Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time. Bug: webrtc:10668 Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/master@{#28434}
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@ -16,6 +16,7 @@
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#include "modules/audio_coding/neteq/tools/output_wav_file.h"
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#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
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#include "rtc_base/checks.h"
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#include "system_wrappers/include/clock.h"
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#include "test/testsupport/file_utils.h"
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namespace webrtc {
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@ -213,7 +214,8 @@ NetEqQualityTest::NetEqQualityTest(
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NetEq::Config config;
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config.sample_rate_hz = out_sampling_khz_ * 1000;
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neteq_.reset(NetEq::Create(config, decoder_factory));
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neteq_.reset(
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NetEq::Create(config, Clock::GetRealTimeClock(), decoder_factory));
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max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t);
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in_data_.reset(new int16_t[in_size_samples_ * channels_]);
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}
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