Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(238)

Unified Diff: webrtc/voice_engine/include/voe_base.h

Issue 2307533004: Moving/renaming webrtc/common.h. (Closed)
Patch Set: rebase Created 4 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
« no previous file with comments | « webrtc/voice_engine/channel_manager.cc ('k') | webrtc/voice_engine/shared_data.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/voice_engine/include/voe_base.h
diff --git a/webrtc/voice_engine/include/voe_base.h b/webrtc/voice_engine/include/voe_base.h
index 91df7d3e4f2c8dd688c2077da24acc0e48cf039c..751394e8d7bd023a967e34be064b5aa2b3522520 100644
--- a/webrtc/voice_engine/include/voe_base.h
+++ b/webrtc/voice_engine/include/voe_base.h
@@ -36,6 +36,7 @@
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/common_types.h"
namespace webrtc {
@@ -43,9 +44,6 @@ namespace webrtc {
class AudioDeviceModule;
class AudioProcessing;
class AudioTransport;
-class Config;
-
-const int kVoEDefault = -1;
// VoiceEngineObserver
class WEBRTC_DLLEXPORT VoiceEngineObserver {
@@ -65,7 +63,6 @@ class WEBRTC_DLLEXPORT VoiceEngine {
// Creates a VoiceEngine object, which can then be used to acquire
// sub-APIs. Returns NULL on failure.
static VoiceEngine* Create();
- static VoiceEngine* Create(const Config& config);
// Deletes a created VoiceEngine object and releases the utilized resources.
// Note that if there are outstanding references held via other interfaces,
@@ -99,6 +96,11 @@ class WEBRTC_DLLEXPORT VoiceEngine {
// VoEBase
class WEBRTC_DLLEXPORT VoEBase {
public:
+ struct ChannelConfig {
+ AudioCodingModule::Config acm_config;
+ bool enable_voice_pacing = false;
+ };
+
// Factory for the VoEBase sub-API. Increases an internal reference
// counter if successful. Returns NULL if the API is not supported or if
// construction fails.
@@ -146,11 +148,13 @@ class WEBRTC_DLLEXPORT VoEBase {
virtual int Terminate() = 0;
// Creates a new channel and allocates the required resources for it.
- // One can use |config| to configure the channel. Currently that is used for
- // choosing between ACM1 and ACM2, when creating Audio Coding Module.
+ // The second version accepts a |config| struct which includes an Audio Coding
+ // Module config and an option to enable voice pacing. Note that the
+ // decoder_factory member of the ACM config will be ignored (the decoder
+ // factory set through Init() will always be used).
// Returns channel ID or -1 in case of an error.
virtual int CreateChannel() = 0;
- virtual int CreateChannel(const Config& config) = 0;
+ virtual int CreateChannel(const ChannelConfig& config) = 0;
// Deletes an existing channel and releases the utilized resources.
// Returns -1 in case of an error, 0 otherwise.
« no previous file with comments | « webrtc/voice_engine/channel_manager.cc ('k') | webrtc/voice_engine/shared_data.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698