The ADM(AudioDeviceModule) is responsible for driving input (microphone) and output (speaker) audio in WebRTC and the API is defined in audio_device.h.
Main functions of the ADM are:
ADM implementations reside at two different locations in the WebRTC repository:
/sdk/. The latest implementations for iOS and Android can be found under
/modules/audio_device/ contains older versions for mobile platforms and also implementations for desktop platforms such as Linux, Windows and Mac OSX. This document is focusing on the parts in
/modules/audio_device/ but implementation specific details such as threading models are omitted to keep the descriptions as simple as possible.
By default, the ADM in WebRTC is created in
WebRtcVoiceEngine::Init but an external implementation can also be injected using
rtc::CreatePeerConnectionFactory. An example of where an external ADM is injected can be found in PeerConnectionInterfaceTest where a so-called fake ADM is utilized to avoid hardware dependency in a gtest. Clients can also inject their own ADMs in situations where functionality is needed that is not provided by the default implementations.
This section contains a historical background of the ADM API.
The ADM interface is old and has undergone many changes over the years. It used to be much more granular but it still contains more than 50 methods and is implemented on several different hardware platforms.
Some APIs are not implemented on all platforms, and functionality can be spread out differently between the methods.
Desktop version are not updated to comply with the latest C++ style guide and more work is also needed to improve the performance and stability of these versions.
WebRtcVoiceEngine does not utilize all methods of the ADM but it still serves as the best example of its architecture and how to use it. For a more detailed view of all methods in the ADM interface, see ADM unit tests.
Basic initialization is done using a utility method called
adm_helpers::Init which calls fundamental ADM APIs like:
AudiDeviceModule::Init- initializes the native audio parts required for each platform.
AudiDeviceModule::SetPlayoutDevice- specifies which speaker to use for playing out audio using an
indexretrieved by the corresponding enumeration method
AudiDeviceModule::SetRecordingDevice- specifies which microphone to use for recording audio using an
indexretrieved by the corresponding enumeration method which is
AudiDeviceModule::InitSpeaker- sets up the parts of the ADM needed to use the selected output device.
AudiDeviceModule::InitMicrophone- sets up the parts of the ADM needed to use the selected input device.
AudiDeviceModule::SetStereoPlayout- enables playout in stereo if the selected audio device supports it.
AudiDeviceModule::SetStereoRecording- enables recording in stereo if the selected audio device supports it.
WebRtcVoiceEngine::Init also calls
AudiDeviceModule::RegisterAudioTransport to register an existing AudioTransport implementation which handles audio callbacks in both directions and therefore serves as the bridge between the native ADM and the upper WebRTC layers.
Recorded audio samples are delivered from the ADM to the
WebRtcVoiceEngine (who owns the
AudioTransport object) via
int32_t RecordedDataIsAvailable(const void* audioSamples, size_t nSamples, size_t nBytesPerSample, size_t nChannels, uint32_t samplesPerSec, uint32_t totalDelayMS, int32_t clockDrift, uint32_t currentMicLevel, bool keyPressed, uint32_t& newMicLevel)
Decoded audio samples ready to be played out are are delivered by the
WebRtcVoiceEngine to the ADM, via
int32_t NeedMorePlayData(size_t nSamples, size_t nBytesPerSample, size_t nChannels, int32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms)
Audio samples are 16-bit linear PCM using regular interleaving of channels within each sample.
WebRtcVoiceEngine also owns an
AudioState member and this class is used has helper to start and stop audio to and from the ADM. To initialize and start recording, it calls:
and to initialize and start playout: