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Added support for logging the SSRC corresponding to AudioPlayout events.

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To do this, the logging of this event was moved from the ACM to
VoiceEngine Channel. A new LogAudioPlayoutEvent function was added on
the RtcEventLog interface, and the LogDebugEvent function was removed
since it is no longer being used.

BUG=webrtc:4741
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, kwiberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/1340283002 .

Cr-Commit-Position: refs/heads/master@{#9972}
This commit is contained in:
Ivo Creusen
2015-09-17 16:30:16 +02:00
parent 48c46dbad2
commit ae856f2c9f
9 changed files with 34 additions and 46 deletions
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8
webrtc/modules/audio_coding/main/interface/audio_coding_module.h
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@ -29,7 +29,6 @@ struct WebRtcRTPHeader;
class AudioDecoder;
class AudioEncoder;
class AudioFrame;
class RtcEventLog;
class RTPFragmentationHeader;
#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
@ -83,16 +82,11 @@ class AudioCodingModule {
public:
struct Config {
Config()
: id(0),
neteq_config(),
clock(Clock::GetRealTimeClock()),
event_log(nullptr) {}
Config() : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {}
int id;
NetEq::Config neteq_config;
Clock* clock;
RtcEventLog* event_log;
};
///////////////////////////////////////////////////////////////////////////