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..902bec940592cb9b1e6c316c4cc6e90567085f60 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 +114,14 @@ 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; |
+ rtc::Optional<int> target_bitrate_bps; |
+ webrtc::AudioCodecSpec format; |
ossu
2017/02/21 11:04:14
This isn't very nice. It leads to us having to ref
|
} send_codec_spec; |
+ |
+ rtc::scoped_refptr<AudioEncoderFactory> encoder_factory; |
}; |
// Starts stream activity. |