Index: webrtc/modules/audio_coding/codecs/audio_decoder.h |
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.h b/webrtc/modules/audio_coding/codecs/audio_decoder.h |
index b6338d2102a490d7c7272def3a62691daebef6eb..af16095117fe2e7a08f47d045343f8c9a91314e8 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.h |
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.h |
@@ -66,7 +66,7 @@ class AudioDecoder { |
struct ParseResult { |
ParseResult(); |
ParseResult(uint32_t timestamp, |
- bool primary, |
+ int priority, |
std::unique_ptr<EncodedAudioFrame> frame); |
ParseResult(ParseResult&& b); |
~ParseResult(); |
@@ -75,7 +75,9 @@ class AudioDecoder { |
// The timestamp of the frame is in samples per channel. |
uint32_t timestamp; |
- bool primary; |
+ // The relative priority of the frame compared to other frames of the same |
+ // payload and the same timeframe. A higher value means a lower priority. |
+ int priority; |
std::unique_ptr<EncodedAudioFrame> frame; |
}; |
@@ -86,8 +88,7 @@ class AudioDecoder { |
// buffer. |timestamp| is the input timestamp, in samples, corresponding to |
// the start of the payload. |
virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, |
- uint32_t timestamp, |
- bool is_primary); |
+ uint32_t timestamp); |
// Decodes |encode_len| bytes from |encoded| and writes the result in |
// |decoded|. The maximum bytes allowed to be written into |decoded| is |