webrtc aec 用法2

 

 

CEchoCanceller3::CEchoCanceller3(int ch, int sample_rate, int framelen)
{
	m_ch = ch;
	m_sample_rate = sample_rate;
	m_bytes_per_frame = framelen;
	m_aec_config.filter.export_linear_aec_output = true;
	m_aec_factory = std::make_unique<webrtc::EchoCanceller3Factory>(m_aec_config);
	m_echo_controler = m_aec_factory->Create(m_sample_rate, m_ch, m_ch);
	m_hp_filter = std::make_unique<webrtc::HighPassFilter>(m_sample_rate, ch);

	m_config = webrtc::StreamConfig(m_sample_rate, m_ch, false);

	m_ref_audio = std::make_unique<webrtc::AudioBuffer>(
		m_config.sample_rate_hz(), m_config.num_channels(),
		m_config.sample_rate_hz(), m_config.num_channels(),
		m_config.sample_rate_hz(), m_config.num_channels());
	m_aec_audio = std::make_unique<webrtc::AudioBuffer>(
		m_config.sample_rate_hz(), m_config.num_channels(),
		m_config.sample_rate_hz(), m_config.num_channels(),
		m_config.sample_rate_hz(), m_config.num_channels());
	int kLinearOutputRateHz = m_sample_rate;
	m_aec_linear_audio = std::make_unique<webrtc::AudioBuffer>(
		kLinearOutputRateHz, m_config.num_channels(),
		kLinearOutputRateHz, m_config.num_channels(),
		kLinearOutputRateHz, m_config.num_channels());

	//m_samples_per_frame = m_sample_rate /100;
	//m_bytes_per_frame = m_samples_per_frame * 16 / 8;
	m_samples_per_frame = m_bytes_per_frame / 2;

}
CEchoCanceller3::~CEchoCanceller3()
{

}
CObjPtr<CObjNetIOBuffer> CEchoCanceller3::Process(CObjNetIOBuffer *mic, CObjNetIOBuffer *ref)
{
//	if (ref == NULL)
	{
		return CObjPtr<CObjNetIOBuffer>(mic);
	}
	if (mic->m_nDataLen != m_bytes_per_frame || mic->m_nDataLen != ref->m_nDataLen)
	{
		return CObjPtr<CObjNetIOBuffer>(mic);
	}
	CObjPtr<CObjNetIOBuffer> out(CObjNetIOBufferSharedMemory::CreateObj(NULL, mic->m_nBufLen), TRUE);
	unsigned char* ref_tmp = (unsigned char*)ref->m_pBuf;
	unsigned char* aec_tmp = (unsigned char*)mic->m_pBuf;

	webrtc::AudioFrame ref_frame, aec_frame;
	ref_frame.UpdateFrame(0, reinterpret_cast<int16_t*>(ref_tmp), m_samples_per_frame, m_sample_rate, webrtc::AudioFrame::kNormalSpeech, webrtc::AudioFrame::kVadActive, 1);
	aec_frame.UpdateFrame(0, reinterpret_cast<int16_t*>(aec_tmp), m_samples_per_frame, m_sample_rate, webrtc::AudioFrame::kNormalSpeech, webrtc::AudioFrame::kVadActive, 1);

	m_ref_audio->CopyFrom(&ref_frame);
	m_aec_audio->CopyFrom(&aec_frame);


	m_ref_audio->SplitIntoFrequencyBands();
	m_echo_controler->AnalyzeRender(m_ref_audio.get());
	m_ref_audio->MergeFrequencyBands();
	m_echo_controler->AnalyzeCapture(m_aec_audio.get());
	m_aec_audio->SplitIntoFrequencyBands();
	m_hp_filter->Process(m_aec_audio.get(), true);
	m_echo_controler->SetAudioBufferDelay(0);
	m_echo_controler->ProcessCapture(m_aec_audio.get(), m_aec_linear_audio.get(), false);
	m_aec_audio->MergeFrequencyBands();


	//aec_audio->CopyTo(&aec_frame);
	//memcpy(aec_tmp, aec_frame.data(), bytes_per_frame);
	//wav_write_data(h_out, aec_tmp, bytes_per_frame);

	aec_frame.UpdateFrame(0, nullptr, m_samples_per_frame, m_sample_rate, webrtc::AudioFrame::kNormalSpeech, webrtc::AudioFrame::kVadActive, 1);
	m_aec_linear_audio->CopyTo(&aec_frame);
	memcpy(out->m_pBuf, aec_frame.data(), m_bytes_per_frame);
	out->m_nDataLen = m_bytes_per_frame;

	return out;

}