在IOS上使用FFMPEG mux flv和发送rtmp
我想使用iphone摄像头和麦克风来捕获通过FFMPEG RTMP流传输的信息在IOS上使用FFMPEG mux flv和发送rtmp,ios,ffmpeg,mp3,h.264,rtmp,Ios,Ffmpeg,Mp3,H.264,Rtmp,我想使用iphone摄像头和麦克风来捕获通过FFMPEG RTMP流传输的信息 - (void)encoderToH264:(CMSampleBufferRef)sampleBuffer { CVPixelBufferRef imageBuffer = CMSampleBufferGetImageBuffer(sampleBuffer); if (CVPixelBufferLockBaseAddress(imageBuffer, 0) == kCVReturnSuccess)
- (void)encoderToH264:(CMSampleBufferRef)sampleBuffer
{
CVPixelBufferRef imageBuffer = CMSampleBufferGetImageBuffer(sampleBuffer);
if (CVPixelBufferLockBaseAddress(imageBuffer, 0) == kCVReturnSuccess)
{
UInt8 *bufferbasePtr = (UInt8 *)CVPixelBufferGetBaseAddress(imageBuffer);
UInt8 *bufferPtr = (UInt8 *)CVPixelBufferGetBaseAddressOfPlane(imageBuffer,0);
UInt8 *bufferPtr1 = (UInt8 *)CVPixelBufferGetBaseAddressOfPlane(imageBuffer,1);
size_t buffeSize = CVPixelBufferGetDataSize(imageBuffer);
size_t width = CVPixelBufferGetWidth(imageBuffer);
size_t height = CVPixelBufferGetHeight(imageBuffer);
size_t bytesPerRow = CVPixelBufferGetBytesPerRow(imageBuffer);
size_t bytesrow0 = CVPixelBufferGetBytesPerRowOfPlane(imageBuffer,0);
size_t bytesrow1 = CVPixelBufferGetBytesPerRowOfPlane(imageBuffer,1);
size_t bytesrow2 = CVPixelBufferGetBytesPerRowOfPlane(imageBuffer,2);
UInt8 *yuv420_data = (UInt8 *)malloc(width * height *3/ 2); // buffer to store YUV with layout YYYYYYYYUUVV
/* convert NV12 data to YUV420*/
UInt8 *pY = bufferPtr ;
UInt8 *pUV = bufferPtr1;
UInt8 *pU = yuv420_data + width*height;
UInt8 *pV = pU + width*height/4;
for(int i =0;i<height;i++)
{
memcpy(yuv420_data+i*width,pY+i*bytesrow0,width);
}
for(int j = 0;j<height/2;j++)
{
for(int i =0;i<width/2;i++)
{
*(pU++) = pUV[i<<1];
*(pV++) = pUV[(i<<1) + 1];
}
pUV+=bytesrow1;
}
//Read raw YUV data
picture_buf = yuv420_data;
pFrame->data[0] = picture_buf; // Y
pFrame->data[1] = picture_buf+ y_size; // U
pFrame->data[2] = picture_buf+ y_size*5/4; // V
int got_picture = 0;
// Encode
pFrame->width = 720;
pFrame->height = 1280;
pFrame->format = PIX_FMT_YUV420P;
AVCodecContext *c = video_st->codec;
int ret = avcodec_encode_video2(c, &pkt, pFrame, &got_picture);
if(ret < 0)
{
printf("Failed to encode! \n");
}
if (got_picture==1)
{
/* Compute current audio and video time. */
video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
pFrame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
if(pkt.size != 0)
{
printf("Succeed to encode frame: %5lld\tsize:%5d\n", pFrame->pts, pkt.size);
pkt.stream_index = video_st->index;
ret = av_write_frame(pFormatCtx, &pkt);
av_free_packet(&pkt);
}
}
free(yuv420_data);
}
CVPixelBufferUnlockBaseAddress(imageBuffer, 0);
}
-(void)encoderToMP3:(CMSampleBufferRef)sampleBuffer
{
CMSampleTimingInfo timing_info;
CMSampleBufferGetSampleTimingInfo(sampleBuffer, 0, &timing_info);
double pts=0;
double dts=0;
AVCodecContext *c;
int got_packet, ret;
c = audio_st->codec;
CMItemCount numSamples = CMSampleBufferGetNumSamples(sampleBuffer);
