static int run_test(AVCodec *enc, AVCodec *dec, AVCodecContext *enc_ctx,
AVCodecContext *dec_ctx)
{
AVPacket enc_pkt;
AVFrame *in_frame, *out_frame;
uint8_t *raw_in = NULL, *raw_out = NULL;
int in_offset = 0, out_offset = 0;
int result = 0;
int got_output = 0;
int i = 0;
int in_frame_bytes, out_frame_bytes;
in_frame = av_frame_alloc();
if (!in_frame) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate input frame\n");
return AVERROR(ENOMEM);
}
in_frame->nb_samples = enc_ctx->frame_size;
in_frame->format = enc_ctx->sample_fmt;
in_frame->channel_layout = enc_ctx->channel_layout;
if (av_frame_get_buffer(in_frame, 32) != 0) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate a buffer for input frame\n");
return AVERROR(ENOMEM);
}
out_frame = av_frame_alloc();
if (!out_frame) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate output frame\n");
return AVERROR(ENOMEM);
}
raw_in = av_malloc(in_frame->linesize[0] * NUMBER_OF_FRAMES);
if (!raw_in) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for raw_in\n");
return AVERROR(ENOMEM);
}
raw_out = av_malloc(in_frame->linesize[0] * NUMBER_OF_FRAMES);
if (!raw_out) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for raw_out\n");
return AVERROR(ENOMEM);
}
for (i = 0; i < NUMBER_OF_FRAMES; i++) {
av_init_packet(&enc_pkt);
enc_pkt.data = NULL;
enc_pkt.size = 0;
generate_raw_frame((uint16_t*)(in_frame->data[0]), i, enc_ctx->sample_rate,
enc_ctx->channels, enc_ctx->frame_size);
in_frame_bytes = in_frame->nb_samples * av_frame_get_channels(in_frame) * sizeof(uint16_t);
if (in_frame_bytes > in_frame->linesize[0]) {
av_log(NULL, AV_LOG_ERROR, "Incorrect value of input frame linesize\n");
return 1;
}
memcpy(raw_in + in_offset, in_frame->data[0], in_frame_bytes);
in_offset += in_frame_bytes;
result = avcodec_encode_audio2(enc_ctx, &enc_pkt, in_frame, &got_output);
if (result < 0) {
av_log(NULL, AV_LOG_ERROR, "Error encoding audio frame\n");
return result;
}
/* if we get an encoded packet, feed it straight to the decoder */
if (got_output) {
result = avcodec_decode_audio4(dec_ctx, out_frame, &got_output, &enc_pkt);
if (result < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding audio packet\n");
return result;
}
if (got_output) {
if (result != enc_pkt.size) {
av_log(NULL, AV_LOG_INFO, "Decoder consumed only part of a packet, it is allowed to do so -- need to update this test\n");
return AVERROR_UNKNOWN;
}
if (in_frame->nb_samples != out_frame->nb_samples) {
av_log(NULL, AV_LOG_ERROR, "Error frames before and after decoding has different number of samples\n");
return AVERROR_UNKNOWN;
}
if (in_frame->channel_layout != out_frame->channel_layout) {
av_log(NULL, AV_LOG_ERROR, "Error frames before and after decoding has different channel layout\n");
return AVERROR_UNKNOWN;
}
if (in_frame->format != out_frame->format) {
av_log(NULL, AV_LOG_ERROR, "Error frames before and after decoding has different sample format\n");
return AVERROR_UNKNOWN;
}
out_frame_bytes = out_frame->nb_samples * av_frame_get_channels(out_frame) * sizeof(uint16_t);
if (out_frame_bytes > out_frame->linesize[0]) {
av_log(NULL, AV_LOG_ERROR, "Incorrect value of output frame linesize\n");
return 1;
}
memcpy(raw_out + out_offset, out_frame->data[0], out_frame_bytes);
out_offset += out_frame_bytes;
}
}
av_free_packet(&enc_pkt);
}
if (memcmp(raw_in, raw_out, out_frame_bytes * NUMBER_OF_FRAMES) != 0) {
av_log(NULL, AV_LOG_ERROR, "Output differs\n");
return 1;
}
av_log(NULL, AV_LOG_INFO, "OK\n");
av_freep(&raw_in);
av_freep(&raw_out);
av_frame_free(&in_frame);
av_frame_free(&out_frame);
return 0;
}
bool AudioEncoder::EncodeFrame(AVFrameWrapper* frame) {
if(frame != NULL) {
#if SSR_USE_AVFRAME_NB_SAMPLES
assert((unsigned int) frame->GetFrame()->nb_samples == GetFrameSize());
#endif
#if SSR_USE_AVFRAME_CHANNELS
assert(frame->GetFrame()->channels == GetCodecContext()->channels);
#endif
#if SSR_USE_AVFRAME_SAMPLE_RATE
assert(frame->GetFrame()->sample_rate == GetCodecContext()->sample_rate);
#endif
#if SSR_USE_AVFRAME_FORMAT
assert(frame->GetFrame()->format == GetCodecContext()->sample_fmt);
#endif
}
#if SSR_USE_AVCODEC_SEND_RECEIVE
// send a frame
AVFrame *avframe = (frame == NULL)? NULL : frame->Release();
try {
if(avcodec_send_frame(GetCodecContext(), avframe) < 0) {
Logger::LogError("[AudioEncoder::EncodeFrame] " + Logger::tr("Error: Sending of audio frame failed!"));
throw LibavException();
}
} catch(...) {
av_frame_free(&avframe);
throw;
}
av_frame_free(&avframe);
// try to receive a packet
for( ; ; ) {
std::unique_ptr<AVPacketWrapper> packet(new AVPacketWrapper());
int res = avcodec_receive_packet(GetCodecContext(), packet->GetPacket());
if(res == 0) { // we have a packet, send the packet to the muxer
GetMuxer()->AddPacket(GetStream()->index, std::move(packet));
IncrementPacketCounter();
} else if(res == AVERROR(EAGAIN)) { // we have no packet
return true;
} else if(res == AVERROR_EOF) { // this is the end of the stream
return false;
} else {
Logger::LogError("[AudioEncoder::EncodeFrame] " + Logger::tr("Error: Receiving of audio packet failed!"));
throw LibavException();
}
}
#elif SSR_USE_AVCODEC_ENCODE_AUDIO2
// allocate a packet
std::unique_ptr<AVPacketWrapper> packet(new AVPacketWrapper());
// encode the frame
int got_packet;
if(avcodec_encode_audio2(GetCodecContext(), packet->GetPacket(), (frame == NULL)? NULL : frame->GetFrame(), &got_packet) < 0) {
Logger::LogError("[AudioEncoder::EncodeFrame] " + Logger::tr("Error: Encoding of audio frame failed!"));
throw LibavException();
}
// do we have a packet?
if(got_packet) {
// send the packet to the muxer
GetMuxer()->AddPacket(GetStream()->index, std::move(packet));
IncrementPacketCounter();
return true;
} else {
return false;
}
#else
// encode the frame
short *data = (frame == NULL)? NULL : (short*) frame->GetFrame()->data[0];
int bytes_encoded = avcodec_encode_audio(GetCodecContext(), m_temp_buffer.data(), m_temp_buffer.size(), data);
if(bytes_encoded < 0) {
Logger::LogError("[AudioEncoder::EncodeFrame] " + Logger::tr("Error: Encoding of audio frame failed!"));
throw LibavException();
}
// do we have a packet?
if(bytes_encoded > 0) {
// allocate a packet
std::unique_ptr<AVPacketWrapper> packet(new AVPacketWrapper(bytes_encoded));
// copy the data
memcpy(packet->GetPacket()->data, m_temp_buffer.data(), bytes_encoded);
// set the timestamp
// note: pts will be rescaled and stream_index will be set by Muxer
if(GetCodecContext()->coded_frame != NULL && GetCodecContext()->coded_frame->pts != (int64_t) AV_NOPTS_VALUE)
packet->GetPacket()->pts = GetCodecContext()->coded_frame->pts;
// send the packet to the muxer
GetMuxer()->AddPacket(GetStream()->index, std::move(packet));
IncrementPacketCounter();
return true;
} else {
return false;
}
#endif
}
AUD_NAMESPACE_BEGIN
void FFMPEGWriter::encode()
{
sample_t* data = m_input_buffer.getBuffer();
if(m_deinterleave)
{
m_deinterleave_buffer.assureSize(m_input_buffer.getSize());
sample_t* dbuf = m_deinterleave_buffer.getBuffer();
// deinterleave
int single_size = sizeof(sample_t);
for(int channel = 0; channel < m_specs.channels; channel++)
{
for(int i = 0; i < m_input_buffer.getSize() / AUD_SAMPLE_SIZE(m_specs); i++)
{
std::memcpy(((data_t*)dbuf) + (m_input_samples * channel + i) * single_size,
((data_t*)data) + ((m_specs.channels * i) + channel) * single_size, single_size);
}
}
// convert first
if(m_input_size)
m_convert(reinterpret_cast<data_t*>(data), reinterpret_cast<data_t*>(dbuf), m_input_samples * m_specs.channels);
else
std::memcpy(data, dbuf, m_input_buffer.getSize());
}
else
// convert first
if(m_input_size)
m_convert(reinterpret_cast<data_t*>(data), reinterpret_cast<data_t*>(data), m_input_samples * m_specs.channels);
AVPacket packet;
packet.data = nullptr;
packet.size = 0;
av_init_packet(&packet);
AVFrame* frame = av_frame_alloc();
av_frame_unref(frame);
int got_packet;
frame->nb_samples = m_input_samples;
frame->format = m_codecCtx->sample_fmt;
frame->channel_layout = m_codecCtx->channel_layout;
if(avcodec_fill_audio_frame(frame, m_specs.channels, m_codecCtx->sample_fmt, reinterpret_cast<data_t*>(data), m_input_buffer.getSize(), 0) < 0)
AUD_THROW(FileException, "File couldn't be written, filling the audio frame failed with ffmpeg.");
AVRational sample_time = { 1, static_cast<int>(m_specs.rate) };
frame->pts = av_rescale_q(m_position - m_input_samples, m_codecCtx->time_base, sample_time);
if(avcodec_encode_audio2(m_codecCtx, &packet, frame, &got_packet))
{
av_frame_free(&frame);
AUD_THROW(FileException, "File couldn't be written, audio encoding failed with ffmpeg.");
}
if(got_packet)
{
packet.flags |= AV_PKT_FLAG_KEY;
packet.stream_index = m_stream->index;
if(av_write_frame(m_formatCtx, &packet) < 0)
{
av_free_packet(&packet);
av_frame_free(&frame);
AUD_THROW(FileException, "Frame couldn't be writen to the file with ffmpeg.");
}
av_free_packet(&packet);
}
av_frame_free(&frame);
}
static bool encode_audio(ffmpeg_t *handle, AVPacket *pkt, bool dry)
{
av_init_packet(pkt);
pkt->data = handle->audio.outbuf;
pkt->size = handle->audio.outbuf_size;
AVFrame *frame = av_frame_alloc();
if (!frame)
return false;
frame->nb_samples = handle->audio.frames_in_buffer;
frame->format = handle->audio.codec->sample_fmt;
frame->channel_layout = handle->audio.codec->channel_layout;
frame->pts = handle->audio.frame_cnt;
planarize_audio(handle);
int samples_size = av_samples_get_buffer_size(NULL, handle->audio.codec->channels,
handle->audio.frames_in_buffer,
handle->audio.codec->sample_fmt, 0);
avcodec_fill_audio_frame(frame, handle->audio.codec->channels,
handle->audio.codec->sample_fmt,
handle->audio.is_planar ? (uint8_t*)handle->audio.planar_buf : handle->audio.buffer,
samples_size, 0);
int got_packet = 0;
if (avcodec_encode_audio2(handle->audio.codec,
pkt, dry ? NULL : frame, &got_packet) < 0)
{
av_frame_free(&frame);
return false;
}
if (!got_packet)
{
pkt->size = 0;
pkt->pts = AV_NOPTS_VALUE;
pkt->dts = AV_NOPTS_VALUE;
av_frame_free(&frame);
return true;
}
if (pkt->pts != (int64_t)AV_NOPTS_VALUE)
{
pkt->pts = av_rescale_q(pkt->pts,
handle->audio.codec->time_base,
handle->muxer.astream->time_base);
}
if (pkt->dts != (int64_t)AV_NOPTS_VALUE)
{
pkt->dts = av_rescale_q(pkt->dts,
handle->audio.codec->time_base,
handle->muxer.astream->time_base);
}
av_frame_free(&frame);
pkt->stream_index = handle->muxer.astream->index;
return true;
}
MediaRecorder::MediaRecorder(const char * outfile,int width, int height)
{
audiofailed = false;
/* INIT SOUND RECORDING */
debug_samples_out = fopen("audiosamples.s16","wb");
audio_samples_written = 0;
pa_context* pactx;
pa_mainloop * m = pa_mainloop_new();
m_api = pa_mainloop_get_api(m);
pactx = pa_context_new(m_api,"Rec1");
if ( pa_context_connect(pactx,NULL,(pa_context_flags_t)0,NULL) < 0 )
printf("Cannot connect to pulseaudio\n");
int ret;
pa_context_set_state_callback(pactx, context_state_callback, this);
pa_mainloop_run(m,&ret);
std::cout << "Use source: " << monitorsources[defaultsink] << std::endl;
static const pa_sample_spec ss = {
.format = PA_SAMPLE_S16LE,
.rate = 44100,
.channels = 2
};
pa_context_disconnect(pactx);
int error;
s = pa_simple_new(NULL,"GLCAP Record",PA_STREAM_RECORD,monitorsources[defaultsink].c_str(), "record", &ss, NULL,NULL , &error);
if ( !s )
{
printf("Cannot create pa_simple\n");
}
run = true;
ready = false;
firstframe = true;
this->width = width;
this->height = height;
pthread_mutex_init(&encode_mutex,NULL);
pthread_mutex_init(&sound_buffer_lock,NULL);
pthread_cond_init(&encode_cond,NULL);
pthread_create(&encode_thread,NULL,(void*(*)(void*))&MediaRecorder::EncodingThread,this);
av_log_set_level(AV_LOG_DEBUG);
outCtx = avformat_alloc_context();
outCtx->oformat = av_guess_format(NULL, outfile, NULL);
snprintf(outCtx->filename, sizeof(outCtx->filename), "%s", outfile);
codec = avcodec_find_encoder(AV_CODEC_ID_MPEG4);
acodec = avcodec_find_encoder(AV_CODEC_ID_MP2);
ctx = avcodec_alloc_context3(codec);
actx = avcodec_alloc_context3(acodec);
avcodec_get_context_defaults3(actx,acodec);
avcodec_get_context_defaults3(ctx,codec);
ctx->width = width;
ctx->height = height;
ctx->bit_rate = 6000*1000;
std::cout << ctx->time_base.den << " " << ctx->time_base.num << std::endl;
ctx->time_base.den = TIMEBASE;
ctx->time_base.num = 1;
ctx->thread_count = 4;
ctx->qmin = 2;
ctx->qmax = 31;
ctx->b_sensitivity = 100;
ctx->gop_size = 1;
ctx->me_method = 1;
ctx->global_quality = 100;
ctx->lowres = 0;
ctx->bit_rate_tolerance = 200000;
actx->sample_fmt = AV_SAMPLE_FMT_S16;
actx->sample_rate = 44100;
actx->channels = 2;
actx->time_base.den = 44100;
actx->time_base.num = 1;
actx->bit_rate = 128000;
actx->frame_size = 8192;
actx->channel_layout = 3;
/* ctx->compression_level = 0;
ctx->trellis = 0;
ctx->gop_size = 1; /* emit one intra frame every ten frames */
/*ctx->me_pre_cmp = 0;
ctx->me_cmp = 0;
ctx->me_sub_cmp = 0;
ctx->mb_cmp = 2;
ctx->pre_dia_size = 0;
ctx->dia_size = 1;
ctx->quantizer_noise_shaping = 0; // qns=0
ctx->noise_reduction = 0; // nr=0
ctx->mb_decision = 0; // mbd=0 ("realtime" encoding)
ctx->flags &= ~CODEC_FLAG_QPEL;
ctx->flags &= ~CODEC_FLAG_4MV;
ctx->trellis = 0;
ctx->flags &= ~CODEC_FLAG_CBP_RD;
ctx->flags &= ~CODEC_FLAG_QP_RD;
ctx->flags &= ~CODEC_FLAG_MV0;*/
//ctx->s
ctx->pix_fmt = PIX_FMT_YUV420P;
if (avcodec_open2(ctx, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
}
if (avcodec_open2(actx, acodec, NULL) < 0) {
fprintf(stderr, "Could not open audio codec\n");
audiofailed = true;
}
printf("frame_size: %d\n",actx->frame_size);
pthread_create(&record_sound_thread,NULL,(void*(*)(void*))&MediaRecorder::RecordingThread,this);
AVStream* s = av_new_stream(outCtx,0);
