Index: webrtc/api/call/audio_send_stream.h |
diff --git a/webrtc/api/call/audio_send_stream.h b/webrtc/api/call/audio_send_stream.h |
index 7ff791e62ad307782382b1243429f971733bdb0f..1956b97bd42e4ea67bc82cfa7412a16fc63f0157 100644 |
--- a/webrtc/api/call/audio_send_stream.h |
+++ b/webrtc/api/call/audio_send_stream.h |
@@ -30,8 +30,6 @@ |
class AudioSendStream { |
public: |
struct Stats { |
- Stats(); |
- |
// TODO(solenberg): Harmonize naming and defaults with receive stream stats. |
uint32_t local_ssrc = 0; |
int64_t bytes_sent = 0; |
@@ -54,13 +52,13 @@ |
struct Config { |
Config() = delete; |
- explicit Config(Transport* send_transport); |
+ explicit Config(Transport* send_transport) |
+ : send_transport(send_transport) {} |
+ |
std::string ToString() const; |
// Send-stream specific RTP settings. |
struct Rtp { |
- Rtp(); |
- ~Rtp(); |
std::string ToString() const; |
// Sender SSRC. |
@@ -93,10 +91,40 @@ |
int max_bitrate_kbps = -1; |
struct SendCodecSpec { |
- SendCodecSpec(); |
- std::string ToString() const; |
- |
- bool operator==(const SendCodecSpec& rhs) const; |
+ SendCodecSpec() { |
+ webrtc::CodecInst empty_inst = {0}; |
+ codec_inst = empty_inst; |
+ codec_inst.pltype = -1; |
+ } |
+ bool operator==(const SendCodecSpec& rhs) const { |
+ { |
+ if (nack_enabled != rhs.nack_enabled) { |
+ return false; |
+ } |
+ if (transport_cc_enabled != rhs.transport_cc_enabled) { |
+ return false; |
+ } |
+ if (enable_codec_fec != rhs.enable_codec_fec) { |
+ return false; |
+ } |
+ if (enable_opus_dtx != rhs.enable_opus_dtx) { |
+ return false; |
+ } |
+ if (opus_max_playback_rate != rhs.opus_max_playback_rate) { |
+ return false; |
+ } |
+ if (cng_payload_type != rhs.cng_payload_type) { |
+ return false; |
+ } |
+ if (cng_plfreq != rhs.cng_plfreq) { |
+ return false; |
+ } |
+ if (codec_inst != rhs.codec_inst) { |
+ return false; |
+ } |
+ return true; |
+ } |
+ } |
bool operator!=(const SendCodecSpec& rhs) const { |
return !(*this == rhs); |
} |