NSUInteger channelIndex = 0;
CMBlockBufferRef audioBlockBuffer = CMSampleBufferGetDataBuffer(sampleBuffer);
size_t audioBlockBufferOffset = (channelIndex * numSamples * sizeof(SInt16));
size_t lengthAtOffset = 0;
size_t totalLength = 0;
SInt16 *samples = NULL;
CMBlockBufferGetDataPointer(audioBlockBuffer, audioBlockBufferOffset, &lengthAtOffset, &totalLength, (char **)(&samples));
const AudioStreamBasicDescription *audioDescription = CMAudioFormatDescriptionGetStreamBasicDescription(CMSampleBufferGetFormatDescription(sampleBuffer));
SwrContext *swr = swr_alloc();
int in_smprt = (int)audioDescription->mSampleRate;
av_opt_set_int(swr, "in_channel_layout", AV_CH_LAYOUT_MONO, 0);
av_opt_set_int(swr, "out_channel_layout", audio_st->codec->channel_layout, 0);
av_opt_set_int(swr, "in_channel_count", audioDescription->mChannelsPerFrame, 0);
av_opt_set_int(swr, "out_channel_count", 1, 0);
av_opt_set_int(swr, "out_channel_layout", audio_st->codec->channel_layout, 0);
av_opt_set_int(swr, "in_sample_rate", audioDescription->mSampleRate,0);
av_opt_set_int(swr, "out_sample_rate", audio_st->codec->sample_rate,0);
av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_sample_fmt(swr, "out_sample_fmt", audio_st->codec->sample_fmt, 0);
swr_init(swr);
uint8_t **input = NULL;
int src_linesize;
int in_samples = (int)numSamples;
ret = av_samples_alloc_array_and_samples(&input, &src_linesize, audioDescription->mChannelsPerFrame, in_samples, AV_SAMPLE_FMT_S16P, 0);
*input=(uint8_t*)samples;
uint8_t *output=NULL;
int out_samples = av_rescale_rnd(swr_get_delay(swr, in_smprt) +in_samples, (int)audio_st->codec->sample_rate, in_smprt, AV_ROUND_UP);
av_samples_alloc(&output, NULL, audio_st->codec->channels, out_samples, audio_st->codec->sample_fmt, 0);
in_samples = (int)numSamples;
out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **)input, in_samples);
aFrame->nb_samples =(int) out_samples;
ret = avcodec_fill_audio_frame(aFrame, audio_st->codec->channels, audio_st->codec->sample_fmt,
(uint8_t *)output,
(int) out_samples *
av_get_bytes_per_sample(audio_st->codec->sample_fmt) *
audio_st->codec->channels, 1);
if (ret < 0)
{
fprintf(stderr, "Error fill audio frame: %s\n", av_err2str(ret));
}
aFrame->channel_layout = audio_st->codec->channel_layout;
aFrame->channels=audio_st->codec->channels;
aFrame->sample_rate= audio_st->codec->sample_rate;
if (timing_info.presentationTimeStamp.timescale!=0)
pts=(double) timing_info.presentationTimeStamp.value/timing_info.presentationTimeStamp.timescale;
aFrame->pts = pts*audio_st->time_base.den;
aFrame->pts = av_rescale_q(aFrame->pts, audio_st->time_base, audio_st->codec->time_base);
ret = avcodec_encode_audio2(c, &pkt2, aFrame, &got_packet);
if (ret < 0)
{
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
}
swr_free(&swr);
if (got_packet)
{
pkt2.stream_index = audio_st->index;
// Write the compressed frame to the media file.