s->codec = ctx;
s->r_frame_rate.den = TIMEBASE;
s->r_frame_rate.num = 1;
if (!audiofailed )
{
AVStream* as = av_new_stream(outCtx,1);
as->codec = actx;
as->r_frame_rate.den = 44100;
as->r_frame_rate.num = 1;
}
picture = alloc_picture(PIX_FMT_YUV420P, ctx->width, ctx->height);
if (!picture) {
fprintf(stderr, "Could not allocate picture\n");
exit(1);
}
tmp_picture = NULL;
tmp_picture = alloc_picture(PIX_FMT_RGBA, ctx->width, ctx->height);
if (!tmp_picture) {
fprintf(stderr, "Could not allocate temporary picture\n");
exit(1);
}
img_convert_ctx = sws_getContext(ctx->width, ctx->height,
PIX_FMT_RGBA,
ctx->width, ctx->height,
PIX_FMT_YUV420P,
SWS_FAST_BILINEAR , NULL, NULL, NULL);
if (img_convert_ctx == NULL) {
fprintf(stderr,
"Cannot initialize the conversion context\n");
exit(1);
}
av_dump_format(outCtx, 0, outfile, 1);
avio_open2(&outCtx->pb, outfile, AVIO_FLAG_WRITE,NULL,NULL);
avformat_write_header(outCtx,NULL);
}
MediaRecorder::~MediaRecorder()
{
run = false;
ready = false;
pthread_cond_broadcast(&encode_cond);
printf("Joining thread..\n");
pthread_join(encode_thread,NULL);
printf("Joining recording thread..\n");
pthread_join(record_sound_thread,NULL);
printf("Done\n");
av_write_trailer(outCtx);
av_free(picture);
avformat_free_context(outCtx);
pa_simple_free(s);
fclose(debug_samples_out);
}
int fcount = 0;
void MediaRecorder::AppendFrame(float time, int width, int height, char* data)
{
if ( !ready )
return ;
printf("AppendFrame\n");
this->time = getcurrenttime2();
if ( firstframe )
{
starttime = getcurrenttime2();
firstframe = false;
}
this->height = height;
this->width = width;
m_data = data;
ready = false;
pthread_cond_broadcast(&encode_cond);
/*int i = 0;
unsigned int numpixels = width * height;
unsigned int ui = numpixels;
unsigned int vi = numpixels + numpixels / 4;
for ( int j = 0; j < height; j++ )
{
for ( int k = 0; k < width; k++ )
{
int sR = data[i*4+0];
int sG = data[i*4+1];
int sB = data[i*4+2];
picture->data[0][i] = ( (66*sR + 129*sG + 25*sB + 128) >> 8) + 16;
if (0 == j%2 && 0 == k%2)
{
picture->data[0][ui++] = ( (-38*sR - 74*sG + 112*sB + 128) >> 8) + 128;
picture->data[0][vi++] = ( (112*sR - 94*sG - 18*sB + 128) >> 8) + 128;
}
i++;
}
}*/
// printf("End flip %f\n",(float)getcurrenttime2());
//memcpy(tmp_picture->data[0],data,width*height*4);
}
void MediaRecorder::EncodingThread()
{
while ( run )
{
printf("Encode thread ready\n");
ready = true;
pthread_cond_wait(&encode_cond,&encode_mutex);
if (!run)
{
printf("Encoding finished\n");
break;
}
for ( int y = 0; y < height; y++ )
{
/*for ( int x = 0; x < width; x++ )
{*/
char r,g,b;
int oldindex = (y*width);
int newindex = ((height-1-y)*width);
memcpy(&tmp_picture->data[0][(newindex)*4],&m_data[oldindex*4],width*4);
/* r = data[oldindex*4+0];
g = data[oldindex*4+1];
b = data[oldindex*4+2];
tmp_picture->data[0][(newindex)*4+0] = r;
tmp_picture->data[0][(newindex)*4+1] = g;
tmp_picture->data[0][(newindex)*4+2] = b; */
// }
}
sws_scale(img_convert_ctx,tmp_picture->data,tmp_picture->linesize,0,height,picture->data,picture->linesize);
AVPacket p;
av_init_packet(&p);
p.data = NULL;
p.size = 0;
picture->pts = int64_t((time-starttime)*TIMEBASE);
uint64_t vpts = picture->pts;
// picture->pts = time*30.0;
int got_frame;
printf("%p %p\n",ctx, picture);
if(avcodec_encode_video2(ctx, &p, picture, &got_frame) < 0) return;
if(got_frame)
{
// outContainer is "mp4"
p.pts = vpts;
p.dts = AV_NOPTS_VALUE;
av_write_frame(outCtx, &p);
av_free_packet(&p);
}
//sleep(1);
printf("End enc frame %f, pts %lld\n",(float)getcurrenttime2(),picture->pts);
AVFrame * aframe = avcodec_alloc_frame();
bool unlocked = false;
while ( sound_buffers.size() > 0 )
{
uint64_t apts = audio_samples_written;
/* if ( apts > vpts )
break;*/
pthread_mutex_lock(&sound_buffer_lock);
short * buf = sound_buffers.front();
sound_buffers.pop_front();
pthread_mutex_unlock(&sound_buffer_lock);
if (!audiofailed )
{
unlocked = true;
aframe->nb_samples = actx->frame_size;
aframe->channel_layout = actx->channel_layout;
aframe->format = AV_SAMPLE_FMT_S16;
aframe->channels = actx->channels;
avcodec_fill_audio_frame(aframe,actx->channels,AV_SAMPLE_FMT_S16,(char*)buf,actx->frame_size*2*2,0);
// avcodec_fill_audio_frame(aframe,actx->channels,actx->sample_fmt,(char*)buf,actx->frame_size*2,0);
printf("sound_buffers.size() = %d\n",sound_buffers.size());
av_init_packet(&p);
p.data = NULL;
p.size = 0;
avcodec_encode_audio2(actx,&p,aframe,&got_frame);
if ( got_frame )
{
p.stream_index = 1;
p.flags |= AV_PKT_FLAG_KEY;
av_write_frame(outCtx,&p);
av_free_packet(&p);
}
audio_samples_written += actx->frame_size;//samples/2 each channel
}
//printf("Consumed 1024 samples\n");
delete[] buf;
}
/* if ( !unlocked )
pthread_mutex_unlock(&sound_buffer_lock);*/
avcodec_free_frame(&aframe);
}
}
bool MediaRecorder::isReady()
{
return ready;
}
int AVFormatWriter::WriteAudioFrame(unsigned char *buf, int fnum, long long &timecode)
{
#if HAVE_BIGENDIAN
bswap_16_buf((short int*) buf, m_audioFrameSize, m_audioChannels);
#endif
int got_packet = 0;
int ret = 0;
int samples_per_avframe = m_audioFrameSize * m_audioChannels;
int sampleSizeIn = AudioOutputSettings::SampleSize(FORMAT_S16);
AudioFormat format =
AudioOutputSettings::AVSampleFormatToFormat(m_audioStream->codec->sample_fmt);
int sampleSizeOut = AudioOutputSettings::SampleSize(format);
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (av_get_packed_sample_fmt(m_audioStream->codec->sample_fmt) == AV_SAMPLE_FMT_FLT)
{
AudioOutputUtil::toFloat(FORMAT_S16, (void *)m_audioInBuf, (void *)buf,
samples_per_avframe * sampleSizeIn);
buf = m_audioInBuf;
}
if (av_sample_fmt_is_planar(m_audioStream->codec->sample_fmt))
{
AudioOutputUtil::DeinterleaveSamples(format,
m_audioChannels,
m_audioInPBuf,
buf,
samples_per_avframe * sampleSizeOut);
// init AVFrame for planar data (input is interleaved)
for (int j = 0, jj = 0; j < m_audioChannels; j++, jj += m_audioFrameSize)
{
m_audPicture->data[j] = (uint8_t*)(m_audioInPBuf + jj * sampleSizeOut);
}
}
else
{
m_audPicture->data[0] = buf;
}
m_audPicture->linesize[0] = m_audioFrameSize;
m_audPicture->nb_samples = m_audioFrameSize;
m_audPicture->format = m_audioStream->codec->sample_fmt;
m_audPicture->extended_data = m_audPicture->data;
m_bufferedAudioFrameTimes.push_back(timecode);
{
QMutexLocker locker(avcodeclock);
ret = avcodec_encode_audio2(m_audioStream->codec, &pkt,
m_audPicture, &got_packet);
}
if (ret < 0)
{
LOG(VB_RECORD, LOG_ERR, "avcodec_encode_audio2() failed");
return ret;
}
if (!got_packet)
{
//LOG(VB_RECORD, LOG_ERR, QString("WriteAudioFrame(): Frame Buffered: cs: %1, mfw: %2, f->tc: %3, fn: %4").arg(m_audPkt->size).arg(m_framesWritten).arg(timecode).arg(fnum));
return ret;
}
long long tc = timecode;
if (m_bufferedAudioFrameTimes.size())
tc = m_bufferedAudioFrameTimes.takeFirst();
if (m_startingTimecodeOffset == -1)
m_startingTimecodeOffset = tc - 1;
tc -= m_startingTimecodeOffset;
if (m_avVideoCodec)
pkt.pts = tc * m_videoStream->time_base.den / m_videoStream->time_base.num / 1000;
else
pkt.pts = tc * m_audioStream->time_base.den / m_audioStream->time_base.num / 1000;
pkt.dts = AV_NOPTS_VALUE;
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = m_audioStream->index;
//LOG(VB_RECORD, LOG_ERR, QString("WriteAudioFrame(): cs: %1, mfw: %2, pkt->pts: %3, tc: %4, fn: %5, f->tc: %6").arg(m_audPkt->size).arg(m_framesWritten).arg(m_audPkt->pts).arg(tc).arg(fnum).arg(timecode));
ret = av_interleaved_write_frame(m_ctx, &pkt);
if (ret != 0)
LOG(VB_RECORD, LOG_ERR, LOC + "WriteAudioFrame(): "
"av_interleaved_write_frame couldn't write Audio");
timecode = tc + m_startingTimecodeOffset;
av_free_packet(&pkt);
return 1;
}
void Movie::EncodeAudio(bool last)
{
AVStream *astream = av->fmt_ctx->streams[av->audio_stream_idx];
AVCodecContext *acodec = astream->codec;
av_fifo_generic_write(av->audio_fifo, &audiobuf[0], audiobuf.size(), NULL);
// bps: bytes per sample
int channels = acodec->channels;
int read_bps = 2;
int write_bps = av_get_bytes_per_sample(acodec->sample_fmt);
int max_read = acodec->frame_size * read_bps * channels;
int min_read = last ? read_bps * channels : max_read;
while (av_fifo_size(av->audio_fifo) >= min_read)
{
int read_bytes = MIN(av_fifo_size(av->audio_fifo), max_read);
av_fifo_generic_read(av->audio_fifo, av->audio_data, read_bytes, NULL);
// convert
int read_samples = read_bytes / (read_bps * channels);
int write_samples = read_samples;
if (read_samples < acodec->frame_size)
{
// shrink or pad audio frame
if (acodec->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME)
acodec->frame_size = write_samples;
else
write_samples = acodec->frame_size;
}
convert_audio(read_samples, acodec->channels, -1,
AV_SAMPLE_FMT_S16, av->audio_data,
write_samples, acodec->channels, write_samples * write_bps,
acodec->sample_fmt, av->audio_data_conv);
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(55,28,1)
avcodec_get_frame_defaults(av->audio_frame);
#else
av_frame_unref(av->audio_frame);
#endif
av->audio_frame->nb_samples = write_samples;
av->audio_frame->pts = av_rescale_q(av->audio_counter,
(AVRational){1, acodec->sample_rate},
acodec->time_base);
av->audio_counter += write_samples;
int asize = avcodec_fill_audio_frame(av->audio_frame, acodec->channels,
acodec->sample_fmt,
av->audio_data_conv,
write_samples * write_bps * channels, 1);
if (asize >= 0)
{
AVPacket pkt;
memset(&pkt, 0, sizeof(AVPacket));
av_init_packet(&pkt);
int got_pkt = 0;
if (0 == avcodec_encode_audio2(acodec, &pkt, av->audio_frame, &got_pkt)
&& got_pkt)
{
if (pkt.pts != AV_NOPTS_VALUE && pkt.pts < pkt.dts)
pkt.pts = pkt.dts;
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, acodec->time_base, astream->time_base);
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, acodec->time_base, astream->time_base);
pkt.duration = av_rescale_q(pkt.duration, acodec->time_base, astream->time_base);
pkt.stream_index = astream->index;
av_interleaved_write_frame(av->fmt_ctx, &pkt);
av_free_packet(&pkt);
}
}
}
if (last)
{
bool done = false;
while (!done)
{
AVPacket pkt;
memset(&pkt, 0, sizeof(AVPacket));
av_init_packet(&pkt);
int got_pkt = 0;
if (0 == avcodec_encode_audio2(acodec, &pkt, NULL, &got_pkt)
&& got_pkt)
{
if (pkt.pts != AV_NOPTS_VALUE && pkt.pts < pkt.dts)
pkt.pts = pkt.dts;
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, acodec->time_base, astream->time_base);
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, acodec->time_base, astream->time_base);
pkt.duration = av_rescale_q(pkt.duration, acodec->time_base, astream->time_base);
pkt.stream_index = astream->index;
av_interleaved_write_frame(av->fmt_ctx, &pkt);
av_free_packet(&pkt);
}
else
{
done = true;
}
}
}
}
static void write_audio_frame(AVFormatContext *oc, AVStream *st)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
int got_packet, ret, dst_nb_samples;
av_init_packet(&pkt);
c = st->codec;
get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
/* convert samples from native format to destination codec format, using the resampler */
if (swr_ctx) {
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_samples_data[0]);
ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
dst_nb_samples, c->sample_fmt, 0);
if (ret < 0)
exit(1);
max_dst_nb_samples = dst_nb_samples;
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
c->sample_fmt, 0);
}
/* convert to destination format */
ret = swr_convert(swr_ctx,
dst_samples_data, dst_nb_samples,
(const uint8_t **)src_samples_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
} else {
dst_nb_samples = src_nb_samples;
}
audio_frame->nb_samples = dst_nb_samples;
audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
dst_samples_data[0], dst_samples_size, 0);
samples_count += dst_nb_samples;
ret = avcodec_encode_audio2(c, &pkt, audio_frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (!got_packet)
return;
/* rescale output packet timestamp values from codec to stream timebase */
pkt.pts = av_rescale_q_rnd(pkt.pts, c->time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.dts = av_rescale_q_rnd(pkt.dts, c->time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.duration = av_rescale_q(pkt.duration, c->time_base, st->time_base);
pkt.stream_index = st->index;
/* Write the compressed frame to the media file. */
ret = av_interleaved_write_frame(oc, &pkt);
if (ret != 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
}
int udpsocket::ts_demux(void)
{
AVCodec *pVideoCodec[VIDEO_NUM];
AVCodec *pAudioCodec[AUDIO_NUM];
AVCodecContext *pVideoCodecCtx[VIDEO_NUM];
AVCodecContext *pAudioCodecCtx[AUDIO_NUM];
AVIOContext * pb;
AVInputFormat *piFmt;
AVFormatContext *pFmt;
uint8_t *buffer;
int videoindex[VIDEO_NUM];
int audioindex[AUDIO_NUM];
AVStream *pVst[VIDEO_NUM];
AVStream *pAst[AUDIO_NUM];
AVFrame *pVideoframe[VIDEO_NUM];
AVFrame *pAudioframe[AUDIO_NUM];
AVFrame *pOutAudioframe[AUDIO_NUM];
AVFrame *pOutAudioframelast[AUDIO_NUM];
AVPacket pkt;
int got_picture;
int video_num[VIDEO_NUM];
int audio_num[AUDIO_NUM];
int frame_size;
//transcodepool
transcodepool* pVideoTransPool[VIDEO_NUM];
transcodepool* pAudioTransPool[AUDIO_NUM];
for( int i=0; i<VIDEO_NUM; i++ ){
pVideoCodec[i] = NULL;
pVideoCodecCtx[i] =NULL;
videoindex[i] = -1;
pVst[i] = NULL;
video_num[i] = 0;
pVideoframe[i] = NULL;
pVideoframe[i] = av_frame_alloc();
pVideoTransPool[i] = NULL;
}
for( int i=0; i<AUDIO_NUM; i++ ){
pAudioCodec[i] = NULL;
pAudioCodecCtx[i] = NULL;
audioindex[i] = -1;
pAst[i] = NULL;
audio_num[i] = 0;
pOutAudioframe[i] = NULL;
pOutAudioframe[i] = av_frame_alloc();
pOutAudioframelast[i] = NULL;
pOutAudioframelast[i] = av_frame_alloc();
pAudioframe[i] = NULL;
