(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:
parent
b92f6f9371
commit
d8524348bb
@ -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
|
||||
|
@ -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;
|
||||
}
|
||||
|
||||
|
@ -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);
|
||||
|
@ -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;
|
||||
|
@ -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);
|
||||
|
@ -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) {
|
||||
|
Loading…
Reference in New Issue
Block a user