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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_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_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 uint32_t nSamples,
const uint8_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 uint32_t nSamples,
const uint8_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
uint32_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,
int 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,
int 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,
int 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, int number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {}
protected:
virtual ~AudioTransport() {}
};
} // namespace webrtc
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
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