which is about the same size as the original .mp4 file used as input). Also, the 83MB output from just using ffmpeg C api, as opposed to using x264 for the encoding step, plays just fine. Does anyone know where I am going wrong? I have tried researching this for a few days now but with no luck :(. I feel that I am close to making it work, however, I just cannot figure out what I am doing wrong. Thank you!
To produce the correct AVPacket, you should write all nals into the same packet, as it is done in http://ffmpeg.org/doxygen/trunk/libx264_8c_source.html (see encode_nals and X264_frame functions)

Error in AudioFileWritePackets

i'm just trying to capture the mic and save data to wav file ! and here is the code:
AudioFileID FileID=remoteIOplayer->mixAudioFile;
if (inBusNumber == 16){
AudioUnitRender(audioUnit, ioActionFlags, inTimeStamp, 1, inNumberFrames, ioData);
OSStatus result = AudioFileWritePackets(FileID, false, (inNumberFrames * 4), NULL, mixpacketNum, &inNumberFrames, ioData->mBuffers[i].mData);
if (result != noErr){
NSLog(#"Error Writing");
mixpacketNum += inNumberFrames;
}
}
and i created the file correctly with the following format that also used to capture the mic:
audioFormat.mSampleRate = 44100.00;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerPacket = 2;
audioFormat.mBytesPerFrame = 2;
the error is result OSStatus -38 fnopnerr (file not open)
I'm guessing that your call to open the audio file either failed or wasn't done before rendering started. What does your call to AudioFileCreateWithURL look like?