在webrtc的API模块中，分析audio文件夹下的内容，该文件主要包括：
这里仅针对Audio_frame..h和Audio_mixer.h文件进行分析，其他文件重要性一般，暂且不分析。
Audio_frame.h文件：
#ifndef API_AUDIO_AUDIO_FRAME_H_
#define API_AUDIO_AUDIO_FRAME_H_
#include <stddef.h>
#include <stdint.h>
#include "rtc_base/constructormagic.h"
namespace webrtc {
/* This class holds up to 120 ms of super-wideband (32 kHz) stereo audio. It
* allows for adding and subtracting frames while keeping track of the resulting
* states.
*
* Notes
* - This is a de-facto api, not designed for external use. The AudioFrame class
* is in need of overhaul or even replacement, and anyone depending on it
* should be prepared for that.
* - The total number of samples is samples_per_channel_ * num_channels_.
* - Stereo data is interleaved starting with the left channel.
*/
class AudioFrame {
public:
// Using constexpr here causes linker errors unless the variable also has an
// out-of-class definition, which is impractical in this header-only class.
// (This makes no sense because it compiles as an enum value, which we most
// certainly cannot take the address of, just fine.) C++17 introduces inline
// variables which should allow us to switch to constexpr and keep this a
// header-only class.
enum : size_t {
// Stereo, 32 kHz, 120 ms (2 * 32 * 120)
// Stereo, 192 kHz, 20 ms (2 * 192 * 20)
kMaxDataSizeSamples = 7680,
kMaxDataSizeBytes = kMaxDataSizeSamples * sizeof(int16_t),
};
enum VADActivity { kVadActive = 0, kVadPassive = 1, kVadUnknown = 2 };
enum SpeechType {
kNormalSpeech = 0,
kPLC = 1,
kCNG = 2,
kPLCCNG = 3,
kUndefined = 4
};
AudioFrame();
// Resets all members to their default state.
void Reset();
// Same as Reset(), but leaves mute state unchanged. Muting a frame requires
// the buffer to be zeroed on the next call to mutable_data(). Callers
// intending to write to the buffer immediately after Reset() can instead use
// ResetWithoutMuting() to skip this wasteful zeroing.
void ResetWithoutMuting();
void UpdateFrame(uint32_t timestamp,
const int16_t* data,
size_t samples_per_channel,
int sample_rate_hz,
SpeechType speech_type,
VADActivity vad_activity,
size_t num_channels = 1);
void CopyFrom(const AudioFrame& src);
// Sets a wall-time clock timestamp in milliseconds to be used for profiling
// of time between two points in the audio chain.
// Example:
// t0: UpdateProfileTimeStamp()
// t1: ElapsedProfileTimeMs() => t1 - t0 [msec]
void UpdateProfileTimeStamp();
// Returns the time difference between now and when UpdateProfileTimeStamp()
// was last called. Returns -1 if UpdateProfileTimeStamp() has not yet been
// called.
int64_t ElapsedProfileTimeMs() const;
// data() returns a zeroed static buffer if the frame is muted.
// mutable_frame() always returns a non-static buffer; the first call to
// mutable_frame() zeros the non-static buffer and marks the frame unmuted.
const int16_t* data() const;
int16_t* mutable_data();
// Prefer to mute frames using AudioFrameOperations::Mute.
void Mute();
// Frame is muted by default.
bool muted() const;
// RTP timestamp of the first sample in the AudioFrame.
uint32_t timestamp_ = 0;
// Time since the first frame in milliseconds.
// -1 represents an uninitialized value.
int64_t elapsed_time_ms_ = -1;
// NTP time of the estimated capture time in local timebase in milliseconds.
// -1 represents an uninitialized value.
int64_t ntp_time_ms_ = -1;
size_t samples_per_channel_ = 0;
int sample_rate_hz_ = 0;
size_t num_channels_ = 0;
SpeechType speech_type_ = kUndefined;
VADActivity vad_activity_ = kVadUnknown;
// Monotonically increasing timestamp intended for profiling of audio frames.
// Typically used for measuring elapsed time between two different points in
// the audio path. No lock is used to save resources and we are thread safe
// by design. Also, absl::optional is not used since it will cause a "complex
// class/struct needs an explicit out-of-line destructor" build error.
int64_t profile_timestamp_ms_ = 0;
private:
// A permamently zeroed out buffer to represent muted frames. This is a
// header-only class, so the only way to avoid creating a separate empty
// buffer per translation unit is to wrap a static in an inline function.
static const int16_t* empty_data();
int16_t data_[kMaxDataSizeSamples];
bool muted_ = true;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
};
} // namespace webrtc
#endif // API_AUDIO_AUDIO_FRAME_H_
Audio_Muxer.h文件：
#ifndef API_AUDIO_AUDIO_MIXER_H_
#define API_AUDIO_AUDIO_MIXER_H_
#include <memory>
#include "api/audio/audio_frame.h"
#include "rtc_base/refcount.h"
namespace webrtc {
// WORK IN PROGRESS
// This class is under development and is not yet intended for for use outside
// of WebRtc/Libjingle.
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 {}
};
} // namespace webrtc
#endif // API_AUDIO_AUDIO_MIXER_H_