Check the mic volume only periodically on Mac.

Ask the OS for the mic volume every 1 second rather than with every 10
ms chunk. The previous behavior was consuming ~2% of the CPU load of
a voice engine call, and is now negligible.

This is consistent with the webrtc Windows Core Audio implementation,
as well as the Chromium Mac implementation:
https://code.google.com/p/chromium/codesearch#chromium/src/media/audio/agc_audio_stream.h

TEST=voe_cmd_test with AGC continues to work well on Mac.

Review URL: https://codereview.webrtc.org/1564223002

Cr-Commit-Position: refs/heads/master@{#11182}
This commit is contained in:
andrew 2016-01-08 01:16:17 -08:00 committed by Commit bot
parent fbeb97e01f
commit ec80f03b3c
2 changed files with 25 additions and 9 deletions

View File

@ -149,7 +149,8 @@ AudioDeviceMac::AudioDeviceMac(const int32_t id) :
_paRenderBuffer(NULL),
_captureBufSizeSamples(0),
_renderBufSizeSamples(0),
prev_key_state_()
prev_key_state_(),
get_mic_volume_counter_ms_(0)
{
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
"%s created", __FUNCTION__);
@ -379,6 +380,8 @@ int32_t AudioDeviceMac::Init()
_recWarning = 0;
_recError = 0;
get_mic_volume_counter_ms_ = 0;
_initialized = true;
return 0;
@ -3181,12 +3184,17 @@ bool AudioDeviceMac::CaptureWorkerThread()
if (AGC())
{
// store current mic level in the audio buffer if AGC is enabled
if (MicrophoneVolume(currentMicLevel) == 0)
{
// this call does not affect the actual microphone volume
_ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
// Use mod to ensure we check the volume on the first pass.
if (get_mic_volume_counter_ms_ % kGetMicVolumeIntervalMs == 0) {
get_mic_volume_counter_ms_ = 0;
// store current mic level in the audio buffer if AGC is enabled
if (MicrophoneVolume(currentMicLevel) == 0)
{
// this call does not affect the actual microphone volume
_ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
}
}
get_mic_volume_counter_ms_ += kBufferSizeMs;
}
_ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);

View File

@ -38,20 +38,26 @@ const uint32_t N_REC_CHANNELS = 1; // default is mono recording
const uint32_t N_PLAY_CHANNELS = 2; // default is stereo playout
const uint32_t N_DEVICE_CHANNELS = 64;
const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC / 100);
const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC / 100);
const int kBufferSizeMs = 10;
const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES =
N_REC_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES =
N_PLAY_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
const int N_BLOCKS_IO = 2;
const int N_BUFFERS_IN = 2; // Must be at least N_BLOCKS_IO.
const int N_BUFFERS_OUT = 3; // Must be at least N_BLOCKS_IO.
const uint32_t TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000);
const uint32_t TIMER_PERIOD_MS = 2 * 10 * N_BLOCKS_IO * 1000000;
const uint32_t REC_BUF_SIZE_IN_SAMPLES =
ENGINE_REC_BUF_SIZE_IN_SAMPLES * N_DEVICE_CHANNELS * N_BUFFERS_IN;
const uint32_t PLAY_BUF_SIZE_IN_SAMPLES =
ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT;
const int kGetMicVolumeIntervalMs = 1000;
class AudioDeviceMac: public AudioDeviceGeneric
{
public:
@ -372,6 +378,8 @@ private:
// Typing detection
// 0x5c is key "9", after that comes function keys.
bool prev_key_state_[0x5d];
int get_mic_volume_counter_ms_;
};
} // namespace webrtc