Delete encoder caching in WebRtcVideoSendStream.
This is a followup to https://webrtc-review.googlesource.com/61640,
which ensures that picture id and tl0 pic idx are continuous,
independent of how the encoder objects are created and destroyed.
The plan is to later move responsibility for encoder creation to
VideoSendStream::ReconfigureVideoEncoder, delegating work to
VideoStreamEncoder.
Bug: webrtc:8830
Change-Id: Idde5c91f24d3c0e3fa6a3bb26eb06f6800896a28
Reviewed-on: https://webrtc-review.googlesource.com/62082
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22473}diff --git a/media/engine/webrtcvideoengine.cc b/media/engine/webrtcvideoengine.cc
index d0516ee..bc90c8f 100644
--- a/media/engine/webrtcvideoengine.cc
+++ b/media/engine/webrtcvideoengine.cc
@@ -1619,8 +1619,7 @@
? webrtc::RtcpMode::kReducedSize
: webrtc::RtcpMode::kCompound;
if (codec_settings) {
- bool force_encoder_allocation = false;
- SetCodec(*codec_settings, force_encoder_allocation);
+ SetCodec(*codec_settings);
}
}
@@ -1651,8 +1650,7 @@
// If screen content settings change, we may need to recreate the codec
// instance so that the correct type is used.
- bool force_encoder_allocation = true;
- SetCodec(*parameters_.codec_settings, force_encoder_allocation);
+ SetCodec(*parameters_.codec_settings);
// Mark screenshare parameter as being updated, then test for any other
// changes that may require codec reconfiguration.
old_options.is_screencast = options->is_screencast;
@@ -1701,33 +1699,27 @@
}
void WebRtcVideoChannel::WebRtcVideoSendStream::SetCodec(
- const VideoCodecSettings& codec_settings,
- bool force_encoder_allocation) {
+ const VideoCodecSettings& codec_settings) {
RTC_DCHECK_RUN_ON(&thread_checker_);
parameters_.encoder_config = CreateVideoEncoderConfig(codec_settings.codec);
RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0);
- // Do not re-create encoders of the same type. We can't overwrite
- // |allocated_encoder_| immediately, because we need to release it after the
- // RecreateWebRtcStream() call.
- std::unique_ptr<webrtc::VideoEncoder> new_encoder;
- if (force_encoder_allocation || !allocated_encoder_ ||
- allocated_codec_ != codec_settings.codec) {
- const webrtc::SdpVideoFormat format(codec_settings.codec.name,
- codec_settings.codec.params);
- new_encoder = encoder_factory_->CreateVideoEncoder(format);
+ const webrtc::SdpVideoFormat format(codec_settings.codec.name,
+ codec_settings.codec.params);
+ // We can't overwrite |allocated_encoder_| immediately, because we
+ // need to release it after the RecreateWebRtcStream() call.
+ std::unique_ptr<webrtc::VideoEncoder> new_encoder =
+ encoder_factory_->CreateVideoEncoder(format);
- parameters_.config.encoder_settings.encoder = new_encoder.get();
+ parameters_.config.encoder_settings.encoder = new_encoder.get();
- const webrtc::VideoEncoderFactory::CodecInfo info =
- encoder_factory_->QueryVideoEncoder(format);
- parameters_.config.encoder_settings.full_overuse_time =
- info.is_hardware_accelerated;
- parameters_.config.encoder_settings.internal_source =
- info.has_internal_source;
- } else {
- new_encoder = std::move(allocated_encoder_);
- }
+ const webrtc::VideoEncoderFactory::CodecInfo info =
+ encoder_factory_->QueryVideoEncoder(format);
+ parameters_.config.encoder_settings.full_overuse_time =
+ info.is_hardware_accelerated;
+ parameters_.config.encoder_settings.internal_source =
+ info.has_internal_source;
+
parameters_.config.encoder_settings.payload_name = codec_settings.codec.name;
parameters_.config.encoder_settings.payload_type = codec_settings.codec.id;
parameters_.config.rtp.ulpfec = codec_settings.ulpfec;
@@ -1754,7 +1746,6 @@
RTC_LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetCodec.";
RecreateWebRtcStream();
allocated_encoder_ = std::move(new_encoder);
- allocated_codec_ = codec_settings.codec;
}
void WebRtcVideoChannel::WebRtcVideoSendStream::SetSendParameters(
@@ -1781,12 +1772,10 @@
// Set codecs and options.
if (params.codec) {
- bool force_encoder_allocation = false;
- SetCodec(*params.codec, force_encoder_allocation);
+ SetCodec(*params.codec);
recreate_stream = false; // SetCodec has already recreated the stream.
} else if (params.conference_mode && parameters_.codec_settings) {
- bool force_encoder_allocation = false;
- SetCodec(*parameters_.codec_settings, force_encoder_allocation);
+ SetCodec(*parameters_.codec_settings);
recreate_stream = false; // SetCodec has already recreated the stream.
}
if (recreate_stream) {
diff --git a/media/engine/webrtcvideoengine.h b/media/engine/webrtcvideoengine.h
index f481b8b..04096e6 100644
--- a/media/engine/webrtcvideoengine.h
+++ b/media/engine/webrtcvideoengine.h
@@ -312,8 +312,7 @@
rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
ConfigureVideoEncoderSettings(const VideoCodec& codec);
- void SetCodec(const VideoCodecSettings& codec,
- bool force_encoder_allocation);
+ void SetCodec(const VideoCodecSettings& codec);
void RecreateWebRtcStream();
webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
const VideoCodec& codec) const;
@@ -355,7 +354,6 @@
webrtc::RtpParameters rtp_parameters_ RTC_GUARDED_BY(&thread_checker_);
std::unique_ptr<webrtc::VideoEncoder> allocated_encoder_
RTC_GUARDED_BY(&thread_checker_);
- VideoCodec allocated_codec_ RTC_GUARDED_BY(&thread_checker_);
bool sending_ RTC_GUARDED_BY(&thread_checker_);
};