Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1209)

Unified Diff: webrtc/modules/audio_coding/codecs/audio_encoder.cc

Issue 1322973004: Fold AudioEncoderMutable into AudioEncoder (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: review fixes Created 5 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/modules/audio_coding/codecs/audio_encoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 72e4265e987bda014badf8c7f615686731542c64..eb553a7e7d1017d49bd5d3e2301452b3d803e4c0 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -13,10 +13,12 @@
namespace webrtc {
-AudioEncoder::EncodedInfo::EncodedInfo() : EncodedInfoLeaf() {
-}
+AudioEncoder::EncodedInfo::EncodedInfo() = default;
+
+AudioEncoder::EncodedInfo::~EncodedInfo() = default;
-AudioEncoder::EncodedInfo::~EncodedInfo() {
+int AudioEncoder::RtpTimestampRateHz() const {
+ return SampleRateHz();
}
AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
@@ -32,8 +34,28 @@ AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
return info;
}
-int AudioEncoder::RtpTimestampRateHz() const {
- return SampleRateHz();
+bool AudioEncoder::SetFec(bool enable) {
+ return !enable;
+}
+
+bool AudioEncoder::SetDtx(bool enable) {
+ return !enable;
}
+bool AudioEncoder::SetApplication(Application application) {
+ return false;
+}
+
+bool AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {
+ return true;
+}
+
+void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
+
+void AudioEncoder::SetTargetBitrate(int target_bps) {}
+
+void AudioEncoder::SetMaxBitrate(int max_bps) {}
+
+void AudioEncoder::SetMaxPayloadSize(int max_payload_size_bytes) {}
+
} // namespace webrtc
« no previous file with comments | « webrtc/modules/audio_coding/codecs/audio_encoder.h ('k') | webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698