(Auto)update libjingle 71107853-> 71115715

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6675 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-07-14 20:05:09 +00:00
parent b92f6f9371
commit d8524348bb
6 changed files with 12 additions and 20 deletions

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@ -730,20 +730,20 @@ void FakeAudioCaptureModule::ReceiveFrameP() {
uint32_t nSamplesOut = 0;
#ifdef USE_WEBRTC_DEV_BRANCH
int64_t elapsed_time_ms = 0;
#else
uint32_t rtp_timestamp = 0;
#endif
int64_t ntp_time_ms = 0;
if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample,
kNumberOfChannels, kSamplesPerSecond,
rec_buffer_, nSamplesOut,
#ifdef USE_WEBRTC_DEV_BRANCH
&elapsed_time_ms, &ntp_time_ms) != 0) {
ASSERT(false);
}
#else
if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample,
kNumberOfChannels, kSamplesPerSecond,
rec_buffer_, nSamplesOut) != 0) {
&rtp_timestamp, &ntp_time_ms) != 0) {
#endif
ASSERT(false);
}
#endif
ASSERT(nSamplesOut == kNumberSamples);
}
// The SetBuffer() function ensures that after decoding, the audio buffer

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@ -84,13 +84,13 @@ class FakeAdmTest : public testing::Test,
const uint8_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
#ifdef USE_WEBRTC_DEV_BRANCH
uint32_t& nSamplesOut,
#ifdef USE_WEBRTC_DEV_BRANCH
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
#else
uint32_t& nSamplesOut) {
uint32_t* rtp_timestamp,
#endif
int64_t* ntp_time_ms) {
++pull_iterations_;
const uint32_t audio_buffer_size = nSamples * nBytesPerSample;
const uint32_t bytes_out = RecordedDataReceived() ?
@ -99,8 +99,10 @@ class FakeAdmTest : public testing::Test,
nSamplesOut = bytes_out / nBytesPerSample;
#ifdef USE_WEBRTC_DEV_BRANCH
*elapsed_time_ms = 0;
*ntp_time_ms = 0;
#else
*rtp_timestamp = 0;
#endif
*ntp_time_ms = 0;
return 0;
}

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@ -789,7 +789,6 @@ class FakeWebRtcVoiceEngine
channels_[channel]->send_audio_level_ext_ = (enable) ? id : -1;
return 0;
}
#ifdef USE_WEBRTC_DEV_BRANCH
WEBRTC_FUNC(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable,
unsigned char id)) {
WEBRTC_CHECK_CHANNEL(channel);
@ -797,7 +796,6 @@ class FakeWebRtcVoiceEngine
channels_[channel]->receive_audio_level_ext_ = (enable) ? id : -1;
return 0;
}
#endif // USE_WEBRTC_DEV_BRANCH
WEBRTC_FUNC(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
unsigned char id)) {
WEBRTC_CHECK_CHANNEL(channel);

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@ -211,9 +211,7 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
virtual int DeliverFrame(unsigned char* buffer,
int buffer_size,
uint32_t rtp_time_stamp,
#ifdef USE_WEBRTC_DEV_BRANCH
int64_t ntp_time_ms,
#endif
int64_t render_time,
void* handle) {
talk_base::CritScope cs(&crit_);
@ -226,11 +224,9 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
int64 elapsed_time_ms =
(rtp_ts_wraparound_handler_.Unwrap(rtp_time_stamp) -
capture_start_rtp_time_stamp_) / kVideoCodecClockratekHz;
#ifdef USE_WEBRTC_DEV_BRANCH
if (ntp_time_ms > 0) {
capture_start_ntp_time_ms_ = ntp_time_ms - elapsed_time_ms;
}
#endif
frame_rate_tracker_.Update(1);
if (renderer_ == NULL) {
return 0;

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@ -2280,7 +2280,6 @@ bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions(
bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions(
int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
#ifdef USE_WEBRTC_DEV_BRANCH
const RtpHeaderExtension* audio_level_extension =
FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
if (!SetHeaderExtension(
@ -2288,7 +2287,6 @@ bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions(
audio_level_extension)) {
return false;
}
#endif // USE_WEBRTC_DEV_BRANCH
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);

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@ -1692,11 +1692,9 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED5) {
TEST_F(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) {
TestSetSendRtpHeaderExtensions(kRtpAudioLevelHeaderExtension);
}
#ifdef USE_WEBRTC_DEV_BRANCH
TEST_F(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(kRtpAudioLevelHeaderExtension);
}
#endif // USE_WEBRTC_DEV_BRANCH
// Test support for absolute send time header extension.
TEST_F(WebRtcVoiceEngineTestFake, SendAbsoluteSendTimeHeaderExtensions) {