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( |