| Index: webrtc/modules/audio_coding/include/audio_coding_module.h
|
| diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
|
| index 305d8ea6d3c09acf2dc0f1098da7b0e37c86bdb2..381e35e639b27061cd83d986fe3b39f03603cdea 100644
|
| --- a/webrtc/modules/audio_coding/include/audio_coding_module.h
|
| +++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
|
| @@ -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(
|
|
|