ret = av_interleaved_write_frame(pFormatCtx, &pkt2);
if (ret != 0)
{
fprintf(stderr, "Error while writing audio frame: %s\n", av_err2str(ret));
av_free_packet(&pkt2);
}
}
}
- (void)captureOutput:(AVCaptureOutput *)captureOutput didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer fromConnection:(AVCaptureConnection *)connection
{
if (connection == videoCaptureConnection)
{
[manager264 encoderToH264:sampleBuffer];
}
else if (connection == audioCaptureConnection)
{
[manager264 encoderToMP3:sampleBuffer];
}
}
- (int)setX264Resource
{
Global_Variables_VVV = (AppDelegate *)[[UIApplication sharedApplication] delegate];
avformat_network_init();
av_register_all();
pFormatCtx = avformat_alloc_context();
avformat_alloc_output_context2(&pFormatCtx, NULL, "flv", out_file);
fmt = pFormatCtx->oformat;
//Open output URL
if (avio_open(&pFormatCtx->pb, out_file, AVIO_FLAG_READ_WRITE) < 0)
{
printf("Failed to open output file! \n");
return -1;
}
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
video_st = NULL;
audio_st = NULL;
if (fmt->video_codec != AV_CODEC_ID_NONE) {
video_st = add_stream(pFormatCtx, &pCodec, AV_CODEC_ID_H264);
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
audio_st = add_stream(pFormatCtx, &aCodec, AV_CODEC_ID_MP3);
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (video_st)
[self open_video:pFormatCtx avcodec:pCodec avstream:video_st];
if (audio_st)
[self open_audio:pFormatCtx avcodec:aCodec avstream:audio_st];
// Show some Information
av_dump_format(pFormatCtx, 0, out_file, 1);
//Write File Header
avformat_write_header(pFormatCtx, NULL);
av_new_packet(&pkt, picture_size);
av_new_packet(&pkt2, picture_size);
AVCodecContext *c = video_st->codec;
y_size = c->width * c->height;
if (pFrame)
pFrame->pts = 0;
if(aFrame)
{
aFrame->pts = 0;
}
return 0;
}
static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec, enum AVCodecID codec_id)
{
AVCodecContext *c;
AVStream *st;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec))
{
NSLog(@"Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
}
st = avformat_new_stream(oc, *codec);
if (!st)
{
NSLog(@"Could not allocate stream\n");
}
st->id = oc->nb_streams-1;
c = st->codec;
switch ((*codec)->type)
{
case AVMEDIA_TYPE_AUDIO:
c->codec_id = AV_CODEC_ID_MP3;
c->codec_type = AVMEDIA_TYPE_AUDIO;
c->channels = 1;
c->sample_fmt = AV_SAMPLE_FMT_S16P;
c->bit_rate = 128000;
c->sample_rate = 44100;
c->channel_layout = AV_CH_LAYOUT_MONO;
break;
case AVMEDIA_TYPE_VIDEO:
c->codec_id = AV_CODEC_ID_H264;
c->codec_type=AVMEDIA_TYPE_VIDEO;
/* Resolution must be a multiple of two. */
c->width = 720;
c->height = 1280;
/* 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 = 30;
c->time_base.num = 1;
c->gop_size = 15; /* emit one intra frame every twelve frames at most */
c->pix_fmt = PIX_FMT_YUV420P;
c->max_b_frames = 0;
c->bit_rate = 3000000;
c->qmin = 10;
c->qmax = 51;
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;
}
通过发送到编码器的样本量增加PTS。 另外，不要忘记从音频流到输出格式上下文重新调整计时 因此，解决办法是：
audioFrame->pts = audioSamplesCounter; // starting from zero
av_packet_rescale_ts(&audioPacket,
audioStream->codec->time_base,
outputFormatContext->streams[audioStream->index]->time_base);
希望它能帮助您：）通过发送到编码器的样本量增加PTS。 另外，不要忘记从音频流到输出格式上下文重新调整计时 因此，解决办法是：
audioFrame->pts = audioSamplesCounter; // starting from zero
av_packet_rescale_ts(&audioPacket,
audioStream->codec->time_base,
outputFormatContext->streams[audioStream->index]->time_base);