pAudioframe[i] = av_frame_alloc();
pAudioTransPool[i] = NULL;
}
pb = NULL;
piFmt = NULL;
pFmt = NULL;
buffer = (uint8_t*)av_mallocz(sizeof(uint8_t)*BUFFER_SIZE);
got_picture = 0;
frame_size = AVCODEC_MAX_AUDIO_FRAME_SIZE*3/2;
//encoder
AVFormatContext *ofmt_ctx = NULL;
AVPacket enc_pkt;
AVStream *out_stream;
AVCodecContext *enc_ctx;
AVCodec *encoder;
AVFormatContext *outAudioFormatCtx[AUDIO_NUM];
AVPacket audio_pkt;
AVStream *audio_stream[AUDIO_NUM];
AVCodecContext *AudioEncodeCtx[AUDIO_NUM];
AVCodec *AudioEncoder[AUDIO_NUM];
fp_v = fopen("OUT.h264","wb+"); //输出文件
fp_a = fopen("audio_out.aac","wb+");
//FFMPEG
av_register_all();
pb = avio_alloc_context(buffer, 4096, 0, NULL, read_data, NULL, NULL);
// printf("thread %d pid %lu tid %lu\n",index,(unsigned long)getpid(),(unsigned long)pthread_self());
if (!pb) {
fprintf(stderr, "avio alloc failed!\n");
return -1;
}
int x = av_probe_input_buffer(pb, &piFmt, "", NULL, 0, 0);
if (x < 0) {
printf("probe error: %d",x);
// fprintf(stderr, "probe failed!\n");
} else {
fprintf(stdout, "probe success!\n");
fprintf(stdout, "format: %s[%s]\n", piFmt->name, piFmt->long_name);
}
pFmt = avformat_alloc_context();
pFmt->pb = pb;
if (avformat_open_input(&pFmt, "", piFmt, NULL) < 0) {
fprintf(stderr, "avformat open failed.\n");
return -1;
} else {
fprintf(stdout, "open stream success!\n");
}
//pFmt->probesize = 4096 * 2000;
//pFmt->max_analyze_duration = 5 * AV_TIME_BASE;
//pFmt->probesize = 2048;
// pFmt->max_analyze_duration = 1000;
pFmt->probesize = 2048 * 1000 ;
pFmt->max_analyze_duration = 2048 * 1000;
if (avformat_find_stream_info(pFmt,0) < 0) {
fprintf(stderr, "could not fine stream.\n");
return -1;
}
printf("dump format\n");
av_dump_format(pFmt, 0, "", 0);
int videox = 0,audiox = 0;
for (int i = 0; i < pFmt->nb_streams; i++) {
if(videox == 7 && audiox == 7)
break;
if ( pFmt->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO && videox < 7 ) {
videoindex[ videox++ ] = i;
}
if ( pFmt->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO && audiox < 7 ) {
audioindex[ audiox++ ] = i;
}
}
for(int i=0; i<VIDEO_NUM; i++)
printf("videoindex %d = %d, audioindex %d = %d\n",i , videoindex[i], i ,audioindex[i]);
if (videoindex[6] < 0 || audioindex[6] < 0) {
fprintf(stderr, "videoindex=%d, audioindex=%d\n", videoindex[6], audioindex[6]);
return -1;
}
for( int i=0; i<VIDEO_NUM; i++ ){
pVst[i] = pFmt->streams[videoindex[i]];
pVideoCodecCtx[i] = pVst[i]->codec;
pVideoCodec[i] = avcodec_find_decoder(pVideoCodecCtx[i]->codec_id);
if (!pVideoCodec[i]) {
fprintf(stderr, "could not find video decoder!\n");
return -1;
}
if (avcodec_open2(pVideoCodecCtx[i], pVideoCodec[i], NULL) < 0) {
fprintf(stderr, "could not open video codec!\n");
return -1;
}
}
for( int i=0; i<AUDIO_NUM; i++ ){
pAst[i] = pFmt->streams[audioindex[i]];
pAudioCodecCtx[i] = pAst[i]->codec;
pAudioCodec[i] = avcodec_find_decoder(pAudioCodecCtx[i]->codec_id);
if (!pAudioCodec[i]) {
fprintf(stderr, "could not find audio decoder!\n");
return -1;
}
if (avcodec_open2(pAudioCodecCtx[i], pAudioCodec[i], NULL) < 0) {
fprintf(stderr, "could not open audio codec!\n");
return -1;
}
}
//video encoder init
avformat_alloc_output_context2(&ofmt_ctx, NULL, "h264", NULL);
unsigned char* outbuffer = NULL;
outbuffer = (unsigned char*)av_malloc(1024*1000);
AVIOContext *avio_out = NULL;
avio_out = avio_alloc_context(outbuffer, 1024*1000, 0, NULL, NULL, write_buffer,NULL);
if(avio_out == NULL){
printf("avio_out error\n");
return -1;
}
ofmt_ctx->pb = avio_out;
ofmt_ctx->flags = AVFMT_FLAG_CUSTOM_IO;
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if(!out_stream){
av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
return -1;
}
enc_ctx = out_stream->codec;
encoder = avcodec_find_encoder(AV_CODEC_ID_H264);
enc_ctx->height = pVideoCodecCtx[0]->height;
enc_ctx->width = pVideoCodecCtx[0]->width;
enc_ctx->sample_aspect_ratio = pVideoCodecCtx[0]->sample_aspect_ratio;
enc_ctx->pix_fmt = encoder->pix_fmts[0];
out_stream->time_base = pVst[0]->time_base;
// out_stream->time_base.num = 1;
// out_stream->time_base.den = 25;
enc_ctx->me_range = 16;
enc_ctx->max_qdiff = 4;
enc_ctx->qmin = 25;
enc_ctx->qmax = 40;
enc_ctx->qcompress = 0.6;
enc_ctx->refs = 3;
enc_ctx->bit_rate = 1000000;
int re = avcodec_open2(enc_ctx, encoder, NULL);
if (re < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream \n");
return re;
}
if(ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
re = avformat_write_header(ofmt_ctx, NULL);
if(re < 0){
av_log(NULL, AV_LOG_ERROR, "Error occured when opening output file\n");
return re;
}
//audio encoder
for( int i=0; i<AUDIO_NUM; i++){
outAudioFormatCtx[i] = NULL;
// audio_pkt = NULL;
audio_stream[i] = NULL;
AudioEncodeCtx[i] = NULL;
AudioEncoder[i] = NULL;
}
const char* out_audio_file = "transcodeaudio.aac"; //Output URL
//Method 1.
outAudioFormatCtx[0] = avformat_alloc_context();
outAudioFormatCtx[0]->oformat = av_guess_format(NULL, out_audio_file, NULL);
AVIOContext *avio_audio_out = NULL;
avio_audio_out = avio_alloc_context(outbuffer, 1024*1000, 0, NULL, NULL, write_buffer,NULL);
if(avio_audio_out == NULL){
printf("avio_out error\n");
return -1;
}
outAudioFormatCtx[0]->pb = avio_audio_out;
//Method 2.
//avformat_alloc_output_context2(&pFormatCtx, NULL, NULL, out_file);
//fmt = pFormatCtx->oformat;
//Open output URL
if (avio_open(&outAudioFormatCtx[0]->pb,out_audio_file, AVIO_FLAG_READ_WRITE) < 0){
printf("Failed to open output file!\n");
return -1;
}
//Show some information
av_dump_format(outAudioFormatCtx[0], 0, out_audio_file, 1);
AudioEncoder[0] = avcodec_find_encoder(AV_CODEC_ID_AAC);
if (!AudioEncoder[0]){
printf("Can not find encoder!\n");
return -1;
}
audio_stream[0] = avformat_new_stream(outAudioFormatCtx[0], AudioEncoder[0]);
if (audio_stream[0]==NULL){
return -1;
}
AudioEncodeCtx[0] = audio_stream[0]->codec;
AudioEncodeCtx[0]->codec_id = outAudioFormatCtx[0]->oformat->audio_codec;
AudioEncodeCtx[0]->codec_type = AVMEDIA_TYPE_AUDIO;
AudioEncodeCtx[0]->sample_fmt = AV_SAMPLE_FMT_S16;
AudioEncodeCtx[0]->sample_rate= 48000;//44100
AudioEncodeCtx[0]->channel_layout=AV_CH_LAYOUT_STEREO;
AudioEncodeCtx[0]->channels = av_get_channel_layout_nb_channels(AudioEncodeCtx[0]->channel_layout);
AudioEncodeCtx[0]->bit_rate = 64000;//64000
/** Allow the use of the experimental AAC encoder */
AudioEncodeCtx[0]->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/** Set the sample rate for the container. */
audio_stream[0]->time_base.den = pAudioCodecCtx[0]->sample_rate;
audio_stream[0]->time_base.num = 1;
if (avcodec_open2(AudioEncodeCtx[0], AudioEncoder[0],NULL) < 0){
printf("Failed to open encoder!\n");
return -1;
}
av_samples_get_buffer_size(NULL, AudioEncodeCtx[0]->channels,AudioEncodeCtx[0]->frame_size,AudioEncodeCtx[0]->sample_fmt, 1);
//uint8_t samples[AVCODEC_MAX_AUDIO_FRAME_SIZE*3/2];
av_init_packet(&pkt);
av_init_packet(&audio_pkt);
av_init_packet(&enc_pkt);
AVAudioFifo *af = NULL;
SwrContext *resample_context = NULL;
long long pts = 0;
/** Initialize the resampler to be able to convert audio sample formats. */
// if (init_resampler(input_codec_context, output_codec_context,
// &resample_context))
for(int i=0; i<1; i++){
printf("work \n");
printf(" samplerate input = %d , samplerate output = %d\n",pAudioCodecCtx[i]->sample_rate, AudioEncodeCtx[i]->sample_rate);
resample_context = swr_alloc_set_opts(NULL, av_get_default_channel_layout(AudioEncodeCtx[i]->channels),
AudioEncodeCtx[i]->sample_fmt,
AudioEncodeCtx[i]->sample_rate,
av_get_default_channel_layout(pAudioCodecCtx[i]->channels),
pAudioCodecCtx[i]->sample_fmt,
pAudioCodecCtx[i]->sample_rate,
0, NULL);
swr_init(resample_context);
}
af = av_audio_fifo_alloc(AudioEncodeCtx[0]->sample_fmt, AudioEncodeCtx[0]->channels, 1);
if(af == NULL)
{
printf("error af \n");
return -1;
}
while(1) {
if (av_read_frame(pFmt, &pkt) >= 0) {
for( int i=0; i<1; i++ ){
if (pkt.stream_index == videoindex[i]) {
// av_frame_free(&pframe);
avcodec_decode_video2(pVideoCodecCtx[i], pVideoframe[i], &got_picture, &pkt);
if (got_picture) {
if(videoindex[i] == 0){
// m_tsRecvPool->write_buffer(pkt.data, pkt.size);
pVideoframe[i]->pts = av_frame_get_best_effort_timestamp(pVideoframe[i]);
pVideoframe[i]->pict_type = AV_PICTURE_TYPE_NONE;
// printf("videoframesize0 = %d, size1 = %d, size2 = %d, size3 = %d, size4 = %d,format = %d\n",pVideoframe[i]->linesize[0],
// pVideoframe[i]->linesize[1],pVideoframe[i]->linesize[2],pVideoframe[i]->linesize[3],pVideoframe[i]->linesize[4],pVideoframe[i]->format);
// pVideoTransPool[i]->PutFrame( pVideoframe[i] ,i);
int enc_got_frame = 0;
/* ffmpeg encoder */
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
re = avcodec_encode_video2(ofmt_ctx->streams[videoindex[i]]->codec, &enc_pkt,
pVideoframe[i], &enc_got_frame);
// printf("enc_got_frame =%d, re = %d \n",enc_got_frame, re);
printf("video Encode 1 Packet\tsize:%d\tpts:%lld\n",enc_pkt.size,enc_pkt.pts);
/* prepare packet for muxing */
// fwrite(enc_pkt.data,enc_pkt.size, 1, fp_v);
}
// printf(" video %d decode %d num\n", i, video_num[i]++);
break;
}
}else if (pkt.stream_index == audioindex[i]) {
if (avcodec_decode_audio4(pAudioCodecCtx[i], pAudioframe[i], &frame_size, &pkt) >= 0) {
if (i == 0){
// fwrite(pAudioframe[i]->data[0],pAudioframe[i]->linesize[0], 1, fp_a);
// printf("index = %d audio %d decode %d num\n", index, i, audio_num[i]++);
uint8_t *converted_input_samples = NULL;
converted_input_samples = (uint8_t *)calloc(AudioEncodeCtx[i]->channels, sizeof(*converted_input_samples));
av_samples_alloc(&converted_input_samples, NULL, AudioEncodeCtx[i]->channels, pAudioframe[i]->nb_samples, AudioEncodeCtx[i]->sample_fmt, 0);
int error = 0;
if((error = swr_convert(resample_context, &converted_input_samples, pAudioframe[i]->nb_samples,
(const uint8_t**)pAudioframe[i]->extended_data, pAudioframe[i]->nb_samples))<0){
printf("error : %d\n",error);
}
// av_audio_fifo_realloc(af, av_audio_fifo_size(af) + pAudioframe[i]->nb_samples);
av_audio_fifo_write(af, (void **)&converted_input_samples, pAudioframe[i]->nb_samples);
// fwrite(pkt.data,pkt.size, 1, fp_a);
// pAudioframe[i]->data[0] = frame_buf;
// init_converted_samples(&converted_input_samples, output_codec_context, pAudioframe[i]->nb_samples);
/** Initialize temporary storage for one output frame. */
// printf("pkt.size = %d , pkt.pts = %d ,pkt.dts = %d\n",pkt.size, pkt.pts, pkt.dts);
// printf("framesize = %d, audioframesize = %d\n", pAudioframe[i]->nb_samples, frame_size);
// pOutAudioframe[i]->pict_type = AV_PICTURE_TYPE_NONE;
int got_frame=0;
//Encode
// av_init_packet(&audio_pkt);
// audio_pkt.data = NULL;
// audio_pkt.size = 0;
// avcodec_encode_audio2(AudioEncodeCtx[0], &audio_pkt, pOutAudioframe[i], &got_frame);
// printf("Encode 1 Packet\tsize:%d\tpts:%lld\n", audio_pkt.size, audio_pkt.pts);
while(av_audio_fifo_size(af) >= AudioEncodeCtx[i]->frame_size){
int frame_size = FFMIN(av_audio_fifo_size(af),AudioEncodeCtx[0]->frame_size);
pOutAudioframe[i]->nb_samples = frame_size;
pOutAudioframe[i]->channel_layout = AudioEncodeCtx[0]->channel_layout;
pOutAudioframe[i]->sample_rate = AudioEncodeCtx[0]->sample_rate;
pOutAudioframe[i]->format = AudioEncodeCtx[0]->sample_fmt;
av_frame_get_buffer(pOutAudioframe[i], 0);
av_audio_fifo_read(af, (void **)&pOutAudioframe[i]->data, frame_size);
pOutAudioframe[i]->pts=pts;
pts += pOutAudioframe[i]->nb_samples;
audio_pkt.data = NULL;
audio_pkt.size = 0;
av_init_packet(&audio_pkt);
avcodec_encode_audio2(AudioEncodeCtx[0], &audio_pkt, pOutAudioframe[i], &got_frame);
printf("audio Encode 1 Packet\tsize:%d\tpts:%lld\n", audio_pkt.size, audio_pkt.pts);
fwrite(audio_pkt.data,audio_pkt.size, 1, fp_a);
}
}
// if(i == 0){
// fwrite(pkt.data,pkt.size, 1, fp_a);
// }
// printf("index = %d audio %d decode %d num\n", index, i, audio_num[i]++);
break;
}
}
}
av_free_packet(&pkt);
av_free_packet(&enc_pkt);
}
}
av_free(buffer);
for(int i=0; i<VIDEO_NUM; i++)
av_free(pVideoframe[i]);
for(int i=0; i<AUDIO_NUM; i++)
av_free(pAudioframe[i]);
return 0;
}
void _ffmpegPostAudioFrame(struct mAVStream* stream, int16_t left, int16_t right) {
struct FFmpegEncoder* encoder = (struct FFmpegEncoder*) stream;
if (!encoder->context || !encoder->audioCodec) {
return;
}
encoder->audioBuffer[encoder->currentAudioSample * 2] = left;
encoder->audioBuffer[encoder->currentAudioSample * 2 + 1] = right;
++encoder->currentAudioFrame;
++encoder->currentAudioSample;
if ((encoder->currentAudioSample * 4) < encoder->audioBufferSize) {
return;
}
encoder->currentAudioSample = 0;
int channelSize = 2 * av_get_bytes_per_sample(encoder->audio->sample_fmt);
avresample_convert(encoder->resampleContext,
0, 0, 0,
(uint8_t**) &encoder->audioBuffer, 0, encoder->audioBufferSize / 4);
if (avresample_available(encoder->resampleContext) < encoder->audioFrame->nb_samples) {
return;
}
#if LIBAVCODEC_VERSION_MAJOR >= 55
av_frame_make_writable(encoder->audioFrame);
#endif
avresample_read(encoder->resampleContext, encoder->audioFrame->data, encoder->postaudioBufferSize / channelSize);
AVRational timeBase = { 1, PREFERRED_SAMPLE_RATE };
encoder->audioFrame->pts = encoder->nextAudioPts;
encoder->nextAudioPts = av_rescale_q(encoder->currentAudioFrame, timeBase, encoder->audioStream->time_base);
AVPacket packet;
av_init_packet(&packet);
packet.data = 0;
packet.size = 0;
int gotData;
#ifdef FFMPEG_USE_PACKETS
avcodec_send_frame(encoder->audio, encoder->audioFrame);
gotData = avcodec_receive_packet(encoder->audio, &packet) == 0;
#else
avcodec_encode_audio2(encoder->audio, &packet, encoder->audioFrame, &gotData);
#endif
if (gotData) {
if (encoder->absf) {
AVPacket tempPacket = packet;
#ifdef FFMPEG_USE_NEW_BSF
