Use backticks not vertical bars to denote variables in comments for /modules/audio_coding
Bug: webrtc:12338 Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34621}
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WebRTC LUCI CQ
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@ -320,7 +320,7 @@ void TestAllCodecs::RegisterSendCodec(char side,
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// If G.722, store half the size to compensate for the timestamp bug in the
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// RFC for G.722.
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// If iSAC runs in adaptive mode, packet size in samples can change on the
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// fly, so we exclude this test by setting |packet_size_samples_| to -1.
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// fly, so we exclude this test by setting `packet_size_samples_` to -1.
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int clockrate_hz = sampling_freq_hz;
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size_t num_channels = 1;
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if (absl::EqualsIgnoreCase(codec_name, "G722")) {
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@ -509,8 +509,8 @@ void TestStereo::Run(TestPackStereo* channel,
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in_file_mono_->FastForward(100);
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while (1) {
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// Simulate packet loss by setting |packet_loss_| to "true" in
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// |percent_loss| percent of the loops.
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// Simulate packet loss by setting `packet_loss_` to "true" in
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// `percent_loss` percent of the loops.
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if (percent_loss > 0) {
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if (counter_ == floor((100 / percent_loss) + 0.5)) {
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counter_ = 0;
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@ -23,7 +23,7 @@
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namespace webrtc {
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// This class records the frame type, and delegates actual sending to the
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// |next_| AudioPacketizationCallback.
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// `next_` AudioPacketizationCallback.
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class MonitoringAudioPacketizationCallback : public AudioPacketizationCallback {
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public:
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explicit MonitoringAudioPacketizationCallback(
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@ -67,9 +67,9 @@ class TestVadDtx {
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// the expectation. Saves result to a file.
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// expects[x] means
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// -1 : do not care,
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// 0 : there have been no packets of type |x|,
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// 1 : there have been packets of type |x|,
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// with |x| indicates the following packet types
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// 0 : there have been no packets of type `x`,
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// 1 : there have been packets of type `x`,
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// with `x` indicates the following packet types
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// 0 - kEmptyFrame
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// 1 - kAudioFrameSpeech
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// 2 - kAudioFrameCN
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@ -277,8 +277,8 @@ void OpusTest::Run(TestPackStereo* channel,
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ASSERT_GE(bitstream_len_byte_int, 0);
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bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);
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// Simulate packet loss by setting |packet_loss_| to "true" in
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// |percent_loss| percent of the loops.
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// Simulate packet loss by setting `packet_loss_` to "true" in
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// `percent_loss` percent of the loops.
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// TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
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if (percent_loss > 0) {
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if (counter_ == floor((100 / percent_loss) + 0.5)) {
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