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

Issue 2260183002: Revert of Cleanup of the AudioDeviceBuffer class (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 months ago
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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 ba484a57c780a828e2a00ff08aa95ccea5c14d26..f49420c98ea22d67b9bee1a727d112babb9def35 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.h
+++ b/webrtc/modules/audio_device/audio_device_buffer.h
@@ -21,6 +21,8 @@
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;
@@ -33,47 +35,40 @@
virtual ~AudioDeviceBuffer();
void SetId(uint32_t id) {};
- int32_t RegisterAudioCallback(AudioTransport* audio_callback);
+ int32_t RegisterAudioCallback(AudioTransport* audioCallback);
int32_t InitPlayout();
int32_t InitRecording();
- int32_t SetRecordingSampleRate(uint32_t fsHz);
- int32_t SetPlayoutSampleRate(uint32_t fsHz);
+ virtual int32_t SetRecordingSampleRate(uint32_t fsHz);
+ virtual int32_t SetPlayoutSampleRate(uint32_t fsHz);
int32_t RecordingSampleRate() const;
int32_t PlayoutSampleRate() const;
- int32_t SetRecordingChannels(size_t channels);
- int32_t SetPlayoutChannels(size_t channels);
+ virtual int32_t SetRecordingChannels(size_t channels);
+ virtual 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* audio_buffer,
- size_t num_samples);
+ virtual int32_t SetRecordedBuffer(const void* audioBuffer, size_t nSamples);
int32_t SetCurrentMicLevel(uint32_t level);
- virtual void SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift);
+ virtual void SetVQEData(int playDelayMS, int recDelayMS, int clockDrift);
virtual int32_t DeliverRecordedData();
uint32_t NewMicLevel() const;
- virtual int32_t RequestPlayoutData(size_t num_samples);
- virtual int32_t GetPlayoutData(void* audio_buffer);
+ virtual int32_t RequestPlayoutData(size_t nSamples);
+ virtual int32_t GetPlayoutData(void* audioBuffer);
- // 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 typing_status);
+ int32_t SetTypingStatus(bool typingStatus);
private:
- void AllocatePlayoutBufferIfNeeded();
- void AllocateRecordingBufferIfNeeded();
-
// Posts the first delayed task in the task queue and starts the periodic
// timer.
void StartTimer();
@@ -91,14 +86,10 @@
// 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
@@ -108,50 +99,45 @@
// Ensures that the timer is only started once.
bool timer_has_started_;
- // Sample rate in Hertz.
- uint32_t rec_sample_rate_;
- uint32_t play_sample_rate_;
+ uint32_t _recSampleRate;
+ uint32_t _playSampleRate;
- // Number of audio channels.
- size_t rec_channels_;
- size_t play_channels_;
+ size_t _recChannels;
+ size_t _playChannels;
// selected recording channel (left/right/both)
- AudioDeviceModule::ChannelType rec_channel_;
+ AudioDeviceModule::ChannelType _recChannel;
- // Number of bytes per audio sample (2 or 4).
- size_t rec_bytes_per_sample_;
- size_t play_bytes_per_sample_;
+ // 2 or 4 depending on mono or stereo
+ size_t _recBytesPerSample;
+ size_t _playBytesPerSample;
- // 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 _recBuffer[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 _recSamples;
+ size_t _recSize; // 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_;
+ // 10ms in stereo @ 96kHz
+ int8_t _playBuffer[kMaxBufferSizeBytes];
- // AGC parameters.
- uint32_t current_mic_level_;
- uint32_t new_mic_level_;
+ // one sample <=> 2 or 4 bytes
+ size_t _playSamples;
+ size_t _playSize; // in bytes
- // Contains true of a key-press has been detected.
- bool typing_status_;
+ FileWrapper& _recFile;
+ FileWrapper& _playFile;
- // Delay values used by the AEC.
- int play_delay_ms_;
- int rec_delay_ms_;
+ uint32_t _currentMicLevel;
+ uint32_t _newMicLevel;
- // Contains a clock-drift measurement.
- int clock_drift_;
+ bool _typingStatus;
+
+ int _playDelayMS;
+ int _recDelayMS;
+ int _clockDrift;
+ int high_delay_counter_;
// Counts number of times LogStats() has been called.
size_t num_stat_reports_;
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