int success = av_bsf_send_packet(encoder->absf, &packet) && av_bsf_receive_packet(encoder->absf, &packet);
#else
int success = av_bitstream_filter_filter(encoder->absf, encoder->audio, 0,
&tempPacket.data, &tempPacket.size,
packet.data, packet.size, 0);
#endif
if (success > 0) {
#if LIBAVUTIL_VERSION_MAJOR >= 53
tempPacket.buf = av_buffer_create(tempPacket.data, tempPacket.size, av_buffer_default_free, 0, 0);
#endif
#ifdef FFMPEG_USE_PACKET_UNREF
av_packet_unref(&packet);
#else
av_free_packet(&packet);
#endif
}
packet = tempPacket;
}
packet.stream_index = encoder->audioStream->index;
av_interleaved_write_frame(encoder->context, &packet);
}
#ifdef FFMPEG_USE_PACKET_UNREF
av_packet_unref(&packet);
#else
av_free_packet(&packet);
#endif
}
static int encode_audio(AVCodecContext *avctx, AVPacket *pkt, int16_t *audio_samples, int nb_samples)
{
int i, ch, buffer_size, ret, got_output = 0;
void *samples = NULL;
AVFrame *frame = NULL;
if (audio_samples) {
frame = av_frame_alloc();
if (!frame)
return AVERROR(ENOMEM);
frame->nb_samples = nb_samples;
frame->format = avctx->sample_fmt;
#if !defined(DISABLE_DYNAMIC_LOADING_FFMPEG) || (LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 13, 0))
frame->channel_layout = avctx->channel_layout;
#endif
buffer_size = av_samples_get_buffer_size(NULL, avctx->channels, frame->nb_samples,
avctx->sample_fmt, 0);
if (buffer_size < 0) {
wxMessageBox(wxString::Format(_("FFmpeg : ERROR - Could not get sample buffer size")),
_("FFmpeg Error"), wxOK|wxCENTER|wxICON_EXCLAMATION);
return buffer_size;
}
samples = av_malloc(buffer_size);
if (!samples) {
wxMessageBox(wxString::Format(_("FFmpeg : ERROR - Could not allocate bytes for samples buffer")),
_("FFmpeg Error"), wxOK|wxCENTER|wxICON_EXCLAMATION);
return AVERROR(ENOMEM);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, avctx->channels, avctx->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
wxMessageBox(wxString::Format(_("FFmpeg : ERROR - Could not setup audio frame")),
_("FFmpeg Error"), wxOK|wxCENTER|wxICON_EXCLAMATION);
return ret;
}
for (ch = 0; ch < avctx->channels; ch++) {
for (i = 0; i < frame->nb_samples; i++) {
switch(avctx->sample_fmt) {
case AV_SAMPLE_FMT_U8:
((uint8_t*)(frame->data[0]))[ch + i*avctx->channels] = audio_samples[ch + i*avctx->channels]/258 + 128;
break;
case AV_SAMPLE_FMT_U8P:
((uint8_t*)(frame->data[ch]))[i] = audio_samples[ch + i*avctx->channels]/258 + 128;
break;
case AV_SAMPLE_FMT_S16:
((int16_t*)(frame->data[0]))[ch + i*avctx->channels] = audio_samples[ch + i*avctx->channels];
break;
case AV_SAMPLE_FMT_S16P:
((int16_t*)(frame->data[ch]))[i] = audio_samples[ch + i*avctx->channels];
break;
case AV_SAMPLE_FMT_S32:
((int32_t*)(frame->data[0]))[ch + i*avctx->channels] = audio_samples[ch + i*avctx->channels]<<16;
break;
case AV_SAMPLE_FMT_S32P:
((int32_t*)(frame->data[ch]))[i] = audio_samples[ch + i*avctx->channels]<<16;
break;
case AV_SAMPLE_FMT_FLT:
((float*)(frame->data[0]))[ch + i*avctx->channels] = audio_samples[ch + i*avctx->channels] / 32767.0;
break;
case AV_SAMPLE_FMT_FLTP:
((float*)(frame->data[ch]))[i] = audio_samples[ch + i*avctx->channels] / 32767.;
break;
case AV_SAMPLE_FMT_NONE:
case AV_SAMPLE_FMT_DBL:
case AV_SAMPLE_FMT_DBLP:
case AV_SAMPLE_FMT_NB:
wxASSERT(false);
break;
}
}
}
}
av_init_packet(pkt);
pkt->data = NULL; // packet data will be allocated by the encoder
pkt->size = 0;
ret = avcodec_encode_audio2(avctx, pkt, frame, &got_output);
if (ret < 0) {
wxMessageBox(wxString::Format(_("FFmpeg : ERROR - encoding frame failed")),
_("FFmpeg Error"), wxOK|wxCENTER|wxICON_EXCLAMATION);
return ret;
}
pkt->dts = pkt->pts = AV_NOPTS_VALUE; // we dont set frame.pts thus dont trust the AVPacket ts
av_frame_free(&frame);
av_freep(&samples);
return got_output;
}
static void encode_audio_and_write(struct ao *ao, AVFrame *frame)
{
// TODO: Can we unify this with the equivalent video code path?
struct priv *ac = ao->priv;
AVPacket packet = {0};
#if HAVE_AVCODEC_NEW_CODEC_API
int status = avcodec_send_frame(ac->codec, frame);
if (status < 0) {
MP_ERR(ao, "error encoding at %d %d/%d\n",
frame ? (int) frame->pts : -1,
ac->codec->time_base.num,
ac->codec->time_base.den);
return;
}
for (;;) {
av_init_packet(&packet);
status = avcodec_receive_packet(ac->codec, &packet);
if (status == AVERROR(EAGAIN)) { // No more packets for now.
if (frame == NULL) {
MP_ERR(ao, "sent flush frame, got EAGAIN");
}
break;
}
if (status == AVERROR_EOF) { // No more packets, ever.
if (frame != NULL) {
MP_ERR(ao, "sent audio frame, got EOF");
}
break;
}
if (status < 0) {
MP_ERR(ao, "error encoding at %d %d/%d\n",
frame ? (int) frame->pts : -1,
ac->codec->time_base.num,
ac->codec->time_base.den);
break;
}
if (frame) {
if (ac->savepts == AV_NOPTS_VALUE)
ac->savepts = frame->pts;
}
encode_lavc_write_stats(ao->encode_lavc_ctx, ac->codec);
write_packet(ao, &packet);
av_packet_unref(&packet);
}
#else
av_init_packet(&packet);
int got_packet = 0;
int status = avcodec_encode_audio2(ac->codec, &packet, frame, &got_packet);
if (status < 0) {
MP_ERR(ao, "error encoding at %d %d/%d\n",
frame ? (int) frame->pts : -1,
ac->codec->time_base.num,
ac->codec->time_base.den);
return;
}
if (!got_packet) {
return;
}
if (frame) {
if (ac->savepts == AV_NOPTS_VALUE)
ac->savepts = frame->pts;
}
encode_lavc_write_stats(ao->encode_lavc_ctx, ac->codec);
write_packet(ao, &packet);
av_packet_unref(&packet);
#endif
}
int main(int argc, char* argv[])
{
AVFormatContext *ifmt_ctx = NULL;
AVFormatContext *ifmt_ctx_a = NULL;
AVFormatContext *ofmt_ctx;
AVInputFormat* ifmt;
AVStream* video_st;
AVStream* audio_st;
AVCodecContext* pCodecCtx;
AVCodecContext* pCodecCtx_a;
AVCodec* pCodec;
AVCodec* pCodec_a;
AVPacket *dec_pkt, enc_pkt;
AVPacket *dec_pkt_a, enc_pkt_a;
AVFrame *pframe, *pFrameYUV;
struct SwsContext *img_convert_ctx;
struct SwrContext *aud_convert_ctx;
char capture_name[80] = { 0 };
char device_name[80] = { 0 };
char device_name_a[80] = { 0 };
int framecnt = 0;
int nb_samples = 0;
int videoindex;
int audioindex;
int i;
int ret;
HANDLE hThread;
const char* out_path = "rtmp://localhost/live/livestream";
int dec_got_frame, enc_got_frame;
int dec_got_frame_a, enc_got_frame_a;
int aud_next_pts = 0;
int vid_next_pts = 0;
int encode_video = 1, encode_audio = 1;
AVRational time_base_q = { 1, AV_TIME_BASE };
av_register_all();
//Register Device
avdevice_register_all();
avformat_network_init();
#if USEFILTER
//Register Filter
avfilter_register_all();
buffersrc = avfilter_get_by_name("buffer");
buffersink = avfilter_get_by_name("buffersink");
#endif
//Show Dshow Device
show_dshow_device();
printf("\nChoose video capture device: ");
if (gets(capture_name) == 0)
{
printf("Error in gets()\n");
return -1;
}
sprintf(device_name, "video=%s", capture_name);
printf("\nChoose audio capture device: ");
if (gets(capture_name) == 0)
{
printf("Error in gets()\n");
return -1;
}
sprintf(device_name_a, "audio=%s", capture_name);
//wchar_t *cam = L"video=Integrated Camera";
//wchar_t *cam = L"video=YY伴侣";
//char *device_name_utf8 = dup_wchar_to_utf8(cam);
//wchar_t *cam_a = L"audio=麦克风阵列 (Realtek High Definition Audio)";
//char *device_name_utf8_a = dup_wchar_to_utf8(cam_a);
ifmt = av_find_input_format("dshow");
// Set device params
AVDictionary *device_param = 0;
//if not setting rtbufsize, error messages will be shown in cmd, but you can still watch or record the stream correctly in most time
//setting rtbufsize will erase those error messages, however, larger rtbufsize will bring latency
//av_dict_set(&device_param, "rtbufsize", "10M", 0);
//Set own video device's name
if (avformat_open_input(&ifmt_ctx, device_name, ifmt, &device_param) != 0){
printf("Couldn't open input video stream.（无法打开输入流）\n");
return -1;
}
//Set own audio device's name
if (avformat_open_input(&ifmt_ctx_a, device_name_a, ifmt, &device_param) != 0){
printf("Couldn't open input audio stream.（无法打开输入流）\n");
return -1;
}
//input video initialize
if (avformat_find_stream_info(ifmt_ctx, NULL) < 0)
{
printf("Couldn't find video stream information.（无法获取流信息）\n");
return -1;
}
videoindex = -1;
for (i = 0; i < ifmt_ctx->nb_streams; i++)
if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
{
videoindex = i;
break;
}
if (videoindex == -1)
{
printf("Couldn't find a video stream.（没有找到视频流）\n");
return -1;
}
if (avcodec_open2(ifmt_ctx->streams[videoindex]->codec, avcodec_find_decoder(ifmt_ctx->streams[videoindex]->codec->codec_id), NULL) < 0)
{
printf("Could not open video codec.（无法打开解码器）\n");
return -1;
}
//input audio initialize
if (avformat_find_stream_info(ifmt_ctx_a, NULL) < 0)
{
printf("Couldn't find audio stream information.（无法获取流信息）\n");
return -1;
}
audioindex = -1;
for (i = 0; i < ifmt_ctx_a->nb_streams; i++)
if (ifmt_ctx_a->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
{
audioindex = i;
break;
}
if (audioindex == -1)
{
printf("Couldn't find a audio stream.（没有找到视频流）\n");
return -1;
}
if (avcodec_open2(ifmt_ctx_a->streams[audioindex]->codec, avcodec_find_decoder(ifmt_ctx_a->streams[audioindex]->codec->codec_id), NULL) < 0)
{
printf("Could not open audio codec.（无法打开解码器）\n");
return -1;
}
//output initialize
avformat_alloc_output_context2(&ofmt_ctx, NULL, "flv", out_path);
//output video encoder initialize
pCodec = avcodec_find_encoder(AV_CODEC_ID_H264);
if (!pCodec){
printf("Can not find output video encoder! (没有找到合适的编码器！)\n");
return -1;
}
pCodecCtx = avcodec_alloc_context3(pCodec);
pCodecCtx->pix_fmt = PIX_FMT_YUV420P;
pCodecCtx->width = ifmt_ctx->streams[videoindex]->codec->width;
pCodecCtx->height = ifmt_ctx->streams[videoindex]->codec->height;
pCodecCtx->time_base.num = 1;
pCodecCtx->time_base.den = 25;
pCodecCtx->bit_rate = 300000;
pCodecCtx->gop_size = 250;
/* Some formats want stream headers to be separate. */
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
pCodecCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
//H264 codec param
//pCodecCtx->me_range = 16;
//pCodecCtx->max_qdiff = 4;
//pCodecCtx->qcompress = 0.6;
pCodecCtx->qmin = 10;
pCodecCtx->qmax = 51;
//Optional Param
pCodecCtx->max_b_frames = 0;
// Set H264 preset and tune
AVDictionary *param = 0;
av_dict_set(¶m, "preset", "fast", 0);
av_dict_set(¶m, "tune", "zerolatency", 0);
if (avcodec_open2(pCodecCtx, pCodec, ¶m) < 0){
printf("Failed to open output video encoder! (编码器打开失败！)\n");
return -1;
}
//Add a new stream to output,should be called by the user before avformat_write_header() for muxing
video_st = avformat_new_stream(ofmt_ctx, pCodec);
if (video_st == NULL){
return -1;
}
video_st->time_base.num = 1;
video_st->time_base.den = 25;
video_st->codec = pCodecCtx;
//output audio encoder initialize
pCodec_a = avcodec_find_encoder(AV_CODEC_ID_AAC);
if (!pCodec_a){
printf("Can not find output audio encoder! (没有找到合适的编码器！)\n");
return -1;
}
pCodecCtx_a = avcodec_alloc_context3(pCodec_a);
pCodecCtx_a->channels = 2;
pCodecCtx_a->channel_layout = av_get_default_channel_layout(2);
pCodecCtx_a->sample_rate = ifmt_ctx_a->streams[audioindex]->codec->sample_rate;
pCodecCtx_a->sample_fmt = pCodec_a->sample_fmts[0];
pCodecCtx_a->bit_rate = 32000;
pCodecCtx_a->time_base.num = 1;
pCodecCtx_a->time_base.den = pCodecCtx_a->sample_rate;
/** Allow the use of the experimental AAC encoder */
pCodecCtx_a->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/* Some formats want stream headers to be separate. */
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
pCodecCtx_a->flags |= CODEC_FLAG_GLOBAL_HEADER;
if (avcodec_open2(pCodecCtx_a, pCodec_a, NULL) < 0){
printf("Failed to open ouput audio encoder! (编码器打开失败！)\n");
return -1;
}
//Add a new stream to output,should be called by the user before avformat_write_header() for muxing
audio_st = avformat_new_stream(ofmt_ctx, pCodec_a);
if (audio_st == NULL){
return -1;
}
audio_st->time_base.num = 1;
audio_st->time_base.den = pCodecCtx_a->sample_rate;
audio_st->codec = pCodecCtx_a;
//Open output URL,set before avformat_write_header() for muxing
if (avio_open(&ofmt_ctx->pb, out_path, AVIO_FLAG_READ_WRITE) < 0){
printf("Failed to open output file! (输出文件打开失败！)\n");
return -1;
}
//Show some Information
av_dump_format(ofmt_ctx, 0, out_path, 1);
//Write File Header
avformat_write_header(ofmt_ctx, NULL);
//prepare before decode and encode
dec_pkt = (AVPacket *)av_malloc(sizeof(AVPacket));
#if USEFILTER
#else
//camera data may has a pix fmt of RGB or sth else,convert it to YUV420
img_convert_ctx = sws_getContext(ifmt_ctx->streams[videoindex]->codec->width, ifmt_ctx->streams[videoindex]->codec->height,
ifmt_ctx->streams[videoindex]->codec->pix_fmt, pCodecCtx->width, pCodecCtx->height, PIX_FMT_YUV420P, SWS_BICUBIC, NULL, NULL, NULL);
// Initialize the resampler to be able to convert audio sample formats
aud_convert_ctx = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(pCodecCtx_a->channels),
pCodecCtx_a->sample_fmt,
pCodecCtx_a->sample_rate,
av_get_default_channel_layout(ifmt_ctx_a->streams[audioindex]->codec->channels),
ifmt_ctx_a->streams[audioindex]->codec->sample_fmt,
ifmt_ctx_a->streams[audioindex]->codec->sample_rate,
0, NULL);
/**
* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
//av_assert0(pCodecCtx_a->sample_rate == ifmt_ctx_a->streams[audioindex]->codec->sample_rate);
swr_init(aud_convert_ctx);
#endif
//Initialize the buffer to store YUV frames to be encoded.
pFrameYUV = av_frame_alloc();
uint8_t *out_buffer = (uint8_t *)av_malloc(avpicture_get_size(PIX_FMT_YUV420P, pCodecCtx->width, pCodecCtx->height));
avpicture_fill((AVPicture *)pFrameYUV, out_buffer, PIX_FMT_YUV420P, pCodecCtx->width, pCodecCtx->height);
//Initialize the FIFO buffer to store audio samples to be encoded.
