fix build with latest webrtc sdk

mpromonet · Sep 7, 2024 · d97b7ad · d97b7ad
1 parent 4cfa382
commit d97b7ad
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Showing 4 changed files with 9 additions and 7 deletions.
diff --git a/inc/EncodedVideoFrameBuffer.h b/inc/EncodedVideoFrameBuffer.h
@@ -28,7 +28,6 @@ class EncodedVideoFrameBuffer : public webrtc::VideoFrameBuffer
   	webrtc::EncodedImage encoded_image;
 		encoded_image.SetEncodedData(webrtc::EncodedImageBuffer::Create(m_encoded_data->data(), m_encoded_data->size()));
 		encoded_image._frameType = m_frameType;
-    encoded_image.SetAtTargetQuality(true);
 		encoded_image.SetRtpTimestamp(rtptime);
 		encoded_image.ntp_time_ms_ = ntptime;
     return encoded_image;

diff --git a/inc/liveaudiosource.h b/inc/liveaudiosource.h
@@ -22,6 +22,7 @@
 #include "environment.h"
 
 #include "pc/local_audio_source.h"
+#include "api/environment/environment_factory.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 
 template <typename T>
@@ -70,7 +71,7 @@ class LiveAudioSource : public webrtc::Notifier<webrtc::AudioSourceInterface>, p
             webrtc::SdpAudioFormat format = webrtc::SdpAudioFormat(codec, m_freq, m_channel, std::move(params));
             if (m_factory->IsSupportedDecoder(format))
             {
-                m_decoder = m_factory->MakeAudioDecoder(format, absl::optional<webrtc::AudioCodecPairId>());
+                m_decoder = m_factory->Create(m_webrtcenv, format, std::optional<webrtc::AudioCodecPairId>());
                 m_codec[id] = codec;
                 success = true;
             }
@@ -164,6 +165,7 @@ class LiveAudioSource : public webrtc::Notifier<webrtc::AudioSourceInterface>, p
     LiveAudioSource(rtc::scoped_refptr<webrtc::AudioDecoderFactory> audioDecoderFactory, const std::string &uri, const std::map<std::string, std::string> &opts, bool wait)
         : m_env(m_stop)
         , m_connection(m_env, this, uri.c_str(), opts, rtc::LogMessage::GetLogToDebug() <= 2)
+        , m_webrtcenv(webrtc::CreateEnvironment())
         , m_factory(audioDecoderFactory)
         , m_freq(8000)
         , m_channel(1)
@@ -185,6 +187,7 @@ class LiveAudioSource : public webrtc::Notifier<webrtc::AudioSourceInterface>, p
 
 private:
     T m_connection;
+    const webrtc::Environment                     m_webrtcenv;
     std::thread m_capturethread;
     rtc::scoped_refptr<webrtc::AudioDecoderFactory> m_factory;
     std::unique_ptr<webrtc::AudioDecoder> m_decoder;

diff --git a/inc/livevideosource.h b/inc/livevideosource.h
@@ -115,7 +115,7 @@ class LiveVideoSource : public VideoDecoder, public T::Callback
                 m_cfg.clear();
                 m_cfg.insert(m_cfg.end(), buffer + index.start_offset, buffer + index.payload_size + index.payload_start_offset);
 
-                absl::optional<webrtc::SpsParser::SpsState> sps = webrtc::SpsParser::ParseSps(buffer + index.payload_start_offset + webrtc::H264::kNaluTypeSize, index.payload_size - webrtc::H264::kNaluTypeSize);
+                std::optional<webrtc::SpsParser::SpsState> sps = webrtc::SpsParser::ParseSps(buffer + index.payload_start_offset + webrtc::H264::kNaluTypeSize, index.payload_size - webrtc::H264::kNaluTypeSize);
                 if (!sps)
                 {
                     RTC_LOG(LS_ERROR) << "cannot parse sps";
@@ -179,7 +179,7 @@ class LiveVideoSource : public VideoDecoder, public T::Callback
                 RTC_LOG(LS_VERBOSE) << "LiveVideoSource:onData SPS";
                 m_cfg.insert(m_cfg.end(), buffer + index.start_offset, buffer + index.payload_size + index.payload_start_offset);
 
-                absl::optional<webrtc::H265SpsParser::SpsState> sps = webrtc::H265SpsParser::ParseSps(buffer + index.payload_start_offset + webrtc::H265::kNaluHeaderSize, index.payload_size - webrtc::H265::kNaluHeaderSize);
+                std::optional<webrtc::H265SpsParser::SpsState> sps = webrtc::H265SpsParser::ParseSps(buffer + index.payload_start_offset + webrtc::H265::kNaluHeaderSize, index.payload_size - webrtc::H265::kNaluHeaderSize);
                 if (!sps)
                 {
                     RTC_LOG(LS_ERROR) << "cannot parse sps";

diff --git a/src/PeerConnectionManager.cpp b/src/PeerConnectionManager.cpp
@@ -1257,9 +1257,9 @@ bool PeerConnectionManager::AddStreams(webrtc::PeerConnectionInterface *peer_con
 		int bitrate = std::stoi(opts.at("bitrate"));
 
 		webrtc::BitrateSettings bitrateParam;
-		bitrateParam.min_bitrate_bps = absl::optional<int>(bitrate / 2);
-		bitrateParam.start_bitrate_bps = absl::optional<int>(bitrate);
-		bitrateParam.max_bitrate_bps = absl::optional<int>(bitrate * 2);
+		bitrateParam.min_bitrate_bps = std::optional<int>(bitrate / 2);
+		bitrateParam.start_bitrate_bps = std::optional<int>(bitrate);
+		bitrateParam.max_bitrate_bps = std::optional<int>(bitrate * 2);
 		peer_connection->SetBitrate(bitrateParam);
 
 		RTC_LOG(LS_WARNING) << "set bitrate:" << bitrate;