webrtc_m130/webrtc/voice_engine/voe_base_impl.h
wu@webrtc.org 94454b71ad Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.

Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.

Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.

BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc

Review URL: https://webrtc-codereview.appspot.com/14559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 20:34:08 +00:00

169 lines
6.1 KiB
C++

/*
* Copyright (c) 2012 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_VOICE_ENGINE_VOE_BASE_IMPL_H
#define WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/voice_engine/shared_data.h"
namespace webrtc
{
class ProcessThread;
class VoEBaseImpl: public VoEBase,
public AudioTransport,
public AudioDeviceObserver
{
public:
virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
virtual int DeRegisterVoiceEngineObserver();
virtual int Init(AudioDeviceModule* external_adm = NULL,
AudioProcessing* audioproc = NULL);
virtual AudioProcessing* audio_processing() {
return _shared->audio_processing();
}
virtual int Terminate();
virtual int CreateChannel();
virtual int CreateChannel(const Config& config);
virtual int DeleteChannel(int channel);
virtual int StartReceive(int channel);
virtual int StartPlayout(int channel);
virtual int StartSend(int channel);
virtual int StopReceive(int channel);
virtual int StopPlayout(int channel);
virtual int StopSend(int channel);
virtual int GetVersion(char version[1024]);
virtual int LastError();
virtual AudioTransport* audio_transport() { return this; }
// AudioTransport
virtual int32_t
RecordedDataIsAvailable(const void* audioSamples,
uint32_t nSamples,
uint8_t nBytesPerSample,
uint8_t nChannels,
uint32_t samplesPerSec,
uint32_t totalDelayMS,
int32_t clockDrift,
uint32_t micLevel,
bool keyPressed,
uint32_t& newMicLevel);
virtual int32_t NeedMorePlayData(uint32_t nSamples,
uint8_t nBytesPerSample,
uint8_t nChannels,
uint32_t samplesPerSec,
void* audioSamples,
uint32_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms);
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 volume,
bool key_pressed,
bool need_audio_processing);
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);
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);
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);
// AudioDeviceObserver
virtual void OnErrorIsReported(ErrorCode error);
virtual void OnWarningIsReported(WarningCode warning);
protected:
VoEBaseImpl(voe::SharedData* shared);
virtual ~VoEBaseImpl();
private:
int32_t StartPlayout();
int32_t StopPlayout();
int32_t StartSend();
int32_t StopSend();
int32_t TerminateInternal();
// Helper function to process the recorded data with AudioProcessing Module,
// demultiplex the data to specific voe channels, encode and send to the
// network. When |number_of_VoE_channels| is 0, it will demultiplex the
// data to all the existing VoE channels.
// It returns new AGC microphone volume or 0 if no volume changes
// should be done.
int ProcessRecordedDataWithAPM(const int voe_channels[],
int number_of_voe_channels,
const void* audio_data,
uint32_t sample_rate,
uint8_t number_of_channels,
uint32_t number_of_frames,
uint32_t audio_delay_milliseconds,
int32_t clock_drift,
uint32_t volume,
bool key_pressed);
void GetPlayoutData(int sample_rate, int number_of_channels,
int number_of_frames, bool feed_data_to_apm,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms);
int32_t AddBuildInfo(char* str) const;
int32_t AddVoEVersion(char* str) const;
// Initialize channel by setting Engine Information then initializing
// channel.
int InitializeChannel(voe::ChannelOwner* channel_owner);
#ifdef WEBRTC_EXTERNAL_TRANSPORT
int32_t AddExternalTransportBuild(char* str) const;
#endif
VoiceEngineObserver* _voiceEngineObserverPtr;
CriticalSectionWrapper& _callbackCritSect;
bool _voiceEngineObserver;
AudioFrame _audioFrame;
voe::SharedData* _shared;
};
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H