AVAudioFifo *fifo = NULL;
fifo = av_audio_fifo_alloc(pCodecCtx_a->sample_fmt, pCodecCtx_a->channels, 1);
//Initialize the buffer to store converted samples to be encoded.
uint8_t **converted_input_samples = NULL;
/**
* Allocate as many pointers as there are audio channels.
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
if (!(converted_input_samples = (uint8_t**)calloc(pCodecCtx_a->channels,
sizeof(**converted_input_samples)))) {
printf("Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
printf("\n --------call started----------\n");
#if USEFILTER
printf("\n Press differnet number for different filters:");
printf("\n 1->Mirror");
printf("\n 2->Add Watermark");
printf("\n 3->Negate");
printf("\n 4->Draw Edge");
printf("\n 5->Split Into 4");
printf("\n 6->Vintage");
printf("\n Press 0 to remove filter\n");
#endif
printf("\nPress enter to stop...\n");
hThread = CreateThread(
NULL, // default security attributes
0, // use default stack size
MyThreadFunction, // thread function name
NULL, // argument to thread function
0, // use default creation flags
NULL); // returns the thread identifier
//start decode and encode
int64_t start_time = av_gettime();
while (encode_video || encode_audio)
{
if (encode_video &&
(!encode_audio || av_compare_ts(vid_next_pts, time_base_q,
aud_next_pts, time_base_q) <= 0))
{
if ((ret=av_read_frame(ifmt_ctx, dec_pkt)) >= 0){
if (exit_thread)
break;
av_log(NULL, AV_LOG_DEBUG, "Going to reencode the frame\n");
pframe = av_frame_alloc();
if (!pframe) {
ret = AVERROR(ENOMEM);
return ret;
}
ret = avcodec_decode_video2(ifmt_ctx->streams[dec_pkt->stream_index]->codec, pframe,
&dec_got_frame, dec_pkt);
if (ret < 0) {
av_frame_free(&pframe);
av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
break;
}
if (dec_got_frame){
#if USEFILTER
pframe->pts = av_frame_get_best_effort_timestamp(pframe);
if (filter_change)
apply_filters(ifmt_ctx);
filter_change = 0;
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame(buffersrc_ctx, pframe) < 0) {
printf("Error while feeding the filtergraph\n");
break;
}
picref = av_frame_alloc();
/* pull filtered pictures from the filtergraph */
while (1) {
ret = av_buffersink_get_frame_flags(buffersink_ctx, picref, 0);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
return ret;
if (picref) {
img_convert_ctx = sws_getContext(picref->width, picref->height, (AVPixelFormat)picref->format, pCodecCtx->width, pCodecCtx->height, AV_PIX_FMT_YUV420P, SWS_BICUBIC, NULL, NULL, NULL);
sws_scale(img_convert_ctx, (const uint8_t* const*)picref->data, picref->linesize, 0, pCodecCtx->height, pFrameYUV->data, pFrameYUV->linesize);
sws_freeContext(img_convert_ctx);
pFrameYUV->width = picref->width;
pFrameYUV->height = picref->height;
pFrameYUV->format = PIX_FMT_YUV420P;
#else
sws_scale(img_convert_ctx, (const uint8_t* const*)pframe->data, pframe->linesize, 0, pCodecCtx->height, pFrameYUV->data, pFrameYUV->linesize);
pFrameYUV->width = pframe->width;
pFrameYUV->height = pframe->height;
pFrameYUV->format = PIX_FMT_YUV420P;
#endif
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
ret = avcodec_encode_video2(pCodecCtx, &enc_pkt, pFrameYUV, &enc_got_frame);
av_frame_free(&pframe);
if (enc_got_frame == 1){
//printf("Succeed to encode frame: %5d\tsize:%5d\n", framecnt, enc_pkt.size);
framecnt++;
enc_pkt.stream_index = video_st->index;
//Write PTS
AVRational time_base = ofmt_ctx->streams[0]->time_base;//{ 1, 1000 };
AVRational r_framerate1 = ifmt_ctx->streams[videoindex]->r_frame_rate;//{ 50, 2 };
//Duration between 2 frames (us)
int64_t calc_duration = (double)(AV_TIME_BASE)*(1 / av_q2d(r_framerate1)); //内部时间戳
//Parameters
//enc_pkt.pts = (double)(framecnt*calc_duration)*(double)(av_q2d(time_base_q)) / (double)(av_q2d(time_base));
enc_pkt.pts = av_rescale_q(framecnt*calc_duration, time_base_q, time_base);
enc_pkt.dts = enc_pkt.pts;
enc_pkt.duration = av_rescale_q(calc_duration, time_base_q, time_base); //(double)(calc_duration)*(double)(av_q2d(time_base_q)) / (double)(av_q2d(time_base));
enc_pkt.pos = -1;
//printf("video pts : %d\n", enc_pkt.pts);
vid_next_pts=framecnt*calc_duration; //general timebase
//Delay
int64_t pts_time = av_rescale_q(enc_pkt.pts, time_base, time_base_q);
int64_t now_time = av_gettime() - start_time;
if ((pts_time > now_time) && ((vid_next_pts + pts_time - now_time)<aud_next_pts))
av_usleep(pts_time - now_time);
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
av_free_packet(&enc_pkt);
}
#if USEFILTER
av_frame_unref(picref);
}
}
#endif
}
else {
av_frame_free(&pframe);
}
av_free_packet(dec_pkt);
}
else
if (ret == AVERROR_EOF)
encode_video = 0;
else
{
printf("Could not read video frame\n");
return ret;
}
}
else
{
//audio trancoding here
const int output_frame_size = pCodecCtx_a->frame_size;
if (exit_thread)
break;
/**
* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples.
*/
while (av_audio_fifo_size(fifo) < output_frame_size) {
/**
* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer.
*/
AVFrame *input_frame = av_frame_alloc();
if (!input_frame)
{
ret = AVERROR(ENOMEM);
return ret;
}
/** Decode one frame worth of audio samples. */
/** Packet used for temporary storage. */
AVPacket input_packet;
av_init_packet(&input_packet);
input_packet.data = NULL;
input_packet.size = 0;
/** Read one audio frame from the input file into a temporary packet. */
if ((ret = av_read_frame(ifmt_ctx_a, &input_packet)) < 0) {
/** If we are at the end of the file, flush the decoder below. */
if (ret == AVERROR_EOF)
{
encode_audio = 0;
}
else
{
printf("Could not read audio frame\n");
return ret;
}
}
/**
* Decode the audio frame stored in the temporary packet.
* The input audio stream decoder is used to do this.
* If we are at the end of the file, pass an empty packet to the decoder
* to flush it.
*/
if ((ret = avcodec_decode_audio4(ifmt_ctx_a->streams[audioindex]->codec, input_frame,
&dec_got_frame_a, &input_packet)) < 0) {
printf("Could not decode audio frame\n");
return ret;
}
av_packet_unref(&input_packet);
/** If there is decoded data, convert and store it */
if (dec_got_frame_a) {
/**
* Allocate memory for the samples of all channels in one consecutive
* block for convenience.
*/
if ((ret = av_samples_alloc(converted_input_samples, NULL,
pCodecCtx_a->channels,
input_frame->nb_samples,
pCodecCtx_a->sample_fmt, 0)) < 0) {
printf("Could not allocate converted input samples\n");
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return ret;
}
/**
* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples.
*/
/** Convert the samples using the resampler. */
if ((ret = swr_convert(aud_convert_ctx,
converted_input_samples, input_frame->nb_samples,
(const uint8_t**)input_frame->extended_data, input_frame->nb_samples)) < 0) {
printf("Could not convert input samples\n");
return ret;
}
/** Add the converted input samples to the FIFO buffer for later processing. */
/**
* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples.
*/
if ((ret = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + input_frame->nb_samples)) < 0) {
printf("Could not reallocate FIFO\n");
return ret;
}
/** Store the new samples in the FIFO buffer. */
if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
input_frame->nb_samples) < input_frame->nb_samples) {
printf("Could not write data to FIFO\n");
return AVERROR_EXIT;
}
}
}
/**
* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder.
*/
if (av_audio_fifo_size(fifo) >= output_frame_size)
/**
* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file.
*/
{
/** Temporary storage of the output samples of the frame written to the file. */
AVFrame *output_frame=av_frame_alloc();
if (!output_frame)
{
ret = AVERROR(ENOMEM);
return ret;
}
/**
* Use the maximum number of possible samples per frame.
* If there is less than the maximum possible frame size in the FIFO
* buffer use this number. Otherwise, use the maximum possible frame size
*/
const int frame_size = FFMIN(av_audio_fifo_size(fifo),
pCodecCtx_a->frame_size);
/** Initialize temporary storage for one output frame. */
/**
* Set the frame's parameters, especially its size and format.
* av_frame_get_buffer needs this to allocate memory for the
* audio samples of the frame.
* Default channel layouts based on the number of channels
* are assumed for simplicity.
*/
output_frame->nb_samples = frame_size;
output_frame->channel_layout = pCodecCtx_a->channel_layout;
output_frame->format = pCodecCtx_a->sample_fmt;
output_frame->sample_rate = pCodecCtx_a->sample_rate;
/**
* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified.
*/
if ((ret = av_frame_get_buffer(output_frame, 0)) < 0) {
printf("Could not allocate output frame samples\n");
av_frame_free(&output_frame);
return ret;
}
/**
* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily.
*/
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
printf("Could not read data from FIFO\n");
return AVERROR_EXIT;
}
/** Encode one frame worth of audio samples. */
/** Packet used for temporary storage. */
AVPacket output_packet;
av_init_packet(&output_packet);
output_packet.data = NULL;
output_packet.size = 0;
/** Set a timestamp based on the sample rate for the container. */
if (output_frame) {
nb_samples += output_frame->nb_samples;
}
/**
* Encode the audio frame and store it in the temporary packet.
* The output audio stream encoder is used to do this.
