Files
platform-external-webrtc/webrtc/voice_engine/output_mixer.h
kwiberg e7edea9759 Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #5 id:80001 of https://codereview.chromium.org/2037623002/ )
Reason for revert:
voice_engine_unittests: FilePlayerTest.PlayWavPcm16File and FilePlayerTest.PlayWavPcmuFile fail on 32-bit android (android_rel and android-dbg try bots, Android32 Tests (L Nexus5) and Android32 Tests (L Nexus7.2) build bots).

Not sure why this would happen, since I just moved the test without modifying it. Some test filtering that no longer manages to disable them? Anyway, reverting until I know how to fix.

This was actually caught by the try bots, but I missed it because I was manually ignoring them because of an error with the bots. :-(

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> R=perkj@webrtc.org, solenberg@webrtc.org
>
> Committed: 65874b163e

TBR=perkj@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2092633002
Cr-Commit-Position: refs/heads/master@{#13267}
2016-06-22 23:29:58 +00:00

131 lines
3.9 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_OUTPUT_MIXER_H_
#define WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_H_
#include "webrtc/base/criticalsection.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/level_indicator.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
namespace webrtc {
class AudioProcessing;
class FileWrapper;
class VoEMediaProcess;
namespace voe {
class Statistics;
class OutputMixer : public AudioMixerOutputReceiver,
public FileCallback
{
public:
static int32_t Create(OutputMixer*& mixer, uint32_t instanceId);
static void Destroy(OutputMixer*& mixer);
int32_t SetEngineInformation(Statistics& engineStatistics);
int32_t SetAudioProcessingModule(
AudioProcessing* audioProcessingModule);
// VoEExternalMedia
int RegisterExternalMediaProcessing(
VoEMediaProcess& proccess_object);
int DeRegisterExternalMediaProcessing();
int32_t MixActiveChannels();
int32_t DoOperationsOnCombinedSignal(bool feed_data_to_apm);
int32_t SetMixabilityStatus(MixerParticipant& participant,
bool mixable);
int32_t SetAnonymousMixabilityStatus(MixerParticipant& participant,
bool mixable);
int GetMixedAudio(int sample_rate_hz, size_t num_channels,
AudioFrame* audioFrame);
// VoEVolumeControl
int GetSpeechOutputLevel(uint32_t& level);
int GetSpeechOutputLevelFullRange(uint32_t& level);
int SetOutputVolumePan(float left, float right);
int GetOutputVolumePan(float& left, float& right);
// VoEFile
int StartRecordingPlayout(const char* fileName,
const CodecInst* codecInst);
int StartRecordingPlayout(OutStream* stream,
const CodecInst* codecInst);
int StopRecordingPlayout();
virtual ~OutputMixer();
// from AudioMixerOutputReceiver
virtual void NewMixedAudio(
int32_t id,
const AudioFrame& generalAudioFrame,
const AudioFrame** uniqueAudioFrames,
uint32_t size);
// For file recording
void PlayNotification(int32_t id, uint32_t durationMs);
void RecordNotification(int32_t id, uint32_t durationMs);
void PlayFileEnded(int32_t id);
void RecordFileEnded(int32_t id);
private:
OutputMixer(uint32_t instanceId);
// uses
Statistics* _engineStatisticsPtr;
AudioProcessing* _audioProcessingModulePtr;
rtc::CriticalSection _callbackCritSect;
// protect the _outputFileRecorderPtr and _outputFileRecording
rtc::CriticalSection _fileCritSect;
AudioConferenceMixer& _mixerModule;
AudioFrame _audioFrame;
// Converts mixed audio to the audio device output rate.
PushResampler<int16_t> resampler_;
// Converts mixed audio to the audio processing rate.
PushResampler<int16_t> audioproc_resampler_;
AudioLevel _audioLevel; // measures audio level for the combined signal
int _instanceId;
VoEMediaProcess* _externalMediaCallbackPtr;
bool _externalMedia;
float _panLeft;
float _panRight;
int _mixingFrequencyHz;
FileRecorder* _outputFileRecorderPtr;
bool _outputFileRecording;
};
} // namespace voe
} // namespace werbtc
#endif // VOICE_ENGINE_OUTPUT_MIXER_H_