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..5363d130776c90366162fc50ad07b726cd58408f 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 { |
@@ -102,7 +103,8 @@ class AudioSendStream { |
rtc::Optional<std::string> audio_network_adaptor_config; |
struct SendCodecSpec { |
- SendCodecSpec(); |
+ SendCodecSpec(int payload_type, const SdpAudioFormat& format); |
+ ~SendCodecSpec(); |
std::string ToString() const; |
bool operator==(const SendCodecSpec& rhs) const; |
@@ -110,19 +112,22 @@ class AudioSendStream { |
return !(*this == rhs); |
} |
+ int payload_type; |
+ SdpAudioFormat format; |
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; |
- } send_codec_spec; |
+ rtc::Optional<int> cng_payload_type; |
+ // If unset, use the encoder's default target bitrate. |
+ rtc::Optional<int> target_bitrate_bps; |
+ }; |
+ |
+ rtc::Optional<SendCodecSpec> send_codec_spec; |
+ rtc::scoped_refptr<AudioEncoderFactory> encoder_factory; |
}; |
+ // Reconfigure the stream according to the Configuration. |
+ virtual void Reconfigure(const Config& config) = 0; |
+ |
// Starts stream activity. |
// When a stream is active, it can receive, process and deliver packets. |
virtual void Start() = 0; |