*/
if ((ret = avcodec_encode_audio2(pCodecCtx_a, &output_packet,
output_frame, &enc_got_frame_a)) < 0) {
printf("Could not encode frame\n");
av_packet_unref(&output_packet);
return ret;
}
/** Write one audio frame from the temporary packet to the output file. */
if (enc_got_frame_a) {
output_packet.stream_index = 1;
AVRational time_base = ofmt_ctx->streams[1]->time_base;
AVRational r_framerate1 = { ifmt_ctx_a->streams[audioindex]->codec->sample_rate, 1 };// { 44100, 1};
int64_t calc_duration = (double)(AV_TIME_BASE)*(1 / av_q2d(r_framerate1)); //内部时间戳
output_packet.pts = av_rescale_q(nb_samples*calc_duration, time_base_q, time_base);
output_packet.dts = output_packet.pts;
output_packet.duration = output_frame->nb_samples;
//printf("audio pts : %d\n", output_packet.pts);
aud_next_pts = nb_samples*calc_duration;
int64_t pts_time = av_rescale_q(output_packet.pts, time_base, time_base_q);
int64_t now_time = av_gettime() - start_time;
if ((pts_time > now_time) && ((aud_next_pts + pts_time - now_time)<vid_next_pts))
av_usleep(pts_time - now_time);
if ((ret = av_interleaved_write_frame(ofmt_ctx, &output_packet)) < 0) {
printf("Could not write frame\n");
av_packet_unref(&output_packet);
return ret;
}
av_packet_unref(&output_packet);
}
av_frame_free(&output_frame);
}
}
}
//Flush Encoder
ret = flush_encoder(ifmt_ctx, ofmt_ctx, 0, framecnt);
if (ret < 0) {
printf("Flushing encoder failed\n");
return -1;
}
ret = flush_encoder_a(ifmt_ctx_a, ofmt_ctx, 1, nb_samples);
if (ret < 0) {
printf("Flushing encoder failed\n");
return -1;
}
//Write file trailer
av_write_trailer(ofmt_ctx);
cleanup:
//Clean
#if USEFILTER
if (filter_graph)
avfilter_graph_free(&filter_graph);
#endif
if (video_st)
avcodec_close(video_st->codec);
if (audio_st)
avcodec_close(audio_st->codec);
av_free(out_buffer);
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
//free(converted_input_samples);
}
if (fifo)
av_audio_fifo_free(fifo);
avio_close(ofmt_ctx->pb);
avformat_free_context(ifmt_ctx);
avformat_free_context(ifmt_ctx_a);
avformat_free_context(ofmt_ctx);
CloseHandle(hThread);
return 0;
}
static int write_audio_frame(FFMpegContext *context)
{
AVCodecContext *c = NULL;
AVPacket pkt;
AVFrame *frame = NULL;
int got_output = 0;
c = context->audio_stream->codec;
av_init_packet(&pkt);
pkt.size = 0;
pkt.data = NULL;
AUD_Device_read(context->audio_mixdown_device, context->audio_input_buffer, context->audio_input_samples);
context->audio_time += (double) context->audio_input_samples / (double) c->sample_rate;
#ifdef FFMPEG_HAVE_ENCODE_AUDIO2
frame = avcodec_alloc_frame();
avcodec_get_frame_defaults(frame);
frame->pts = context->audio_time / av_q2d(c->time_base);
frame->nb_samples = context->audio_input_samples;
frame->format = c->sample_fmt;
#ifdef FFMPEG_HAVE_FRAME_CHANNEL_LAYOUT
frame->channel_layout = c->channel_layout;
#endif
if (context->audio_deinterleave) {
int channel, i;
uint8_t *temp;
for (channel = 0; channel < c->channels; channel++) {
for (i = 0; i < frame->nb_samples; i++) {
memcpy(context->audio_deinterleave_buffer + (i + channel * frame->nb_samples) * context->audio_sample_size,
context->audio_input_buffer + (c->channels * i + channel) * context->audio_sample_size, context->audio_sample_size);
}
}
temp = context->audio_deinterleave_buffer;
context->audio_deinterleave_buffer = context->audio_input_buffer;
context->audio_input_buffer = temp;
}
avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, context->audio_input_buffer,
context->audio_input_samples * c->channels * context->audio_sample_size, 1);
if (avcodec_encode_audio2(c, &pkt, frame, &got_output) < 0) {
// XXX error("Error writing audio packet");
return -1;
}
if (!got_output) {
avcodec_free_frame(&frame);
return 0;
}
#else
pkt.size = avcodec_encode_audio(c, context->audio_output_buffer, context->audio_outbuf_size, (short *) context->audio_input_buffer);
if (pkt.size < 0) {
// XXX error("Error writing audio packet");
return -1;
}
pkt.data = context->audio_output_buffer;
got_output = 1;
#endif
if (got_output) {
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, c->time_base, context->audio_stream->time_base);
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, c->time_base, context->audio_stream->time_base);
if (pkt.duration > 0)
pkt.duration = av_rescale_q(pkt.duration, c->time_base, context->audio_stream->time_base);
pkt.stream_index = context->audio_stream->index;
pkt.flags |= AV_PKT_FLAG_KEY;
if (av_interleaved_write_frame(context->outfile, &pkt) != 0) {
fprintf(stderr, "Error writing audio packet!\n");
if (frame)
avcodec_free_frame(&frame);
return -1;
}
av_free_packet(&pkt);
}
if (frame)
avcodec_free_frame(&frame);
return 0;
}
static void *start_encoder( void *ptr )
{
obe_aud_enc_params_t *enc_params = ptr;
obe_t *h = enc_params->h;
obe_encoder_t *encoder = enc_params->encoder;
obe_output_stream_t *stream = enc_params->stream;
obe_raw_frame_t *raw_frame;
obe_coded_frame_t *coded_frame;
void *audio_buf = NULL;
int64_t cur_pts = -1, pts_increment;
int i, frame_size, ret, got_pkt, num_frames = 0, total_size = 0, audio_buf_len;
AVFifoBuffer *out_fifo = NULL;
AVAudioResampleContext *avr = NULL;
AVPacket pkt;
AVCodecContext *codec = NULL;
AVFrame *frame = NULL;
AVDictionary *opts = NULL;
char is_latm[2];
avcodec_register_all();
codec = avcodec_alloc_context3( NULL );
if( !codec )
{
fprintf( stderr, "Malloc failed\n" );
goto finish;
}
for( i = 0; lavc_encoders[i].obe_name != -1; i++ )
{
if( lavc_encoders[i].obe_name == stream->stream_format )
break;
}
if( lavc_encoders[i].obe_name == -1 )
{
fprintf( stderr, "[lavc] Could not find encoder1\n" );
goto finish;
}
AVCodec *enc = avcodec_find_encoder( lavc_encoders[i].lavc_name );
if( !enc )
{
fprintf( stderr, "[lavc] Could not find encoder2\n" );
goto finish;
}
if( enc->sample_fmts[0] == -1 )
{
fprintf( stderr, "[lavc] No valid sample formats\n" );
goto finish;
}
codec->sample_rate = enc_params->sample_rate;
codec->bit_rate = stream->bitrate * 1000;
codec->sample_fmt = enc->sample_fmts[0];
codec->channels = av_get_channel_layout_nb_channels( stream->channel_layout );
codec->channel_layout = stream->channel_layout;
codec->time_base.num = 1;
codec->time_base.den = OBE_CLOCK;
codec->profile = stream->aac_opts.aac_profile == AAC_HE_V2 ? FF_PROFILE_AAC_HE_V2 :
stream->aac_opts.aac_profile == AAC_HE_V1 ? FF_PROFILE_AAC_HE :
FF_PROFILE_AAC_LOW;
snprintf( is_latm, sizeof(is_latm), "%i", stream->aac_opts.latm_output );
av_dict_set( &opts, "latm", is_latm, 0 );
av_dict_set( &opts, "header_period", "2", 0 );
if( avcodec_open2( codec, enc, &opts ) < 0 )
{
fprintf( stderr, "[lavc] Could not open encoder\n" );
goto finish;
}
avr = avresample_alloc_context();
if( !avr )
{
fprintf( stderr, "Malloc failed\n" );
goto finish;
}
av_opt_set_int( avr, "in_channel_layout", codec->channel_layout, 0 );
av_opt_set_int( avr, "in_sample_fmt", enc_params->input_sample_format, 0 );
av_opt_set_int( avr, "in_sample_rate", enc_params->sample_rate, 0 );
av_opt_set_int( avr, "out_channel_layout", codec->channel_layout, 0 );
av_opt_set_int( avr, "out_sample_fmt", codec->sample_fmt, 0 );
av_opt_set_int( avr, "dither_method", AV_RESAMPLE_DITHER_TRIANGULAR_NS, 0 );
if( avresample_open( avr ) < 0 )
{
fprintf( stderr, "Could not open AVResample\n" );
goto finish;
}
/* The number of samples per E-AC3 frame is unknown until the encoder is ready */
if( stream->stream_format == AUDIO_E_AC_3 || stream->stream_format == AUDIO_AAC )
{
pthread_mutex_lock( &encoder->queue.mutex );
encoder->is_ready = 1;
encoder->num_samples = codec->frame_size;
/* Broadcast because input and muxer can be stuck waiting for encoder */
pthread_cond_broadcast( &encoder->queue.in_cv );
pthread_mutex_unlock( &encoder->queue.mutex );
}
frame_size = (double)codec->frame_size * 125 * stream->bitrate *
enc_params->frames_per_pes / enc_params->sample_rate;
/* NB: libfdk-aac already doubles the frame size appropriately */
pts_increment = (double)codec->frame_size * OBE_CLOCK * enc_params->frames_per_pes / enc_params->sample_rate;
out_fifo = av_fifo_alloc( frame_size );
if( !out_fifo )
{
fprintf( stderr, "Malloc failed\n" );
goto finish;
}
audio_buf_len = codec->frame_size * av_get_bytes_per_sample( codec->sample_fmt ) * codec->channels;
audio_buf = av_malloc( audio_buf_len );
if( !audio_buf )
{
fprintf( stderr, "Malloc failed\n" );
goto finish;
}
frame = avcodec_alloc_frame();
if( !frame )
{
fprintf( stderr, "Could not allocate frame\n" );
goto finish;
}
while( 1 )
{
/* TODO: detect bitrate or channel reconfig */
pthread_mutex_lock( &encoder->queue.mutex );
while( !encoder->queue.size && !encoder->cancel_thread )
pthread_cond_wait( &encoder->queue.in_cv, &encoder->queue.mutex );
if( encoder->cancel_thread )
{
pthread_mutex_unlock( &encoder->queue.mutex );
goto finish;
}
raw_frame = encoder->queue.queue[0];
pthread_mutex_unlock( &encoder->queue.mutex );
if( cur_pts == -1 )
cur_pts = raw_frame->pts;
if( avresample_convert( avr, NULL, 0, raw_frame->audio_frame.num_samples, raw_frame->audio_frame.audio_data,
raw_frame->audio_frame.linesize, raw_frame->audio_frame.num_samples ) < 0 )
{
syslog( LOG_ERR, "[lavc] Sample format conversion failed\n" );
break;
}
raw_frame->release_data( raw_frame );
raw_frame->release_frame( raw_frame );
remove_from_queue( &encoder->queue );
while( avresample_available( avr ) >= codec->frame_size )
{
got_pkt = 0;
avcodec_get_frame_defaults( frame );
frame->nb_samples = codec->frame_size;
avresample_read( avr, &audio_buf, codec->frame_size );
if( avcodec_fill_audio_frame( frame, codec->channels, codec->sample_fmt, audio_buf, audio_buf_len, 0 ) < 0 )
{
syslog( LOG_ERR, "[lavc] Could not fill audio frame\n" );
break;
}
av_init_packet( &pkt );
pkt.data = NULL;
pkt.size = 0;
ret = avcodec_encode_audio2( codec, &pkt, frame, &got_pkt );
if( ret < 0 )
{
syslog( LOG_ERR, "[lavc] Audio encoding failed\n" );
goto finish;
}
if( !got_pkt )
continue;
total_size += pkt.size;
num_frames++;
if( av_fifo_realloc2( out_fifo, av_fifo_size( out_fifo ) + pkt.size ) < 0 )
{
syslog( LOG_ERR, "Malloc failed\n" );
break;
}
av_fifo_generic_write( out_fifo, pkt.data, pkt.size, NULL );
obe_free_packet( &pkt );
if( num_frames == enc_params->frames_per_pes )
{
coded_frame = new_coded_frame( encoder->output_stream_id, total_size );
if( !coded_frame )
{
syslog( LOG_ERR, "Malloc failed\n" );
goto finish;
}
av_fifo_generic_read( out_fifo, coded_frame->data, total_size, NULL );
coded_frame->pts = cur_pts;
coded_frame->random_access = 1; /* Every frame output is a random access point */
add_to_queue( &h->mux_queue, coded_frame );
/* We need to generate PTS because frame sizes have changed */
cur_pts += pts_increment;
total_size = num_frames = 0;
}
}
}
finish:
if( frame )
avcodec_free_frame( &frame );
if( audio_buf )
av_free( audio_buf );
if( out_fifo )
av_fifo_free( out_fifo );
if( avr )
avresample_free( &avr );
if( codec )
{
avcodec_close( codec );
av_free( codec );
}
free( enc_params );
return NULL;
}
static void *
test_generator_thread(void *aux)
{
media_pipe_t *mp = aux;
media_buf_t *mb = NULL;
event_t *e;
media_queue_t *mq = &mp->mp_audio;
pcm_sound_t voices[8];
AVFrame *frame = av_frame_alloc();
AVCodec *codec = avcodec_find_encoder(AV_CODEC_ID_AC3);
AVCodecContext *ctx = avcodec_alloc_context3(codec);
ctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
ctx->sample_rate = 48000;
ctx->channel_layout = AV_CH_LAYOUT_5POINT1;
if(avcodec_open2(ctx, codec, NULL) < 0) {
TRACE(TRACE_ERROR, "audio", "Unable to open encoder");
return NULL;
}
float *genbuf[8];
for(int i = 0; i < 8; i++) {
genbuf[i] = av_malloc(ctx->frame_size * sizeof(float));
frame->data[i] = (void *)genbuf[i];
}
frame->nb_samples = ctx->frame_size;
media_codec_t *mc = media_codec_create(AV_CODEC_ID_AC3, 0,
NULL, NULL, NULL, mp);
mp_set_playstatus_by_hold(mp, 0, NULL);
mp->mp_audio.mq_stream = 0;
mp_configure(mp, 0, MP_BUFFER_NONE, 0, "testsignal");
mp_become_primary(mp);
int sample = 0;
unpack_speaker_positions(voices);
while(1) {
if(mb == NULL) {
int got_packet;
AVPacket pkt = {0};
generator_t *g;
switch(signal_type) {
case 0:
for(int c = 0; c < 8; c++) {
int z = ctx->frame_size;
if(test_channels[c]) {
int j = sample & 0xffff;
int to_copy = MIN(ctx->frame_size, (voices[c].samples - j));
if(to_copy < 0)
to_copy = 0;
for(int i = 0; i < to_copy; i++) {
genbuf[c][i] = voices[c].data[j+i] / 32767.0f;
}
z = ctx->frame_size - to_copy;
}
memset(genbuf[c] + ctx->frame_size - z, 0, sizeof(float) * z);
}
sample += ctx->frame_size;
goto encode;
default: g = &gen_pink_noise; break;
case 2: g = &gen_sinewave; break;
}
for(int i = 0; i < ctx->frame_size; i++) {
float x = g(sample);
for(int c = 0; c < 8; c++) {
genbuf[c][i] = test_channels[c] ? x : 0;
}
sample++;
}
encode:
av_init_packet(&pkt);
int r = avcodec_encode_audio2(ctx, &pkt, frame, &got_packet);
if(!r && got_packet) {
mb = media_buf_from_avpkt_unlocked(mp, &pkt);
av_free_packet(&pkt);
} else {
sleep(1);
}
mb->mb_cw = media_codec_ref(mc);
mb->mb_data_type = MB_AUDIO;
mb->mb_pts = PTS_UNSET;
}
if((e = mb_enqueue_with_events(mp, mq, mb)) == NULL) {
mb = NULL; /* Enqueue succeeded */
continue;
}
if(event_is_type(e, EVENT_EXIT)) {
mp_flush(mp, 0);
break;
}
event_release(e);
}
av_frame_free(&frame);
for(int i = 0; i < 8; i++)
av_freep(&genbuf[i]);
for(int i = 0; i < 8; i++)
free(voices[i].data);
media_codec_deref(mc);
return NULL;
}
static hb_buffer_t* Encode(hb_work_object_t *w)
{
hb_work_private_t *pv = w->private_data;
hb_audio_t *audio = w->audio;
uint64_t pts, pos;
if (hb_list_bytes(pv->list) < pv->input_samples * sizeof(float))
{
return NULL;
}
hb_list_getbytes(pv->list, pv->input_buf, pv->input_samples * sizeof(float),
&pts, &pos);
// Prepare input frame
int out_linesize;
int out_size = av_samples_get_buffer_size(&out_linesize,
pv->context->channels,
pv->samples_per_frame,
pv->context->sample_fmt, 1);
AVFrame frame = { .nb_samples = pv->samples_per_frame, };
avcodec_fill_audio_frame(&frame,
pv->context->channels, pv->context->sample_fmt,
pv->output_buf, out_size, 1);
if (pv->avresample != NULL)
{
int in_linesize;
av_samples_get_buffer_size(&in_linesize, pv->context->channels,
frame.nb_samples, AV_SAMPLE_FMT_FLT, 1);
int out_samples = avresample_convert(pv->avresample,
frame.extended_data, out_linesize,
frame.nb_samples,
&pv->input_buf, in_linesize,
frame.nb_samples);
if (out_samples != pv->samples_per_frame)
{
// we're not doing sample rate conversion, so this shouldn't happen
hb_log("encavcodecaWork: avresample_convert() failed");
return NULL;
}
}
// Libav requires that timebase of audio frames be in sample_rate units
frame.pts = pts + (90000 * pos / (sizeof(float) *
pv->out_discrete_channels *
audio->config.out.samplerate));
frame.pts = av_rescale(frame.pts, pv->context->sample_rate, 90000);
// Prepare output packet
AVPacket pkt;
int got_packet;
hb_buffer_t *out = hb_buffer_init(pv->max_output_bytes);
av_init_packet(&pkt);
pkt.data = out->data;
pkt.size = out->alloc;
// Encode
int ret = avcodec_encode_audio2(pv->context, &pkt, &frame, &got_packet);
if (ret < 0)
{
hb_log("encavcodeca: avcodec_encode_audio failed");
hb_buffer_close(&out);
return NULL;
}
if (got_packet && pkt.size)
{
out->size = pkt.size;
// The output pts from libav is in context->time_base. Convert it back
// to our timebase.
out->s.start = av_rescale_q(pkt.pts, pv->context->time_base,
(AVRational){1, 90000});
out->s.duration = (double)90000 * pv->samples_per_frame /
audio->config.out.samplerate;
out->s.stop = out->s.start + out->s.duration;
out->s.type = AUDIO_BUF;
out->s.frametype = HB_FRAME_AUDIO;
}
else
{
hb_buffer_close(&out);
return Encode(w);
}
return out;
}
static hb_buffer_t * Flush( hb_work_object_t * w )
{
hb_buffer_t *first, *buf, *last;
first = last = buf = Encode( w );
while( buf )
{
last = buf;
buf->next = Encode( w );
buf = buf->next;
}
if( last )
{
last->next = hb_buffer_init( 0 );
}
else
{
first = hb_buffer_init( 0 );
}
return first;
}
static int filter_out(struct af_instance *af)
{
af_ac3enc_t *s = af->priv;
if (!s->pending)
return 0;
AVFrame *frame = av_frame_alloc();
if (!frame) {
MP_FATAL(af, "Could not allocate memory \n");
return -1;
}
int err = -1;
AVPacket pkt = {0};
av_init_packet(&pkt);
#if HAVE_AVCODEC_NEW_CODEC_API
// Send input as long as it wants.
while (1) {
err = read_input_frame(af, frame);
if (err < 0)
goto done;
if (err == 0)
break;
err = -1;
int lavc_ret = avcodec_send_frame(s->lavc_actx, frame);
// On EAGAIN, we're supposed to read remaining output.
if (lavc_ret == AVERROR(EAGAIN))
break;
if (lavc_ret < 0) {
MP_FATAL(af, "Encode failed.\n");
goto done;
}
s->encoder_buffered += s->input->samples;
s->input->samples = 0;
}
int lavc_ret = avcodec_receive_packet(s->lavc_actx, &pkt);
if (lavc_ret == AVERROR(EAGAIN)) {
// Need to buffer more input.
err = 0;
goto done;
}
if (lavc_ret < 0) {
MP_FATAL(af, "Encode failed.\n");
goto done;
}
#else
err = read_input_frame(af, frame);
if (err < 0)
goto done;
if (err == 0)
goto done;
err = -1;
int ok;
int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &pkt, frame, &ok);
s->input->samples = 0;
if (lavc_ret < 0 || !ok) {
MP_FATAL(af, "Encode failed.\n");
goto done;
}
#endif
MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
pkt.size, s->pending->samples + s->input->samples);
s->encoder_buffered -= AC3_FRAME_SIZE;
struct mp_audio *out =
mp_audio_pool_get(af->out_pool, af->data, s->out_samples);
if (!out)
goto done;
mp_audio_copy_attributes(out, s->pending);
int frame_size = pkt.size;
int header_len = 0;
char hdr[8];
if (s->cfg_add_iec61937_header && pkt.size > 5) {
int bsmod = pkt.data[5] & 0x7;
int len = frame_size;
frame_size = AC3_FRAME_SIZE * 2 * 2;
header_len = 8;
AV_WL16(hdr, 0xF872); // iec 61937 syncword 1
AV_WL16(hdr + 2, 0x4E1F); // iec 61937 syncword 2
hdr[5] = bsmod; // bsmod
hdr[4] = 0x01; // data-type ac3
AV_WL16(hdr + 6, len << 3); // number of bits in payload
}
if (frame_size > out->samples * out->sstride)
abort();
char *buf = (char *)out->planes[0];
memcpy(buf, hdr, header_len);
memcpy(buf + header_len, pkt.data, pkt.size);
memset(buf + header_len + pkt.size, 0,
frame_size - (header_len + pkt.size));
swap_16((uint16_t *)(buf + header_len), pkt.size / 2);
out->samples = frame_size / out->sstride;
af_add_output_frame(af, out);
err = 0;
done:
av_packet_unref(&pkt);
av_frame_free(&frame);
update_delay(af);
return err;
}
int main(int argc, char* argv[])
{
AVFormatContext* pFormatCtx;
AVOutputFormat* fmt;
AVStream* audio_st;
AVCodecContext* pCodecCtx;
AVCodec* pCodec;
uint8_t* frame_buf;
AVFrame* pFrame;
AVPacket pkt;
int got_frame=0;
int ret=0;
int size=0;
FILE *in_file=NULL; //Raw PCM data
int framenum=1000; //Audio frame number
const char* out_file = "tdjm.aac"; //Output URL
int i;
in_file= fopen("tdjm.pcm", "rb");
av_register_all();
//Method 1.
pFormatCtx = avformat_alloc_context();
fmt = av_guess_format(NULL, out_file, NULL);
pFormatCtx->oformat = fmt;
//Method 2.
