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/*
* Copyright (c) 2011 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_DEVICE_DEFINES_H_
#define API_AUDIO_AUDIO_DEVICE_DEFINES_H_
#include <stddef.h>
#include <cstdint>
#include <optional>
#include <string>
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
static const int kAdmMaxDeviceNameSize = 128;
static const int kAdmMaxFileNameSize = 512;
static const int kAdmMaxGuidSize = 128;
static const int kAdmMinPlayoutBufferSizeMs = 10;
static const int kAdmMaxPlayoutBufferSizeMs = 250;
// ----------------------------------------------------------------------------
// AudioTransport
// ----------------------------------------------------------------------------
class AudioTransport {
public:
// TODO(bugs.webrtc.org/13620) Deprecate this function
virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
size_t nSamples,
size_t nBytesPerSample,
size_t nChannels,
uint32_t samplesPerSec,
uint32_t totalDelayMS,
int32_t clockDrift,
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel) = 0; // NOLINT
virtual int32_t RecordedDataIsAvailable(
const void* audioSamples,
size_t nSamples,
size_t nBytesPerSample,
size_t nChannels,
uint32_t samplesPerSec,
uint32_t totalDelayMS,
int32_t clockDrift,
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel,
std::optional<int64_t> /* estimatedCaptureTimeNS */) { // NOLINT
// TODO(webrtc:13620) Make the default behaver of the new API to behave as
// the old API. This can be pure virtual if all uses of the old API is
// removed.
return RecordedDataIsAvailable(
audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
}
// Implementation has to setup safe values for all specified out parameters.
virtual int32_t NeedMorePlayData(size_t nSamples,
size_t nBytesPerSample,
size_t nChannels,
uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut, // NOLINT
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) = 0; // NOLINT
// Method to pull mixed render audio data from all active VoE channels.
// The data will not be passed as reference for audio processing internally.
virtual void PullRenderData(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) = 0;
protected:
virtual ~AudioTransport() {}
};
// Helper class for storage of fundamental audio parameters such as sample rate,
// number of channels, native buffer size etc.
// Note that one audio frame can contain more than one channel sample and each
// sample is assumed to be a 16-bit PCM sample. Hence, one audio frame in
// stereo contains 2 * (16/8) = 4 bytes of data.
class AudioParameters {
public:
// This implementation does only support 16-bit PCM samples.
static const size_t kBitsPerSample = 16;
AudioParameters()
: sample_rate_(0),
channels_(0),
frames_per_buffer_(0),
frames_per_10ms_buffer_(0) {}
AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer)
: sample_rate_(sample_rate),
channels_(channels),
frames_per_buffer_(frames_per_buffer),
frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
void reset(int sample_rate, size_t channels, size_t frames_per_buffer) {
sample_rate_ = sample_rate;
channels_ = channels;
frames_per_buffer_ = frames_per_buffer;
frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100);
}
size_t bits_per_sample() const { return kBitsPerSample; }
void reset(int sample_rate, size_t channels, double buffer_duration) {
reset(sample_rate, channels,
static_cast<size_t>(sample_rate * buffer_duration + 0.5));
}
void reset(int sample_rate, size_t channels) {
reset(sample_rate, channels, static_cast<size_t>(0));
}
int sample_rate() const { return sample_rate_; }
size_t channels() const { return channels_; }
size_t frames_per_buffer() const { return frames_per_buffer_; }
size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
size_t GetBytesPerBuffer() const {
return frames_per_buffer_ * GetBytesPerFrame();
}
// The WebRTC audio device buffer (ADB) only requires that the sample rate
// and number of channels are configured. Hence, to be "valid", only these
// two attributes must be set.
bool is_valid() const { return ((sample_rate_ > 0) && (channels_ > 0)); }
// Most platforms also require that a native buffer size is defined.
// An audio parameter instance is considered to be "complete" if it is both
// "valid" (can be used by the ADB) and also has a native frame size.
bool is_complete() const { return (is_valid() && (frames_per_buffer_ > 0)); }
size_t GetBytesPer10msBuffer() const {
return frames_per_10ms_buffer_ * GetBytesPerFrame();
}
double GetBufferSizeInMilliseconds() const {
if (sample_rate_ == 0)
return 0.0;
return frames_per_buffer_ / (sample_rate_ / 1000.0);
}
double GetBufferSizeInSeconds() const {
if (sample_rate_ == 0)
return 0.0;
return static_cast<double>(frames_per_buffer_) / (sample_rate_);
}
std::string ToString() const {
char ss_buf[1024];
SimpleStringBuilder ss(ss_buf);
ss << "AudioParameters: ";
ss << "sample_rate=" << sample_rate() << ", channels=" << channels();
ss << ", frames_per_buffer=" << frames_per_buffer();
ss << ", frames_per_10ms_buffer=" << frames_per_10ms_buffer();
ss << ", bytes_per_frame=" << GetBytesPerFrame();
ss << ", bytes_per_buffer=" << GetBytesPerBuffer();
ss << ", bytes_per_10ms_buffer=" << GetBytesPer10msBuffer();
ss << ", size_in_ms=" << GetBufferSizeInMilliseconds();
return ss.str();
}
private:
int sample_rate_;
size_t channels_;
size_t frames_per_buffer_;
size_t frames_per_10ms_buffer_;
};
} // namespace webrtc
#endif // API_AUDIO_AUDIO_DEVICE_DEFINES_H_
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