8804a29951
TEST=try bots BUG=1205 R=henrike@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5019 4adac7df-926f-26a2-2b94-8c16560cd09d
295 lines
13 KiB
C++
295 lines
13 KiB
C++
/*
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* libjingle
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* Copyright 2012, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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// This class implements an AudioCaptureModule that can be used to detect if
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// audio is being received properly if it is fed by another AudioCaptureModule
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// in some arbitrary audio pipeline where they are connected. It does not play
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// out or record any audio so it does not need access to any hardware and can
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// therefore be used in the gtest testing framework.
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// Note P postfix of a function indicates that it should only be called by the
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// processing thread.
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#ifndef TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
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#define TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
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#include "talk/base/basictypes.h"
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#include "talk/base/criticalsection.h"
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#include "talk/base/messagehandler.h"
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#include "talk/base/scoped_ref_ptr.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_device/include/audio_device.h"
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namespace talk_base {
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class Thread;
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} // namespace talk_base
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class FakeAudioCaptureModule
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: public webrtc::AudioDeviceModule,
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public talk_base::MessageHandler {
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public:
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typedef uint16 Sample;
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// The value for the following constants have been derived by running VoE
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// using a real ADM. The constants correspond to 10ms of mono audio at 44kHz.
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enum{kNumberSamples = 440};
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enum{kNumberBytesPerSample = sizeof(Sample)};
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// Creates a FakeAudioCaptureModule or returns NULL on failure.
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// |process_thread| is used to push and pull audio frames to and from the
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// returned instance. Note: ownership of |process_thread| is not handed over.
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static talk_base::scoped_refptr<FakeAudioCaptureModule> Create(
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talk_base::Thread* process_thread);
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// Returns the number of frames that have been successfully pulled by the
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// instance. Note that correctly detecting success can only be done if the
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// pulled frame was generated/pushed from a FakeAudioCaptureModule.
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int frames_received() const;
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// Following functions are inherited from webrtc::AudioDeviceModule.
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// Only functions called by PeerConnection are implemented, the rest do
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// nothing and return success. If a function is not expected to be called by
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// PeerConnection an assertion is triggered if it is in fact called.
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virtual int32_t Version(char* version,
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uint32_t& remaining_buffer_in_bytes,
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uint32_t& position) const;
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virtual int32_t TimeUntilNextProcess();
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virtual int32_t Process();
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virtual int32_t ChangeUniqueId(const int32_t id);
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virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const;
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virtual ErrorCode LastError() const;
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virtual int32_t RegisterEventObserver(
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webrtc::AudioDeviceObserver* event_callback);
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// Note: Calling this method from a callback may result in deadlock.
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virtual int32_t RegisterAudioCallback(webrtc::AudioTransport* audio_callback);
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virtual int32_t Init();
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virtual int32_t Terminate();
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virtual bool Initialized() const;
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virtual int16_t PlayoutDevices();
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virtual int16_t RecordingDevices();
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virtual int32_t PlayoutDeviceName(uint16_t index,
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char name[webrtc::kAdmMaxDeviceNameSize],
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char guid[webrtc::kAdmMaxGuidSize]);
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virtual int32_t RecordingDeviceName(uint16_t index,
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char name[webrtc::kAdmMaxDeviceNameSize],
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char guid[webrtc::kAdmMaxGuidSize]);
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virtual int32_t SetPlayoutDevice(uint16_t index);
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virtual int32_t SetPlayoutDevice(WindowsDeviceType device);
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virtual int32_t SetRecordingDevice(uint16_t index);
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virtual int32_t SetRecordingDevice(WindowsDeviceType device);
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virtual int32_t PlayoutIsAvailable(bool* available);
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virtual int32_t InitPlayout();
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virtual bool PlayoutIsInitialized() const;
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virtual int32_t RecordingIsAvailable(bool* available);
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virtual int32_t InitRecording();
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virtual bool RecordingIsInitialized() const;
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virtual int32_t StartPlayout();
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virtual int32_t StopPlayout();
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virtual bool Playing() const;
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virtual int32_t StartRecording();
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virtual int32_t StopRecording();
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virtual bool Recording() const;
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virtual int32_t SetAGC(bool enable);
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virtual bool AGC() const;
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virtual int32_t SetWaveOutVolume(uint16_t volume_left,
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uint16_t volume_right);
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virtual int32_t WaveOutVolume(uint16_t* volume_left,
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uint16_t* volume_right) const;
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virtual int32_t SpeakerIsAvailable(bool* available);
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virtual int32_t InitSpeaker();
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virtual bool SpeakerIsInitialized() const;
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virtual int32_t MicrophoneIsAvailable(bool* available);
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virtual int32_t InitMicrophone();
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virtual bool MicrophoneIsInitialized() const;
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virtual int32_t SpeakerVolumeIsAvailable(bool* available);
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virtual int32_t SetSpeakerVolume(uint32_t volume);
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virtual int32_t SpeakerVolume(uint32_t* volume) const;
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virtual int32_t MaxSpeakerVolume(uint32_t* max_volume) const;
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virtual int32_t MinSpeakerVolume(uint32_t* min_volume) const;
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virtual int32_t SpeakerVolumeStepSize(uint16_t* step_size) const;
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virtual int32_t MicrophoneVolumeIsAvailable(bool* available);
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virtual int32_t SetMicrophoneVolume(uint32_t volume);
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virtual int32_t MicrophoneVolume(uint32_t* volume) const;
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virtual int32_t MaxMicrophoneVolume(uint32_t* max_volume) const;
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virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const;
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virtual int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const;
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virtual int32_t SpeakerMuteIsAvailable(bool* available);
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virtual int32_t SetSpeakerMute(bool enable);
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virtual int32_t SpeakerMute(bool* enabled) const;
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virtual int32_t MicrophoneMuteIsAvailable(bool* available);
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virtual int32_t SetMicrophoneMute(bool enable);
