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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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4
webrtc/modules/audio_device/include/audio_device_defines.h
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@ -71,7 +71,7 @@ public:
const uint32_t samplesPerSec,
void* audioSamples,
uint32_t& nSamplesOut,
uint32_t* rtp_timestamp,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) = 0;
// Method to pass captured data directly and unmixed to network channels.
@ -128,7 +128,7 @@ public:
virtual void PullRenderData(int bits_per_sample, int sample_rate,
int number_of_channels, int number_of_frames,
void* audio_data,
uint32_t* rtp_timestamp,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {}
protected: