❝Webrtc的混音主要由AudioMixer实现，可混合多条音频流。❞
1. AudioMixer接口类
代码语言：javascript复制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() {}
};
// Returns true if adding was successful. A source is never added
// twice. Addition and removal can happen on different threads.
virtual bool AddSource(Source* audio_source) = 0;
// Removal is never attempted if a source has not been successfully
// added to the mixer.
virtual void RemoveSource(Source* audio_source) = 0;
// Performs mixing by asking registered audio sources for audio. The
// mixed result is placed in the provided AudioFrame. This method
// will only be called from a single thread. The channels argument
// specifies the number of channels of the mix result. The mixer
// should mix at a rate that doesn't cause quality loss of the
// sources' audio. The mixing rate is one of the rates listed in
// AudioProcessing::NativeRate. All fields in
// |audio_frame_for_mixing| must be updated.
virtual void Mix(size_t number_of_channels,
AudioFrame* audio_frame_for_mixing) = 0;
protected:
// Since the mixer is reference counted, the destructor may be
// called from any thread.
~AudioMixer() override {}
};
2. AudioMixer实现类
代码语言：javascript复制class AudioMixerImpl : public AudioMixer {
public:
...
static rtc::scoped_refptr<AudioMixerImpl> Create();
...
// AudioMixer functions
bool AddSource(Source* audio_source) override;
void RemoveSource(Source* audio_source) override;
void Mix(size_t number_of_channels,
AudioFrame* audio_frame_for_mixing) override
RTC_LOCKS_EXCLUDED(crit_);
...
};
3. 混音实现
通过使用AddSource接口添加不同的音频流，然后通过调用Mix接口进行混音操作，其中AudioFrame* audio_frame_for_mixing是混音数据。
在AddSource添加的流(AudioMixer::Source)中，我们还要分别实现以下接口：
// audio_frame必须在其实现中更新，用于AudioMixer的Mix接口回调。
virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
AudioFrame* audio_frame) = 0;
//混频器实现区分参与者混音的一种方法。
virtual int Ssrc() const = 0;
// 表示以此采样率或更高的采样率调用GetAudioFrameWithInfo的方式不会导致音质下降。
virtual int PreferredSampleRate() const = 0;
将两个单声道16k的PCM文件混音简略代码：
代码语言：javascript复制// 创建AudioMixer::Source实现类
class Source1 : public AudioMixer::Source
{
public:
AudioFrameInfo GetAudioFrameWithInfo(int, AudioFrame *audio_frame)
{
// 读取音频文件并填充数据
audio_frame->UpdateFrame(0,
fileData, // 音频文件数据
16000/100,
16000,
AudioFrame::SpeechType::kNormalSpeech,
AudioFrame::VADActivity::kVadUnknown,
1);
return AudioFrameInfo::kNormal;
}
int Ssrc() const
{
return -1;
}
int PreferredSampleRate() const
{
return 16000;
}
};
// 创建混音实例
audioMixer = AudioMixerImpl::Create();
audioMixer->AddSource(new Source1);
audioMixer->AddDource(new Source2);
// 混音操作
while (true) {
AudioFrame audioFrame;
audioMixer.Mix(1, &audioFrame);
/* 混音数据长度 */
int outMixingSize = audioFrame.num_channels * audioFrame.sample_per_channel_;
/* 混音数据 */
const int16_t * outMixingData = audioFrame.data();
}
