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Unified Diff: webrtc/api/call/audio_send_stream.cc

Issue 2397573006: Using AudioOption to enable audio network adaptor. (Closed)
Patch Set: fixing a unittest Created 4 years, 2 months ago
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Index: webrtc/api/call/audio_send_stream.cc
diff --git a/webrtc/api/call/audio_send_stream.cc b/webrtc/api/call/audio_send_stream.cc
index 06cbc545d9313846b057ed3432e2862c5c8b9b14..52c30f0987b9efe23ee04821dd2dc9db4e783027 100644
--- a/webrtc/api/call/audio_send_stream.cc
+++ b/webrtc/api/call/audio_send_stream.cc
@@ -34,6 +34,8 @@ AudioSendStream::Stats::Stats() = default;
AudioSendStream::Config::Config(Transport* send_transport)
: send_transport(send_transport) {}
+AudioSendStream::Config::~Config() = default;
+
std::string AudioSendStream::Config::ToString() const {
std::stringstream ss;
ss << "{rtp: " << rtp.ToString();
@@ -82,6 +84,8 @@ std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
ss << ", opus_max_playback_rate: " << opus_max_playback_rate;
ss << ", cng_payload_type: " << cng_payload_type;
ss << ", cng_plfreq: " << cng_plfreq;
+ ss << ", min_ptime: " << min_ptime_ms;
+ ss << ", max_ptime: " << max_ptime_ms;
ss << ", codec_inst: " << ::ToString(codec_inst);
ss << '}';
return ss.str();
@@ -89,30 +93,16 @@ std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
bool AudioSendStream::Config::SendCodecSpec::operator==(
const AudioSendStream::Config::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;
+ if (nack_enabled == rhs.nack_enabled &&
+ transport_cc_enabled == rhs.transport_cc_enabled &&
+ enable_codec_fec == rhs.enable_codec_fec &&
+ enable_opus_dtx == rhs.enable_opus_dtx &&
+ opus_max_playback_rate == rhs.opus_max_playback_rate &&
+ cng_payload_type == rhs.cng_payload_type &&
+ cng_plfreq == rhs.cng_plfreq && max_ptime_ms == rhs.max_ptime_ms &&
+ min_ptime_ms == rhs.min_ptime_ms && codec_inst == rhs.codec_inst) {
+ return true;
}
- return true;
+ return false;
}
} // namespace webrtc
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