Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(806)

Unified Diff: webrtc/modules/audio_device/audio_device_buffer.h

Issue 2256833003: Cleanup of the AudioDeviceBuffer class (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Ensures that all tests passes Created 4 years, 4 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « no previous file | webrtc/modules/audio_device/audio_device_buffer.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/modules/audio_device/audio_device_buffer.h
diff --git a/webrtc/modules/audio_device/audio_device_buffer.h b/webrtc/modules/audio_device/audio_device_buffer.h
index f49420c98ea22d67b9bee1a727d112babb9def35..ba484a57c780a828e2a00ff08aa95ccea5c14d26 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.h
+++ b/webrtc/modules/audio_device/audio_device_buffer.h
@@ -21,8 +21,6 @@
namespace webrtc {
class CriticalSectionWrapper;
-const uint32_t kPulsePeriodMs = 1000;
-const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
// Delta times between two successive playout callbacks are limited to this
// value before added to an internal array.
const size_t kMaxDeltaTimeInMs = 500;
@@ -35,40 +33,47 @@ class AudioDeviceBuffer {
virtual ~AudioDeviceBuffer();
void SetId(uint32_t id) {};
- int32_t RegisterAudioCallback(AudioTransport* audioCallback);
+ int32_t RegisterAudioCallback(AudioTransport* audio_callback);
int32_t InitPlayout();
int32_t InitRecording();
- virtual int32_t SetRecordingSampleRate(uint32_t fsHz);
- virtual int32_t SetPlayoutSampleRate(uint32_t fsHz);
+ int32_t SetRecordingSampleRate(uint32_t fsHz);
+ int32_t SetPlayoutSampleRate(uint32_t fsHz);
int32_t RecordingSampleRate() const;
int32_t PlayoutSampleRate() const;
- virtual int32_t SetRecordingChannels(size_t channels);
- virtual int32_t SetPlayoutChannels(size_t channels);
+ int32_t SetRecordingChannels(size_t channels);
+ int32_t SetPlayoutChannels(size_t channels);
size_t RecordingChannels() const;
size_t PlayoutChannels() const;
int32_t SetRecordingChannel(const AudioDeviceModule::ChannelType channel);
int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const;
- virtual int32_t SetRecordedBuffer(const void* audioBuffer, size_t nSamples);
+ virtual int32_t SetRecordedBuffer(const void* audio_buffer,
+ size_t num_samples);
int32_t SetCurrentMicLevel(uint32_t level);
- virtual void SetVQEData(int playDelayMS, int recDelayMS, int clockDrift);
+ virtual void SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift);
virtual int32_t DeliverRecordedData();
uint32_t NewMicLevel() const;
- virtual int32_t RequestPlayoutData(size_t nSamples);
- virtual int32_t GetPlayoutData(void* audioBuffer);
+ virtual int32_t RequestPlayoutData(size_t num_samples);
+ virtual int32_t GetPlayoutData(void* audio_buffer);
+ // TODO(henrika): these methods should not be used and does not contain any
+ // valid implementation. Investigate the possibility to either remove them
+ // or add a proper implementation if needed.
int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]);
int32_t StopInputFileRecording();
int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]);
int32_t StopOutputFileRecording();
- int32_t SetTypingStatus(bool typingStatus);
+ int32_t SetTypingStatus(bool typing_status);
private:
+ void AllocatePlayoutBufferIfNeeded();
+ void AllocateRecordingBufferIfNeeded();
+
// Posts the first delayed task in the task queue and starts the periodic
// timer.
void StartTimer();
@@ -86,11 +91,15 @@ class AudioDeviceBuffer {
// creates this object.
rtc::ThreadChecker thread_checker_;
+ // Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
+ // and it must outlive this object.
+ AudioTransport* audio_transport_cb_;
+
+ // TODO(henrika): given usage of thread checker, it should be possible to
+ // remove all locks in this class.
rtc::CriticalSection _critSect;
rtc::CriticalSection _critSectCb;
- AudioTransport* _ptrCbAudioTransport;
-
// Task queue used to invoke LogStats() periodically. Tasks are executed on a
// worker thread but it does not necessarily have to be the same thread for
// each task.
@@ -99,45 +108,50 @@ class AudioDeviceBuffer {
// Ensures that the timer is only started once.
bool timer_has_started_;
- uint32_t _recSampleRate;
- uint32_t _playSampleRate;
+ // Sample rate in Hertz.
+ uint32_t rec_sample_rate_;
+ uint32_t play_sample_rate_;
- size_t _recChannels;
- size_t _playChannels;
+ // Number of audio channels.
+ size_t rec_channels_;
+ size_t play_channels_;
// selected recording channel (left/right/both)
- AudioDeviceModule::ChannelType _recChannel;
-
- // 2 or 4 depending on mono or stereo
- size_t _recBytesPerSample;
- size_t _playBytesPerSample;
+ AudioDeviceModule::ChannelType rec_channel_;
- // 10ms in stereo @ 96kHz
- int8_t _recBuffer[kMaxBufferSizeBytes];
+ // Number of bytes per audio sample (2 or 4).
+ size_t rec_bytes_per_sample_;
+ size_t play_bytes_per_sample_;
- // one sample <=> 2 or 4 bytes
- size_t _recSamples;
- size_t _recSize; // in bytes
+ // Number of audio samples/bytes per 10ms.
+ size_t rec_samples_per_10ms_;
+ size_t rec_bytes_per_10ms_;
+ size_t play_samples_per_10ms_;
+ size_t play_bytes_per_10ms_;
- // 10ms in stereo @ 96kHz
- int8_t _playBuffer[kMaxBufferSizeBytes];
+ // Buffer used for recorded audio samples. Size is given by
+ // |rec_bytes_per_10ms_| and the buffer is allocated in InitRecording() on the
+ // main/creating thread.
+ std::unique_ptr<int8_t[]> rec_buffer_;
- // one sample <=> 2 or 4 bytes
- size_t _playSamples;
- size_t _playSize; // in bytes
+ // Buffer used for audio samples to be played out. Size is given by
+ // |play_bytes_per_10ms_| and the buffer is allocated in InitPlayout() on the
+ // main/creating thread.
+ std::unique_ptr<int8_t[]> play_buffer_;
- FileWrapper& _recFile;
- FileWrapper& _playFile;
+ // AGC parameters.
+ uint32_t current_mic_level_;
+ uint32_t new_mic_level_;
- uint32_t _currentMicLevel;
- uint32_t _newMicLevel;
+ // Contains true of a key-press has been detected.
+ bool typing_status_;
- bool _typingStatus;
+ // Delay values used by the AEC.
+ int play_delay_ms_;
+ int rec_delay_ms_;
- int _playDelayMS;
- int _recDelayMS;
- int _clockDrift;
- int high_delay_counter_;
+ // Contains a clock-drift measurement.
+ int clock_drift_;
// Counts number of times LogStats() has been called.
size_t num_stat_reports_;
« no previous file with comments | « no previous file | webrtc/modules/audio_device/audio_device_buffer.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698