static int WriteAudioFrame(AVFormatContext *oc, OutputStream *ost, AVI6 *avi)
{
AVCodecContext *c = NULL;
AVPacket pkt = { 0 };
AVFrame *frame = NULL;
int ret = 0;
int got_packet = 0;
int dst_nb_samples = 0;
av_init_packet(&pkt);
c = ost->st->codec;
frame = GetAudioFrame(ost, avi);
if (frame) {
// フォーマット変換後のサンプル数を決定
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, frame->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
//av_assert0(dst_nb_samples == frame->nb_samples);
// フレームを書き込み可能にする
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
// 音声フォーマットを変換
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
return 0;
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", MakeErrorString(ret));
return 0;
}
if (got_packet) {
ret = WriteFrame(oc, &c->time_base, ost->st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
MakeErrorString(ret));
return 0;
}
}
}
return (frame || got_packet) ? 0 : 1;
}
void User::AddToPlaybackBuffer(MumbleVoip::PCMAudioFrame* frame)
{
received_voice_packet_count_++;
// Buffer overflow handling: We drop the oldest packet in the buffer
if (PlaybackBufferLengthMs() > playback_buffer_max_length_ms )
{
MumbleVoip::PCMAudioFrame* frame = GetAudioFrame();
SAFE_DELETE(frame);
}
playback_queue_.push_back(frame);
last_audio_frame_time_.restart();
if (!speaking_)
{
speaking_ = true;
emit StartReceivingAudio();
}
}
int main(int argc, char **argv)
{
int ret = -1;
uint8_t *data = NULL;
unsigned long size = 0;
int times = 0;
int width = 0;
int height = 0;
char *dest = NULL;
unsigned long dest_size = 0;
long long pts = 0;
long long dts = 0;
AUDIOPACKET ap[30] = {0};
int ap_len = 0;
int i = 0;
CAPTURECONFIG captureConfig;
PCAPTURECONFIG pCaptureConfig = &captureConfig;
ENCODECONFIG encodeConfig;
PENCODECONFIG pEncodeConfig = &encodeConfig;
PENCODER pEncoder;
PCAPTURE pCapture;
DWORD start_time, end_time;
pCaptureConfig->fps = 5;
pCaptureConfig->channels = 2;
pCaptureConfig->bits_per_sample = 16;
pCaptureConfig->samples_per_sec = 48000;
pCaptureConfig->avg_bytes_per_sec = 48000;
pEncodeConfig->fps = 5;
pEncodeConfig->width = 1366;
pEncodeConfig->height = 768;
pEncodeConfig->bit_rate = 400000;
pEncodeConfig->channels = 2;
pEncodeConfig->bits_per_sample = 16;
pEncodeConfig->sample_rate = 48000;
pEncodeConfig->avg_bytes_per_sec = 48000;
pEncodeConfig->record = 1;
memcpy(pEncodeConfig->record_file, "D:\\desktop_live.mp4", 20);
InitLog(LOG_DEBUG, OUT_FILE);
pCapture = InitCapture(pCaptureConfig);
if (NULL == pCapture)
{
printf("init capture failed\n");
return -1;
}
pEncoder = InitEncoder(pEncodeConfig);
if (NULL == pEncoder)
{
printf("init encoder failed\n");
return -1;
}
ret = StartCapture(pCapture);
if (SECCESS != ret)
{
printf("start capture failed\n");
return -1;
}
start_time = end_time = timeGetTime();
while(10*1000 > (end_time - start_time))
{
if (SECCESS == GetVideoFrame(pCapture, &data, &size, &width, &height))
{
ret = EncodeVideo(pEncoder, data, width, height, &dest, &dest_size, &pts, &dts);
if (ret == SECCESS)
{
free(dest);
}
times++;
printf("video data size = %d\n", size);
free(data);
}
if (SECCESS == GetAudioFrame(pCapture, &data, &size))
{
ap_len = 0;
ret = EncodeAudio(pEncoder, data, size, ap, &ap_len);
if (ret == SECCESS)
{
for (i=0; i<ap_len; i++)
{
free(ap[i].data);
}
}
printf("audio data size = %d\n", size);
free(data);
}
end_time = timeGetTime();
}
StopCapture(pCapture);
FreeCapture(pCapture);
FflushEncoder(pEncoder);
FreeEncoder(pEncoder);
FreeLog();
_CrtDumpMemoryLeaks();
return 0;
}
bool AudioMixerImpl::GetMixerAudio( webrtc::AudioFrame* audioFrame )
{
if (m_participants.empty())
{
return false;
}
// 分类
bool bMix = false;
for ( auto& v : m_participants )
{
auto participant = v.participant;
auto audioFrame = PopAudioFrame();
bool ret = participant->GetAudioFrame( audioFrame );
if ( !ret )
{
v.needMixCount = 0;
v.weightFactor = 0.0f;
v.audioFrame = nullptr;
v.energy = 0;
v.isMixed = false;
PushAudioFrame( audioFrame );
continue;
}
bMix = true;
v.energy = CalculateEnergy( audioFrame );
v.isSilent = audioFrame->vad_activity_ != webrtc::AudioFrame::kVadActive;
v.audioFrame = audioFrame;
}
if (!bMix)
{
return false;
}
if (m_participants.size() > (uint32_t)m_limitCount)
{
m_participants.sort( [] ( MixerParticipantInfo& lhs, MixerParticipantInfo& rhs )
{
return (lhs.energy > rhs.energy);
} );
}
MixerParticipantList mixerlist;
int nMixframeCounter = 0;
for ( auto it = m_participants.begin(); it != m_participants.end(); ++it )
{
if ( m_limitCount >= nMixframeCounter )
{
if (it->energy > 0)
{
nMixframeCounter++;
it->needMixCount = 5;
it->isMixed = true;
// 这里系数的调整至关重要，目前先简单的处理下
if (it->isSilent)
{
it->weightFactor = 0.7f;
}
else
{
it->weightFactor = 1.0f;
}
mixerlist.push_back(*it);
}
}
else
{
if (it->isMixed)
{
if ( it->needMixCount-- > 0 )
it->weightFactor /= 2; // 渐渐淡出
mixerlist.push_back( *it );
}
}
}
MixFrameList( mixerlist, audioFrame );
for ( auto&v : m_participants )
{
if ( v.audioFrame )
PushAudioFrame( v.audioFrame );
}
return true;
}
