Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1722)

Unified Diff: webrtc/modules/audio_coding/neteq/include/neteq.h

Issue 1769883002: Remove the type parameter to NetEq::GetAudio (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@acm-rec-delete-vad
Patch Set: After review Created 4 years, 9 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/neteq/include/neteq.h
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index dff09db3db9ae568751c0fd5a4a83ee5958edb65..d53551f89798202500439864f7709d997a6e2648 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -54,14 +54,6 @@ struct NetEqNetworkStatistics {
int max_waiting_time_ms;
};
-enum NetEqOutputType {
- kOutputNormal,
- kOutputPLC,
- kOutputCNG,
- kOutputPLCtoCNG,
- kOutputVADPassive
-};
-
enum NetEqPlayoutMode {
kPlayoutOn,
kPlayoutOff,
@@ -165,11 +157,11 @@ class NetEq {
// Instructs NetEq to deliver 10 ms of audio data. The data is written to
// |audio_frame|. All data in |audio_frame| is wiped; |data_|, |interleaved_|,
- // |num_channels_|, and |samples_per_channel_| are updated upon success. If
- // an error is returned, some fields may not have been updated.
- // The speech type is written to |type|, if |type| is not NULL.
+ // |num_channels_|, |samples_per_channel_|, |speech_type_|, and
+ // |vad_activity_| are updated upon success. If an error is returned, some
+ // fields may not have been updated.
// Returns kOK on success, or kFail in case of an error.
- virtual int GetAudio(AudioFrame* audio_frame, NetEqOutputType* type) = 0;
+ virtual int GetAudio(AudioFrame* audio_frame) = 0;
// Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
// information in the codec database. Returns 0 on success, -1 on failure.
« no previous file with comments | « webrtc/modules/audio_coding/acm2/acm_receiver.cc ('k') | webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698