Index: webrtc/call/audio_send_stream.h |
diff --git a/webrtc/call/audio_send_stream.h b/webrtc/call/audio_send_stream.h |
index 42914301911968878d4559a91ac7336c7d622c47..869f7524e655a6ee2bd91f40c64080aa2a939f7c 100644 |
--- a/webrtc/call/audio_send_stream.h |
+++ b/webrtc/call/audio_send_stream.h |
@@ -15,10 +15,11 @@ |
#include <string> |
#include <vector> |
+#include "webrtc/api/audio_codecs/audio_format.h" |
#include "webrtc/api/call/transport.h" |
#include "webrtc/base/optional.h" |
#include "webrtc/config.h" |
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
+#include "webrtc/modules/audio_coding/codecs/audio_encoder_factory.h" |
#include "webrtc/typedefs.h" |
namespace webrtc { |
@@ -103,6 +104,7 @@ class AudioSendStream { |
struct SendCodecSpec { |
SendCodecSpec(); |
+ ~SendCodecSpec(); |
std::string ToString() const; |
bool operator==(const SendCodecSpec& rhs) const; |
@@ -112,15 +114,13 @@ class AudioSendStream { |
bool nack_enabled = false; |
bool transport_cc_enabled = false; |
- bool enable_codec_fec = false; |
- bool enable_opus_dtx = false; |
- int opus_max_playback_rate = 0; |
int cng_payload_type = -1; |
- int cng_plfreq = -1; |
- int max_ptime_ms = -1; |
- int min_ptime_ms = -1; |
- webrtc::CodecInst codec_inst; |
+ int payload_type; |
the sun
2017/03/16 08:48:19
default init
ossu
2017/03/20 18:19:48
It's now required by the constructor.
|
+ SdpAudioFormat format; |
+ rtc::Optional<int> target_bitrate_bps; |
kwiberg-webrtc
2017/03/01 12:26:47
Does an unset value mean default bitrate?
ossu
2017/03/02 01:30:28
Yes. I'll add a comment clarifying that.
|
} send_codec_spec; |
+ |
+ rtc::scoped_refptr<AudioEncoderFactory> encoder_factory; |
}; |
// Starts stream activity. |