2f84afad30
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk. Add new interface to support multiple sources in webrtc. CaptureData() will be called by chrome with a flag |need_audio_processing| to indicate if the data needs to be processed by APM or not. Different from the old interface that will send the data to all voe channels, the new interface will specify a list of voe channels that the data is demultiplexing to. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4449 4adac7df-926f-26a2-2b94-8c16560cd09d
108 lines
4.3 KiB
C++
108 lines
4.3 KiB
C++
/*
|
|
* 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) = 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): Make the interface pure virtual after libjingle has its
|
|
// implementation.
|
|
virtual int OnDataAvailable(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; }
|
|
|
|
protected:
|
|
virtual ~AudioTransport() {}
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
|