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virtual int32_t MicrophoneMute(bool* enabled) const;
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virtual int32_t MicrophoneBoostIsAvailable(bool* available);
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virtual int32_t SetMicrophoneBoost(bool enable);
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virtual int32_t MicrophoneBoost(bool* enabled) const;
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virtual int32_t StereoPlayoutIsAvailable(bool* available) const;
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virtual int32_t SetStereoPlayout(bool enable);
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virtual int32_t StereoPlayout(bool* enabled) const;
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virtual int32_t StereoRecordingIsAvailable(bool* available) const;
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virtual int32_t SetStereoRecording(bool enable);
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virtual int32_t StereoRecording(bool* enabled) const;
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virtual int32_t SetRecordingChannel(const ChannelType channel);
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virtual int32_t RecordingChannel(ChannelType* channel) const;
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virtual int32_t SetPlayoutBuffer(const BufferType type,
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uint16_t size_ms = 0);
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virtual int32_t PlayoutBuffer(BufferType* type,
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uint16_t* size_ms) const;
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virtual int32_t PlayoutDelay(uint16_t* delay_ms) const;
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virtual int32_t RecordingDelay(uint16_t* delay_ms) const;
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virtual int32_t CPULoad(uint16_t* load) const;
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virtual int32_t StartRawOutputFileRecording(
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const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]);
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virtual int32_t StopRawOutputFileRecording();
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virtual int32_t StartRawInputFileRecording(
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const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]);
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virtual int32_t StopRawInputFileRecording();
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virtual int32_t SetRecordingSampleRate(const uint32_t samples_per_sec);
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virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const;
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virtual int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec);
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virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const;
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virtual int32_t ResetAudioDevice();
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virtual int32_t SetLoudspeakerStatus(bool enable);
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virtual int32_t GetLoudspeakerStatus(bool* enabled) const;
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// End of functions inherited from webrtc::AudioDeviceModule.
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// The following function is inherited from talk_base::MessageHandler.
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virtual void OnMessage(talk_base::Message* msg);
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protected:
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// The constructor is protected because the class needs to be created as a
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// reference counted object (for memory managment reasons). It could be
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// exposed in which case the burden of proper instantiation would be put on
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// the creator of a FakeAudioCaptureModule instance. To create an instance of
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// this class use the Create(..) API.
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explicit FakeAudioCaptureModule(talk_base::Thread* process_thread);
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// The destructor is protected because it is reference counted and should not
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// be deleted directly.
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virtual ~FakeAudioCaptureModule();
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private:
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// Initializes the state of the FakeAudioCaptureModule. This API is called on
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// creation by the Create() API.
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bool Initialize();
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// SetBuffer() sets all samples in send_buffer_ to |value|.
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void SetSendBuffer(int value);
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// Resets rec_buffer_. I.e., sets all rec_buffer_ samples to 0.
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void ResetRecBuffer();
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// Returns true if rec_buffer_ contains one or more sample greater than or
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// equal to |value|.
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bool CheckRecBuffer(int value);
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// Returns true/false depending on if recording or playback has been
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// enabled/started.
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bool ShouldStartProcessing();
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// Starts or stops the pushing and pulling of audio frames.
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void UpdateProcessing(bool start);
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// Starts the periodic calling of ProcessFrame() in a thread safe way.
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void StartProcessP();
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// Periodcally called function that ensures that frames are pulled and pushed
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// periodically if enabled/started.
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void ProcessFrameP();
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// Pulls frames from the registered webrtc::AudioTransport.
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void ReceiveFrameP();
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// Pushes frames to the registered webrtc::AudioTransport.
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void SendFrameP();
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// Stops the periodic calling of ProcessFrame() in a thread safe way.
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void StopProcessP();
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// The time in milliseconds when Process() was last called or 0 if no call
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// has been made.
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uint32 last_process_time_ms_;
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// Callback for playout and recording.
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webrtc::AudioTransport* audio_callback_;
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bool recording_; // True when audio is being pushed from the instance.
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bool playing_; // True when audio is being pulled by the instance.
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bool play_is_initialized_; // True when the instance is ready to pull audio.
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bool rec_is_initialized_; // True when the instance is ready to push audio.
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// Input to and output from RecordedDataIsAvailable(..) makes it possible to
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// modify the current mic level. The implementation does not care about the
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// mic level so it just feeds back what it receives.
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uint32_t current_mic_level_;
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// next_frame_time_ is updated in a non-drifting manner to indicate the next
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// wall clock time the next frame should be generated and received. started_
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// ensures that next_frame_time_ can be initialized properly on first call.
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bool started_;
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uint32 next_frame_time_;
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// User provided thread context.
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talk_base::Thread* process_thread_;
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// Buffer for storing samples received from the webrtc::AudioTransport.
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char rec_buffer_[kNumberSamples * kNumberBytesPerSample];
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// Buffer for samples to send to the webrtc::AudioTransport.
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char send_buffer_[kNumberSamples * kNumberBytesPerSample];
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// Counter of frames received that have samples of high enough amplitude to
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// indicate that the frames are not faked somewhere in the audio pipeline
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// (e.g. by a jitter buffer).
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int frames_received_;
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// Protects variables that are accessed from process_thread_ and
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// the main thread.
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mutable talk_base::CriticalSection crit_;
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// Protects |audio_callback_| that is accessed from process_thread_ and
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// the main thread.
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talk_base::CriticalSection crit_callback_;
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};
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#endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
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