//avformat_alloc_output_context2(&pFormatCtx, NULL, NULL, 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;
}
audio_st = avformat_new_stream(pFormatCtx, 0);
if (audio_st==NULL){
return -1;
}
pCodecCtx = audio_st->codec;
pCodecCtx->codec_id = fmt->audio_codec;
pCodecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
pCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
pCodecCtx->sample_rate= 44100;
pCodecCtx->channel_layout=AV_CH_LAYOUT_STEREO;
pCodecCtx->channels = av_get_channel_layout_nb_channels(pCodecCtx->channel_layout);
pCodecCtx->bit_rate = 64000;
//Show some information
av_dump_format(pFormatCtx, 0, out_file, 1);
pCodec = avcodec_find_encoder(pCodecCtx->codec_id);
if (!pCodec){
printf("Can not find encoder!\n");
return -1;
}
if (avcodec_open2(pCodecCtx, pCodec,NULL) < 0){
printf("Failed to open encoder!\n");
return -1;
}
pFrame = av_frame_alloc();
pFrame->nb_samples= pCodecCtx->frame_size;
pFrame->format= pCodecCtx->sample_fmt;
size = av_samples_get_buffer_size(NULL, pCodecCtx->channels,pCodecCtx->frame_size,pCodecCtx->sample_fmt, 1);
frame_buf = (uint8_t *)av_malloc(size);
avcodec_fill_audio_frame(pFrame, pCodecCtx->channels, pCodecCtx->sample_fmt,(const uint8_t*)frame_buf, size, 1);
//Write Header
avformat_write_header(pFormatCtx,NULL);
av_new_packet(&pkt,size);
for (i=0; i<framenum; i++){
//Read PCM
if (fread(frame_buf, 1, size, in_file) <= 0){
printf("Failed to read raw data! \n");
return -1;
}else if(feof(in_file)){
break;
}
pFrame->data[0] = frame_buf; //PCM Data
pFrame->pts=i*100;
got_frame=0;
//Encode
ret = avcodec_encode_audio2(pCodecCtx, &pkt,pFrame, &got_frame);
if(ret < 0){
printf("Failed to encode!\n");
return -1;
}
if (got_frame==1){
printf("Succeed to encode 1 frame! \tsize:%5d\n",pkt.size);
pkt.stream_index = audio_st->index;
ret = av_write_frame(pFormatCtx, &pkt);
av_free_packet(&pkt);
}
}
//Flush Encoder
ret = flush_encoder(pFormatCtx,0);
if (ret < 0) {
printf("Flushing encoder failed\n");
return -1;
}
//Write Trailer
av_write_trailer(pFormatCtx);
//Clean
if (audio_st){
avcodec_close(audio_st->codec);
av_free(pFrame);
av_free(frame_buf);
}
avio_close(pFormatCtx->pb);
avformat_free_context(pFormatCtx);
fclose(in_file);
return 0;
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
FILE *f;
uint16_t *samples;
float t, tincr;
printf("Audio encoding\n");
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "encoder does not support %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
c->sample_fmt, 0);
samples = av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for(i=0;i<200;i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "error encoding audio frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
}
}
fclose(f);
av_freep(&samples);
av_frame_free(&frame);
avcodec_close(c);
av_free(c);
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
FILE *f;
uint16_t *samples;
float t, tincr;
printf("Encode audio file %s\n", filename);
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
printf("Channels:%d %d %d\n",c->channels,c->sample_rate,c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
/* frame containing input raw audio */
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
printf("FrameSize:%d\n",c->frame_size);
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
c->sample_fmt, 0);
samples = (uint16_t *)av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
exit(1);
}
int t1=0;
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for(i=0;i<200;i++)
{
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
/* encode the samples */
if(t1==0)
{
FILE *f=fopen("C:/o11.dump","w");
fwrite(samples,buffer_size,1,f);
fclose(f);
}
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
if(t1==1000000)
{
AVFrame *dframe=avcodec_alloc_frame();
avcodec_get_frame_defaults(dframe);
AVCodec *decoder=avcodec_find_decoder(AV_CODEC_ID_MP2);
AVCodecContext *decodecontext=avcodec_alloc_context3(decoder);
avcodec_open2(decodecontext, decoder,NULL);
int pic;
int ret=avcodec_decode_audio4(decodecontext,dframe,&pic,&pkt);
if(ret<0)
{
printf("Seem f**k\n");
}
printf("LINE SIZE:%d<->%d\n",dframe->linesize[0],dframe->nb_samples);
FILE *f=fopen("c:/o12.dump","w");
fwrite(dframe->data[0],dframe->linesize[0],1,f);
fclose(f);
}
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
t1++;
}
/* get the delayed frames */
fclose(f);
av_freep(&samples);
avcodec_free_frame(&frame);
avcodec_close(c);
av_free(c);
printf("OK");
}
static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
int got_packet, ret, dst_nb_samples;
av_init_packet(&pkt);
c = st->codec;
if (!flush) {
get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
/* convert samples from native format to destination codec format, using the resampler */
if (swr_ctx) {
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_samples_data[0]);
ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
dst_nb_samples, c->sample_fmt, 0);
if (ret < 0)
exit(1);
max_dst_nb_samples = dst_nb_samples;
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
c->sample_fmt, 0);
}
/* convert to destination format */
ret = swr_convert(swr_ctx,
dst_samples_data, dst_nb_samples,
(const uint8_t **)src_samples_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
} else {
dst_nb_samples = src_nb_samples;
}
audio_frame->nb_samples = dst_nb_samples;
audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
dst_samples_data[0], dst_samples_size, 0);
samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (!got_packet) {
if (flush)
audio_is_eof = 1;
return;
}
ret = write_frame(oc, &c->time_base, st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
}
static bool encode_audio(ffemu_t *handle, AVPacket *pkt, bool dry)
{
av_init_packet(pkt);
pkt->data = handle->audio.outbuf;
pkt->size = handle->audio.outbuf_size;
#ifdef HAVE_FFMPEG_AVCODEC_ENCODE_AUDIO2
AVFrame frame;
avcodec_get_frame_defaults(&frame);
frame.nb_samples = handle->audio.frames_in_buffer;
frame.pts = handle->audio.frame_cnt;
int samples_size = frame.nb_samples *
handle->audio.codec->channels *
sizeof(int16_t);
avcodec_fill_audio_frame(&frame, handle->audio.codec->channels,
handle->audio.codec->sample_fmt, (const uint8_t*)handle->audio.buffer,
samples_size, 1);
int got_packet = 0;
if (avcodec_encode_audio2(handle->audio.codec,
pkt, dry ? NULL : &frame, &got_packet) < 0)
return false;
if (!got_packet)
{
pkt->size = 0;
pkt->pts = AV_NOPTS_VALUE;
pkt->dts = AV_NOPTS_VALUE;
return true;
}
if (pkt->pts != (int64_t)AV_NOPTS_VALUE)
{
pkt->pts = av_rescale_q(pkt->pts,
handle->audio.codec->time_base,
handle->muxer.astream->time_base);
}
if (pkt->dts != (int64_t)AV_NOPTS_VALUE)
{
pkt->dts = av_rescale_q(pkt->dts,
handle->audio.codec->time_base,
handle->muxer.astream->time_base);
}
#else
if (dry)
return false;
memset(handle->audio.buffer + handle->audio.frames_in_buffer * handle->audio.codec->channels, 0,
(handle->audio.codec->frame_size - handle->audio.frames_in_buffer) *
handle->audio.codec->channels * sizeof(int16_t));
int out_size = avcodec_encode_audio(handle->audio.codec,
handle->audio.outbuf, handle->audio.outbuf_size, handle->audio.buffer);
if (out_size < 0)
return false;
if (out_size == 0)
{
pkt->size = 0;
return true;
}
pkt->size = out_size;
if (handle->audio.codec->coded_frame->pts != (int64_t)AV_NOPTS_VALUE)
{
pkt->pts = av_rescale_q(handle->audio.codec->coded_frame->pts,
handle->audio.codec->time_base,
handle->muxer.astream->time_base);
}
else
pkt->pts = AV_NOPTS_VALUE;
if (handle->audio.codec->coded_frame->key_frame)
pkt->flags |= AV_PKT_FLAG_KEY;
#endif
pkt->stream_index = handle->muxer.astream->index;
return true;
}
/**
* encodes raw audio to various encodings like mp3
* @param args it holds various parameters like desired output encoding
* @see libav_encode_args
*/
void audio_encode(libav_encode_args* a) {
libav_encode_args& args = *a;
avcodec_register_all();
AVCodec *codec;
AVCodecContext *c = NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int max_input_buffer_size = args.input_buffer.periodbuffer->size();
int input_buffer_size = args.initptr < args.termptr ? (args.termptr - args.
initptr + 1) : (args.termptr + args.input_buffer.periodbuffer->
size() - args.initptr + 1);
FILE *f;
uint16_t *samples;
float t, tincr;
// printf("Encode audio file %s\n", filename);
/* find the MP2 encoder */
codec = avcodec_find_encoder(args.codecID);
if (!codec) {
// fprintf(stderr, "Codec not found\n");
// exit(1);
ErrnoExit("Codec not found");
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = args.bitrate;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = args.av_smpl_fmt; //AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
// f = fopen(filename, "wb");
// if (!f) {
// fprintf(stderr, "Could not open %s\n", filename);
// exit(1);
// }
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
int computed_buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, c->sample_fmt, 0);
samples = (uint16_t*) av_malloc(computed_buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n", computed_buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
if (computed_buffer_size != (*args.input_buffer.periodbuffer)[(args.initptr) % max_input_buffer_size].length) {
fp_err(FPOL_LAV, "buffer statement mismatch. Encoder exited");
return;
}
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, (const uint8_t*) samples, computed_buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
// t = 0;
//tincr = 2 * M_PI * 440.0 / c->sample_rate;
args.output_buffer = (char*) malloc(input_buffer_size * (*args.input_buffer.periodbuffer)[(args.initptr) % max_input_buffer_size].length);
args.output_buffer_size = 0;
int non_cycled_termptr = args.initptr + input_buffer_size;
for (i = args.initptr; i < non_cycled_termptr; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
// for (j = 0; j < c->frame_size; j++) {
// samples[2 * j] = (int) (sin(t) * 10000);
// for (k = 1; k < c->channels; k++) {
// samples[2 * j + k] = samples[2 * j];
// }
// t += tincr;
// }
/* setup the data pointers in the AVFrame */
int k = 0;
int l = computed_buffer_size / 4;
uint16_t* obuffer = (uint16_t*) ((*args.input_buffer.periodbuffer)[i % max_input_buffer_size].period);
for (j = 0; j < computed_buffer_size / 2; j++) {
samples[k++] = obuffer[j];
j++;
samples[l++] = obuffer[j];
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
//fwrite(pkt.data, 1, pkt.size, f);
//pkt.data[pkt.size] = 0;
memcpy(args.output_buffer + args.output_buffer_size, pkt.data, pkt.size);
args.output_buffer_size += pkt.size;
av_packet_unref(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
//fwrite(pkt.data, 1, pkt.size, f);
memcpy(args.output_buffer + args.output_buffer_size, pkt.data, pkt.size);
args.output_buffer_size += pkt.size;
av_packet_unref(&pkt);
}
}
// fclose(f);
av_freep(&samples);
av_frame_free(&frame);
avcodec_close(c);
av_free(c);
}
void _ffmpegPostAudioFrame(struct mAVStream* stream, int16_t left, int16_t right) {
struct FFmpegEncoder* encoder = (struct FFmpegEncoder*) stream;
if (!encoder->context || !encoder->audioCodec) {
return;
}
if (encoder->absf && !left) {
// XXX: AVBSF doesn't like silence. Figure out why.
