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Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t>

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This is in preparation for changes to when the playout timestamp is
valid.

BUG=webrtc:5669

Review URL: https://codereview.webrtc.org/1853183002

Cr-Commit-Position: refs/heads/master@{#12256}
This commit is contained in:
henrik.lundin
2016-04-06 01:39:22 -07:00
committed by Commit bot
parent 05255b0e8a
commit 9a410dd082
13 changed files with 77 additions and 58 deletions
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14
webrtc/modules/audio_coding/include/audio_coding_module.h
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@ -14,6 +14,7 @@
#include <string>
#include <vector>
#include "webrtc/base/deprecation.h"
#include "webrtc/base/optional.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
@ -647,7 +648,6 @@ class AudioCodingModule {
//
virtual int LeastRequiredDelayMs() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutTimestamp()
// The send timestamp of an RTP packet is associated with the decoded
// audio of the packet in question. This function returns the timestamp of
@ -660,8 +660,16 @@ class AudioCodingModule {
// 0 if the output is a correct timestamp.
// -1 if failed to output the correct timestamp.
//
// TODO(tlegrand): Change function to return the timestamp.
virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0;
RTC_DEPRECATED virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutTimestamp()
// The send timestamp of an RTP packet is associated with the decoded
// audio of the packet in question. This function returns the timestamp of
// the latest audio obtained by calling PlayoutData10ms(), or empty if no
// valid timestamp is available.
//
virtual rtc::Optional<uint32_t> PlayoutTimestamp() = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutData10Ms(