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More trace events

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The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.

BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
hclam@chromium.org
2013-04-09 19:54:10 +00:00
parent 4d2f5de67a
commit 806dc3b0e6
20 changed files with 199 additions and 97 deletions
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15
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
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@@ -1304,6 +1304,10 @@ int32_t AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
// Add 10MS of raw (PCM) audio data to the encoder.
int32_t AudioCodingModuleImpl::Add10MsData(
const AudioFrame& audio_frame) {
TRACE_EVENT2("webrtc", "ACM::Add10MsData",
"timestamp", audio_frame.timestamp_,
"samples_per_channel", audio_frame.samples_per_channel_);
if (audio_frame.samples_per_channel_ <= 0) {
assert(false);
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
@@ -2232,11 +2236,14 @@ AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const {
// Automatic resample to the requested frequency.
int32_t AudioCodingModuleImpl::PlayoutData10Ms(
int32_t desired_freq_hz, AudioFrame* audio_frame) {
TRACE_EVENT0("webrtc_voe", "ACM::PlayoutData10Ms");
TRACE_EVENT_ASYNC_BEGIN0("webrtc", "ACM::PlayoutData10Ms", 0);
bool stereo_mode;
if (GetSilence(desired_freq_hz, audio_frame))
if (GetSilence(desired_freq_hz, audio_frame)) {
TRACE_EVENT_ASYNC_END1("webrtc", "ACM::PlayoutData10Ms", 0,
"silence", true);
return 0; // Silence is generated, return.
}
// RecOut always returns 10 ms.
if (neteq_.RecOut(audio_frame_) != 0) {
@@ -2264,6 +2271,8 @@ int32_t AudioCodingModuleImpl::PlayoutData10Ms(
CriticalSectionScoped lock(acm_crit_sect_);
if ((receive_freq != desired_freq_hz) && (desired_freq_hz != -1)) {
TRACE_EVENT_ASYNC_END2("webrtc", "ACM::PlayoutData10Ms", 0,
"stereo", stereo_mode, "resample", true);
// Resample payload_data.
int16_t temp_len = output_resampler_.Resample10Msec(
audio_frame_.data_, receive_freq, audio_frame->data_,
@@ -2280,6 +2289,8 @@ int32_t AudioCodingModuleImpl::PlayoutData10Ms(
// Set the sampling frequency.
audio_frame->sample_rate_hz_ = desired_freq_hz;
} else {
TRACE_EVENT_ASYNC_END2("webrtc", "ACM::PlayoutData10Ms", 0,
"stereo", stereo_mode, "resample", false);
memcpy(audio_frame->data_, audio_frame_.data_,
audio_frame_.samples_per_channel_ * audio_frame->num_channels_
* sizeof(int16_t));