left = 1;
}
encoder->audioBuffer[encoder->currentAudioSample * 2] = left;
encoder->audioBuffer[encoder->currentAudioSample * 2 + 1] = right;
++encoder->currentAudioSample;
if (encoder->currentAudioSample * 4 < encoder->audioBufferSize) {
return;
}
int channelSize = 2 * av_get_bytes_per_sample(encoder->audio->sample_fmt);
encoder->currentAudioSample = 0;
#ifdef USE_LIBAVRESAMPLE
avresample_convert(encoder->resampleContext, 0, 0, 0,
(uint8_t**) &encoder->audioBuffer, 0, encoder->audioBufferSize / 4);
if (avresample_available(encoder->resampleContext) < encoder->audioFrame->nb_samples) {
return;
}
#if LIBAVCODEC_VERSION_MAJOR >= 55
av_frame_make_writable(encoder->audioFrame);
#endif
int samples = avresample_read(encoder->resampleContext, encoder->audioFrame->data, encoder->postaudioBufferSize / channelSize);
#else
#if LIBAVCODEC_VERSION_MAJOR >= 55
av_frame_make_writable(encoder->audioFrame);
#endif
if (swr_get_out_samples(encoder->resampleContext, encoder->audioBufferSize / 4) < encoder->audioFrame->nb_samples) {
swr_convert(encoder->resampleContext, NULL, 0, (const uint8_t**) &encoder->audioBuffer, encoder->audioBufferSize / 4);
return;
}
int samples = swr_convert(encoder->resampleContext, encoder->audioFrame->data, encoder->postaudioBufferSize / channelSize,
(const uint8_t**) &encoder->audioBuffer, encoder->audioBufferSize / 4);
#endif
encoder->audioFrame->pts = av_rescale_q(encoder->currentAudioFrame, encoder->audio->time_base, encoder->audioStream->time_base);
encoder->currentAudioFrame += samples;
AVPacket packet;
av_init_packet(&packet);
packet.data = 0;
packet.size = 0;
packet.pts = encoder->audioFrame->pts;
int gotData;
#ifdef FFMPEG_USE_PACKETS
avcodec_send_frame(encoder->audio, encoder->audioFrame);
gotData = avcodec_receive_packet(encoder->audio, &packet);
gotData = (gotData == 0) && packet.size;
#else
avcodec_encode_audio2(encoder->audio, &packet, encoder->audioFrame, &gotData);
#endif
if (gotData) {
if (encoder->absf) {
AVPacket tempPacket;
#ifdef FFMPEG_USE_NEW_BSF
int success = av_bsf_send_packet(encoder->absf, &packet);
if (success >= 0) {
success = av_bsf_receive_packet(encoder->absf, &tempPacket);
}
#else
int success = av_bitstream_filter_filter(encoder->absf, encoder->audio, 0,
&tempPacket.data, &tempPacket.size,
packet.data, packet.size, 0);
#endif
if (success >= 0) {
#if LIBAVUTIL_VERSION_MAJOR >= 53
tempPacket.buf = av_buffer_create(tempPacket.data, tempPacket.size, av_buffer_default_free, 0, 0);
#endif
#ifdef FFMPEG_USE_PACKET_UNREF
av_packet_move_ref(&packet, &tempPacket);
#else
av_free_packet(&packet);
packet = tempPacket;
#endif
packet.stream_index = encoder->audioStream->index;
av_interleaved_write_frame(encoder->context, &packet);
}
} else {
packet.stream_index = encoder->audioStream->index;
av_interleaved_write_frame(encoder->context, &packet);
}
}
#ifdef FFMPEG_USE_PACKET_UNREF
av_packet_unref(&packet);
#else
av_free_packet(&packet);
#endif
}
int dc_audio_encoder_encode(AudioOutputFile *audio_output_file, AudioInputData *audio_input_data)
{
int got_pkt;
AVCodecContext *audio_codec_ctx = audio_output_file->codec_ctx;
while (av_fifo_size(audio_output_file->fifo) >= audio_output_file->frame_bytes) {
#ifdef DC_AUDIO_RESAMPLER
uint8_t **data; //mirror AVFrame::data
int num_planes_out;
#endif
Bool resample;
av_fifo_generic_read(audio_output_file->fifo, audio_output_file->adata_buf, audio_output_file->frame_bytes, NULL);
audio_output_file->aframe->data[0] = audio_output_file->adata_buf;
audio_output_file->aframe->linesize[0] = audio_output_file->frame_bytes;
audio_output_file->aframe->linesize[1] = 0;
av_init_packet(&audio_output_file->packet);
audio_output_file->packet.data = NULL;
audio_output_file->packet.size = 0;
/*
* Set PTS (method 1)
*/
//audio_output_file->aframe->pts = audio_input_data->next_pts;
/*
* Set PTS (method 2)
*/
//{
// int64_t now = av_gettime();
// AVRational avr;
// avr.num = 1;
// avr.den = AV_TIME_BASE;
// audio_output_file->aframe->pts = av_rescale_q(now, avr, audio_codec_ctx->time_base);
//}
resample = (DC_AUDIO_SAMPLE_FORMAT != audio_codec_ctx->sample_fmt
|| DC_AUDIO_SAMPLE_RATE != audio_codec_ctx->sample_rate
|| DC_AUDIO_NUM_CHANNELS != audio_codec_ctx->channels
|| DC_AUDIO_CHANNEL_LAYOUT != audio_codec_ctx->channel_layout);
/* Resample if needed */
if (resample) {
#ifndef DC_AUDIO_RESAMPLER
GF_LOG(GF_LOG_ERROR, GF_LOG_DASH, ("Audio resampling is needed at the encoding stage, but not supported by your version of DashCast. Aborting.\n"));
exit(1);
#else
if (ensure_resampler(audio_output_file, audio_codec_ctx)) {
return -1;
}
data = audio_output_file->aframe->extended_data;
if (resample_audio(audio_output_file, audio_codec_ctx, &num_planes_out)) {
return -1;
}
#endif
}
/* Encode audio */
if (avcodec_encode_audio2(audio_codec_ctx, &audio_output_file->packet, audio_output_file->aframe, &got_pkt) != 0) {
GF_LOG(GF_LOG_ERROR, GF_LOG_DASH, ("Error while encoding audio.\n"));
#ifdef DC_AUDIO_RESAMPLER
if (resample) {
int i;
for (i=0; i<num_planes_out; ++i) {
av_free(audio_output_file->aframe->extended_data[i]);
}
av_free(audio_output_file->aframe->extended_data);
audio_output_file->aframe->extended_data = data;
}
#endif
return -1;
}
#ifdef DC_AUDIO_RESAMPLER
if (resample) {
int i;
for (i=0; i<num_planes_out; ++i) {
av_free(audio_output_file->aframe->extended_data[i]);
}
av_free(audio_output_file->aframe->extended_data);
audio_output_file->aframe->extended_data = data;
}
#endif
if (got_pkt) {
//audio_output_file->acc_samples += audio_output_file->aframe->nb_samples;
return 0;
}
av_free_packet(&audio_output_file->packet);
}
return 1;
}
static int filter_out(struct af_instance *af)
{
af_ac3enc_t *s = af->priv;
if (!fill_buffer(af))
return 0; // need more input
AVFrame *frame = av_frame_alloc();
if (!frame) {
MP_FATAL(af, "Could not allocate memory \n");
return -1;
}
frame->nb_samples = s->in_samples;
frame->format = s->lavc_actx->sample_fmt;
frame->channel_layout = s->lavc_actx->channel_layout;
assert(s->input->num_planes <= AV_NUM_DATA_POINTERS);
frame->extended_data = frame->data;
for (int n = 0; n < s->input->num_planes; n++)
frame->data[n] = s->input->planes[n];
frame->linesize[0] = s->input->samples * s->input->sstride;
int ok;
int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &s->pkt, frame, &ok);
av_frame_free(&frame);
s->input->samples = 0;
if (lavc_ret < 0 || !ok) {
MP_FATAL(af, "Encode failed.\n");
return -1;
}
MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
s->pkt.size, s->pending->samples);
struct mp_audio *out =
mp_audio_pool_get(af->out_pool, af->data, s->out_samples);
if (!out)
return -1;
mp_audio_copy_attributes(out, s->pending);
int frame_size = s->pkt.size;
int header_len = 0;
char hdr[8];
if (s->cfg_add_iec61937_header && s->pkt.size > 5) {
int bsmod = s->pkt.data[5] & 0x7;
int len = frame_size;
frame_size = AC3_FRAME_SIZE * 2 * 2;
header_len = 8;
AV_WL16(hdr, 0xF872); // iec 61937 syncword 1
AV_WL16(hdr + 2, 0x4E1F); // iec 61937 syncword 2
hdr[5] = bsmod; // bsmod
hdr[4] = 0x01; // data-type ac3
AV_WL16(hdr + 6, len << 3); // number of bits in payload
}
if (frame_size > out->samples * out->sstride)
abort();
char *buf = (char *)out->planes[0];
memcpy(buf, hdr, header_len);
memcpy(buf + header_len, s->pkt.data, s->pkt.size);
memset(buf + header_len + s->pkt.size, 0,
frame_size - (header_len + s->pkt.size));
swap_16((uint16_t *)(buf + header_len), s->pkt.size / 2);
out->samples = frame_size / out->sstride;
af_add_output_frame(af, out);
update_delay(af);
return 0;
}
int AVFormatWriter::WriteAudioFrame(unsigned char *buf, int fnum, long long &timecode)
{
#if HAVE_BIGENDIAN
bswap_16_buf((short int*) buf, m_audioFrameSize, m_audioChannels);
#endif
int got_packet = 0;
int ret = 0;
av_init_packet(m_audPkt);
if (m_audioStream->codec->sample_fmt == AV_SAMPLE_FMT_FLT)
{
AudioOutputUtil::toFloat(FORMAT_S16, (void *)m_audioFltBuf, (void *)buf,
m_audioFrameSize * 2 * m_audioChannels);
m_audPicture->data[0] = (unsigned char *)m_audioFltBuf;
}
else
{
m_audPicture->data[0] = buf;
}
m_audPkt->data = m_audioOutBuf;
m_audPkt->size = m_audioOutBufSize;
m_audPicture->linesize[0] = m_audioFrameSize;
m_audPicture->nb_samples = m_audioFrameSize;
m_audPicture->format = m_audioStream->codec->sample_fmt;
m_audPicture->extended_data = m_audPicture->data;
m_bufferedAudioFrameTimes.push_back(timecode);
{
QMutexLocker locker(avcodeclock);
ret = avcodec_encode_audio2(m_audioStream->codec, m_audPkt,
m_audPicture, &got_packet);
}
if (ret < 0)
{
LOG(VB_RECORD, LOG_ERR, "avcodec_encode_audio2() failed");
return ret;
}
if (!got_packet)
{
//LOG(VB_RECORD, LOG_ERR, QString("WriteAudioFrame(): Frame Buffered: cs: %1, mfw: %2, f->tc: %3, fn: %4").arg(m_audPkt->size).arg(m_framesWritten).arg(timecode).arg(fnum));
return ret;
}
long long tc = timecode;
if (m_bufferedAudioFrameTimes.size())
tc = m_bufferedAudioFrameTimes.takeFirst();
if (m_startingTimecodeOffset == -1)
m_startingTimecodeOffset = tc - 1;
tc -= m_startingTimecodeOffset;
if (m_avVideoCodec)
m_audPkt->pts = tc * m_videoStream->time_base.den / m_videoStream->time_base.num / 1000;
else
m_audPkt->pts = tc * m_audioStream->time_base.den / m_audioStream->time_base.num / 1000;
m_audPkt->dts = AV_NOPTS_VALUE;
m_audPkt->flags |= AV_PKT_FLAG_KEY;
m_audPkt->stream_index = m_audioStream->index;
//LOG(VB_RECORD, LOG_ERR, QString("WriteAudioFrame(): cs: %1, mfw: %2, pkt->pts: %3, tc: %4, fn: %5, f->tc: %6").arg(m_audPkt->size).arg(m_framesWritten).arg(m_audPkt->pts).arg(tc).arg(fnum).arg(timecode));
ret = av_interleaved_write_frame(m_ctx, m_audPkt);
if (ret != 0)
LOG(VB_RECORD, LOG_ERR, LOC + "WriteAudioFrame(): "
"av_interleaved_write_frame couldn't write Audio");
timecode = tc + m_startingTimecodeOffset;
return 1;
}
void
write_audio_frame (GeglProperties *o, AVFormatContext * oc, AVStream * st)
{
Priv *p = (Priv*)o->user_data;
AVCodecContext *c = st->codec;
int sample_count = 100000;
static AVPacket pkt = { 0 };
if (pkt.size == 0)
{
av_init_packet (&pkt);
}
/* first we add incoming frames audio samples */
{
int i;
int sample_count = gegl_audio_fragment_get_sample_count (o->audio);
GeglAudioFragment *af = gegl_audio_fragment_new (gegl_audio_fragment_get_sample_rate (o->audio),
gegl_audio_fragment_get_channels (o->audio),
gegl_audio_fragment_get_channel_layout (o->audio),
sample_count);
gegl_audio_fragment_set_sample_count (af, sample_count);
for (i = 0; i < sample_count; i++)
{
af->data[0][i] = o->audio->data[0][i];
af->data[1][i] = o->audio->data[1][i];
}
gegl_audio_fragment_set_pos (af, p->audio_pos);
p->audio_pos += sample_count;
p->audio_track = g_list_append (p->audio_track, af);
}
if (!(c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE))
sample_count = c->frame_size;
/* then we encode as much as we can in a loop using the codec frame size */
while (p->audio_pos - p->audio_read_pos > sample_count)
{
long i;
int ret;
int got_packet = 0;
AVFrame *frame = alloc_audio_frame (c->sample_fmt, c->channel_layout,
c->sample_rate, sample_count);
switch (c->sample_fmt) {
case AV_SAMPLE_FMT_FLT:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((float*)frame->data[0])[c->channels*i+0] = left;
((float*)frame->data[0])[c->channels*i+1] = right;
}
break;
case AV_SAMPLE_FMT_FLTP:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((float*)frame->data[0])[i] = left;
((float*)frame->data[1])[i] = right;
}
break;
case AV_SAMPLE_FMT_S16:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((int16_t*)frame->data[0])[c->channels*i+0] = left * (1<<15);
((int16_t*)frame->data[0])[c->channels*i+1] = right * (1<<15);
}
break;
case AV_SAMPLE_FMT_S32:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((int32_t*)frame->data[0])[c->channels*i+0] = left * (1<<31);
((int32_t*)frame->data[0])[c->channels*i+1] = right * (1<<31);
}
break;
case AV_SAMPLE_FMT_S32P:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((int32_t*)frame->data[0])[i] = left * (1<<31);
((int32_t*)frame->data[1])[i] = right * (1<<31);
}
break;
case AV_SAMPLE_FMT_S16P:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((int16_t*)frame->data[0])[i] = left * (1<<15);
((int16_t*)frame->data[1])[i] = right * (1<<15);
}
break;
default:
fprintf (stderr, "eeeek unhandled audio format\n");
break;
}
frame->pts = p->next_apts;
p->next_apts += sample_count;
av_frame_make_writable (frame);
ret = avcodec_encode_audio2 (c, &pkt, frame, &got_packet);
av_packet_rescale_ts (&pkt, st->codec->time_base, st->time_base);
if (ret < 0) {
fprintf (stderr, "Error encoding audio frame: %s\n", av_err2str (ret));
}
if (got_packet)
{
pkt.stream_index = st->index;
av_interleaved_write_frame (oc, &pkt);
av_free_packet (&pkt);
}
av_frame_free (&frame);
p->audio_read_pos += sample_count;
}
}
// must get exactly ac->aframesize amount of data
static int encode(struct ao *ao, double apts, void **data)
{
AVPacket packet;
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
double realapts = ac->aframecount * (double) ac->aframesize /
ao->samplerate;
int status, gotpacket;
ac->aframecount++;
if (data)
ectx->audio_pts_offset = realapts - apts;
av_init_packet(&packet);
packet.data = ac->buffer;
packet.size = ac->buffer_size;
if(data) {
AVFrame *frame = av_frame_alloc();
frame->format = af_to_avformat(ao->format);
frame->nb_samples = ac->aframesize;
assert(ao->channels.num <= AV_NUM_DATA_POINTERS);
for (int n = 0; n < ao->channels.num; n++)
frame->extended_data[n] = data[n];
frame->linesize[0] = frame->nb_samples * ao->sstride;
if (ectx->options->rawts || ectx->options->copyts) {
// real audio pts
frame->pts = floor(apts * ac->stream->codec->time_base.den / ac->stream->codec->time_base.num + 0.5);
} else {
// audio playback time
frame->pts = floor(realapts * ac->stream->codec->time_base.den / ac->stream->codec->time_base.num + 0.5);
}
int64_t frame_pts = av_rescale_q(frame->pts, ac->stream->codec->time_base, ac->worst_time_base);
if (ac->lastpts != MP_NOPTS_VALUE && frame_pts <= ac->lastpts) {
// this indicates broken video
// (video pts failing to increase fast enough to match audio)
MP_WARN(ao, "audio frame pts went backwards (%d <- %d), autofixed\n",
(int)frame->pts, (int)ac->lastpts);
frame_pts = ac->lastpts + 1;
frame->pts = av_rescale_q(frame_pts, ac->worst_time_base, ac->stream->codec->time_base);
}
ac->lastpts = frame_pts;
frame->quality = ac->stream->codec->global_quality;
status = avcodec_encode_audio2(ac->stream->codec, &packet, frame, &gotpacket);
if (!status) {
if (ac->savepts == MP_NOPTS_VALUE)
ac->savepts = frame->pts;
}
av_frame_free(&frame);
}
else
{
status = avcodec_encode_audio2(ac->stream->codec, &packet, NULL, &gotpacket);
}
if(status) {
MP_ERR(ao, "error encoding\n");
return -1;
}
if(!gotpacket)
return 0;
MP_DBG(ao, "got pts %f (playback time: %f); out size: %d\n",
apts, realapts, packet.size);
encode_lavc_write_stats(ao->encode_lavc_ctx, ac->stream);
packet.stream_index = ac->stream->index;
// Do we need this at all? Better be safe than sorry...
if (packet.pts == AV_NOPTS_VALUE) {
MP_WARN(ao, "encoder lost pts, why?\n");
if (ac->savepts != MP_NOPTS_VALUE)
packet.pts = ac->savepts;
}
if (packet.pts != AV_NOPTS_VALUE)
packet.pts = av_rescale_q(packet.pts, ac->stream->codec->time_base,
ac->stream->time_base);
if (packet.dts != AV_NOPTS_VALUE)
packet.dts = av_rescale_q(packet.dts, ac->stream->codec->time_base,
ac->stream->time_base);
if(packet.duration > 0)
packet.duration = av_rescale_q(packet.duration, ac->stream->codec->time_base,
ac->stream->time_base);
ac->savepts = MP_NOPTS_VALUE;
if (encode_lavc_write_frame(ao->encode_lavc_ctx, &packet) < 0) {
MP_ERR(ao, "error writing at %f %f/%f\n",
realapts, (double) ac->stream->time_base.num,
(double) ac->stream->time_base.den);
return -1;
}
return packet.size;
}
