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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 WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
#define WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
#include <stddef.h>
#include "webrtc/typedefs.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;
// ----------------------------------------------------------------------------
// AudioDeviceObserver
// ----------------------------------------------------------------------------
class AudioDeviceObserver {
public:
enum ErrorCode { kRecordingError = 0, kPlayoutError = 1 };
enum WarningCode { kRecordingWarning = 0, kPlayoutWarning = 1 };
virtual void OnErrorIsReported(const ErrorCode error) = 0;
virtual void OnWarningIsReported(const WarningCode warning) = 0;
protected:
virtual ~AudioDeviceObserver() {}
};
// ----------------------------------------------------------------------------
// AudioTransport
// ----------------------------------------------------------------------------
class AudioTransport {
public:
virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
const uint32_t currentMicLevel,
const bool keyPressed,
uint32_t& newMicLevel) = 0;
virtual int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) = 0;
// Method to pass captured data directly and unmixed to network channels.
// |channel_ids| contains a list of VoE channels which are the
// sinks to the capture data. |audio_delay_milliseconds| is the sum of
// recording delay and playout delay of the hardware. |current_volume| is
// in the range of [0, 255], representing the current microphone analog
// volume. |key_pressed| is used by the typing detection.
// |need_audio_processing| specify if the data needs to be processed by APM.
// Currently WebRtc supports only one APM, and Chrome will make sure only
// one stream goes through APM. When |need_audio_processing| is false, the
// values of |audio_delay_milliseconds|, |current_volume| and |key_pressed|
// will be ignored.
// The return value is the new microphone volume, in the range of |0, 255].
// When the volume does not need to be updated, it returns 0.
// TODO(xians): Remove this interface after Chrome and Libjingle switches
// to OnData().
virtual int OnDataAvailable(const int voe_channels[],
int number_of_voe_channels,
const int16_t* audio_data,
int sample_rate,
int number_of_channels,
size_t number_of_frames,
int audio_delay_milliseconds,
int current_volume,
bool key_pressed,
bool need_audio_processing) {
return 0;
}
// Method to pass the captured audio data to the specific VoE channel.
// |voe_channel| is the id of the VoE channel which is the sink to the
// capture data.
// TODO(xians): Remove this interface after Libjingle switches to
// PushCaptureData().
virtual void OnData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
int number_of_channels,
size_t number_of_frames) {}
// Method to push the captured audio data to the specific VoE channel.
// The data will not undergo audio processing.
// |voe_channel| is the id of the VoE channel which is the sink to the
// capture data.
// TODO(xians): Make the interface pure virtual after Libjingle
// has its implementation.
virtual void PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
int number_of_channels,
size_t number_of_frames) {}
// Method to pull mixed render audio data from all active VoE channels.
// The data will not be passed as reference for audio processing internally.
// TODO(xians): Support getting the unmixed render data from specific VoE
// channel.
virtual void PullRenderData(int bits_per_sample,
int sample_rate,
int number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {}
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, int 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, int 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, int channels, double ms_per_buffer) {
reset(sample_rate, channels,
static_cast<size_t>(sample_rate * ms_per_buffer + 0.5));
}
void reset(int sample_rate, int channels) {
reset(sample_rate, channels, static_cast<size_t>(0));
}
int sample_rate() const { return sample_rate_; }
int 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_);
}
private:
int sample_rate_;
int channels_;
size_t frames_per_buffer_;
size_t frames_per_10ms_buffer_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
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