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Fix the chain that propagates the audio frame's rtp and ntp timestamp including:

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* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.

Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.

Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.

BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc

Review URL: https://webrtc-codereview.appspot.com/14559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org
2014-06-05 20:34:08 +00:00
parent 130fa64d4c
commit 94454b71ad
26 changed files with 168 additions and 100 deletions
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6
webrtc/modules/audio_device/audio_device_buffer.cc
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@ -548,15 +548,15 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(uint32_t nSamples)
if (_ptrCbAudioTransport)
{
uint32_t res(0);
uint32_t rtp_timestamp = 0;
int64_t ntp_time_ms = 0;
int64_t elapsed_time_ms = -1;
int64_t ntp_time_ms = -1;
res = _ptrCbAudioTransport->NeedMorePlayData(_playSamples,
playBytesPerSample,
playChannels,
playSampleRate,
&_playBuffer[0],
nSamplesOut,
&rtp_timestamp,
&elapsed_time_ms,
&ntp_time_ms);
if (res != 0)
{