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diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -12,24 +12,16 @@
#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_device/audio_device_config.h"
#include "webrtc/modules/audio_device/linux/audio_device_alsa_linux.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/system_wrappers/include/trace.h"
-
-#include "Latency.h"
-
-#define LOG_FIRST_CAPTURE(x) LogTime(AsyncLatencyLogger::AudioCaptureBase, \
- reinterpret_cast<uint64_t>(x), 0)
-#define LOG_CAPTURE_FRAMES(x, frames) LogLatency(AsyncLatencyLogger::AudioCapture, \
- reinterpret_cast<uint64_t>(x), frames)
-
webrtc_adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable;
// Accesses ALSA functions through our late-binding symbol table instead of
// directly. This way we don't have to link to libasound, which means our binary
// will work on systems that don't have it.
#define LATE(sym) \
LATESYM_GET(webrtc_adm_linux_alsa::AlsaSymbolTable, &AlsaSymbolTable, sym)
@@ -2138,20 +2130,18 @@ bool AudioDeviceLinuxALSA::RecThreadProc
buffer, size);
_recordingFramesLeft -= frames;
if (!_recordingFramesLeft)
{ // buf is full
_recordingFramesLeft = _recordingFramesIn10MS;
if (_firstRecord) {
- LOG_FIRST_CAPTURE(this);
_firstRecord = false;
}
- LOG_CAPTURE_FRAMES(this, _recordingFramesIn10MS);
// store the recorded buffer (no action will be taken if the
// #recorded samples is not a full buffer)
_ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
_recordingFramesIn10MS);
uint32_t currentMicLevel = 0;
uint32_t newMicLevel = 0;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
--- a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
@@ -13,22 +13,16 @@
#include "webrtc/modules/audio_device/audio_device_config.h"
#include "webrtc/modules/audio_device/sndio/audio_device_sndio.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/system_wrappers/include/trace.h"
-#include "Latency.h"
-
-#define LOG_FIRST_CAPTURE(x) LogTime(AsyncLatencyLogger::AudioCaptureBase, \
- reinterpret_cast<uint64_t>(x), 0)
-#define LOG_CAPTURE_FRAMES(x, frames) LogLatency(AsyncLatencyLogger::AudioCapture, \
- reinterpret_cast<uint64_t>(x), frames)
extern "C"
{
static void playOnmove(void *arg, int delta)
{
static_cast<webrtc::AudioDeviceSndio *>(arg)->_playDelay -= delta;
}
static void recOnmove(void *arg, int delta)
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