webrtc_module_audio_mixer_webrtc audiomix-CSDN博客

本文链接：https://blog.csdn.net/u013692429/article/details/102820938

原因：由于需要将webrtc_module_audio_mixer模块进行封装,故在此进行调用分析

概况：通过内部代码可以看出webrtc_module_audio_mixer内部实现为首先将pcm进行halfgain处理,然后进行pcm对应累加，通过LimitMixedAudio将mixed的数据进行平滑处理然后进行简单的pcm扩大2倍.重点为Audioprocessing中的processStream处理.

伪代码如下：

首先需要实现Source类，每次在调用Mix时内部会调用GetAudioFrameWithInfo方法，所以在该方法内部需要进行AudioFrame的数据填充.Ssrc方法为Mix内部进行区分混音成员的标识，故不同的混音成员需要返回的该值也不一样。

class AudioMixer : public rtc::RefCountInterface {
 public:
  // A callback class that all mixer participants must inherit from/implement.
  class Source {
   public:
    enum class AudioFrameInfo {
      kNormal,  // The samples in audio_frame are valid and should be used.
      kMuted,   // The samples in audio_frame should not be used, but
                // should be implicitly interpreted as zero. Other
                // fields in audio_frame may be read and should
                // contain meaningful values.
      kError,   // The audio_frame will not be used.
    };

    // Overwrites |audio_frame|. The data_ field is overwritten with
    // 10 ms of new audio (either 1 or 2 interleaved channels) at
    // |sample_rate_hz|. All fields in |audio_frame| must be updated.
    virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
                                                 AudioFrame* audio_frame) = 0;

    // A way for a mixer implementation to distinguish participants.
    virtual int Ssrc() const = 0;

    // A way for this source to say that GetAudioFrameWithInfo called
    // with this sample rate or higher will not cause quality loss.
    virtual int PreferredSampleRate() const = 0;

    virtual ~Source() {}
  };

实现伪代码如下：可以看出在GetAudioFrameWithInfo中进行数据填充.

class AudioSrc : public webrtc::AudioMixer::Source
{
public:

	virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
		webrtc::AudioFrame* audio_frame) {
		
		printf("GetAudioFrameWithInfo %d\n",sample_rate_hz);

		audio_frame->CopyFrom(m_audioFrame);

		return AudioFrameInfo::kNormal;
	}

	virtual int Ssrc() const
	{
		return m_isrc;
	}

	virtual int PreferredSampleRate() const
	{
		return DEF_SAMPLE_RATE;
	}
	void fillaudioframe(short *pdata,int iSample)
	{
		m_audioFrame.samples_per_channel_ = DEF_SAMPLE_RATE / 100;
		m_audioFrame.sample_rate_hz_ = DEF_SAMPLE_RATE;
		m_audioFrame.id_ = 1;
		m_audioFrame.num_channels_ = 2;

		memcpy(m_audioFrame.data_, pdata, iSample * sizeof(short));

		m_audioFrame.vad_activity_ = webrtc::AudioFrame::kVadActive;
		m_audioFrame.speech_type_ = webrtc::AudioFrame::kNormalSpeech;
	}
	webrtc::AudioFrame m_audioFrame;
	int m_isrc;
};

内部每次的填充数据为10ms的sample.通过波形图可以看出多路混音后不会进行衰减。