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/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef API_AUDIO_AUDIO_FRAME_H_
#define API_AUDIO_AUDIO_FRAME_H_
#include <stdint.h>
#include <stdlib.h>
#include "rtc_base/constructormagic.h"
#include "rtc_base/deprecation.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
/* This class holds up to 60 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, 60 ms (2 * 32 * 60)
kMaxDataSizeSamples = 3840,
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();
// TODO(solenberg): Remove once downstream users of AudioFrame have updated.
RTC_DEPRECATED
void UpdateFrame(int id, 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) {
RTC_UNUSED(id);
UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
speech_type, vad_activity, num_channels);
}
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;
// These methods are deprecated. Use the functions in
// webrtc/audio/utility instead. These methods will exists for a
// short period of time until webrtc clients have updated. See
// webrtc:6548 for details.
RTC_DEPRECATED AudioFrame& operator>>=(const int rhs);
RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs);
// 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, rtc::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_