sdp: temporarily relax clockrate requirements for statically assigned payload types
to allow for downstream users to upgrade. BUG=chromium:1338902 Change-Id: If6b56ab63f7859c13e9ebc70326e1088e5dfff1a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268141 Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37475}
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@ -3636,9 +3636,9 @@ bool ParseRtpmapAttribute(absl::string_view line,
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AudioContentDescription* audio_desc = media_desc->as_audio();
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AudioContentDescription* audio_desc = media_desc->as_audio();
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for (const cricket::AudioCodec& existing_codec : audio_desc->codecs()) {
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for (const cricket::AudioCodec& existing_codec : audio_desc->codecs()) {
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// TODO(crbug.com/1338902) re-add check for clockrate.
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if (!existing_codec.name.empty() && payload_type == existing_codec.id &&
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if (!existing_codec.name.empty() && payload_type == existing_codec.id &&
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(!absl::EqualsIgnoreCase(encoding_name, existing_codec.name) ||
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(!absl::EqualsIgnoreCase(encoding_name, existing_codec.name) ||
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clock_rate != existing_codec.clockrate ||
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channels != existing_codec.channels)) {
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channels != existing_codec.channels)) {
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rtc::StringBuilder description;
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rtc::StringBuilder description;
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description
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description
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@ -4665,11 +4665,10 @@ TEST_F(WebRtcSdpTest, DuplicateAudioRtpmapWithConflict) {
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"s=-\r\n"
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"s=-\r\n"
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"t=0 0\r\n"
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"t=0 0\r\n"
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"m=audio 49232 RTP/AVP 108\r\n"
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"m=audio 49232 RTP/AVP 108\r\n"
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// Same name but different payload type.
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"a=rtpmap:108 ISAC/16000\r\n"
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"a=rtpmap:108 ISAC/16000\r\n"
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"a=rtpmap:108 ISAC/32000\r\n";
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"a=rtpmap:108 G711/16000\r\n";
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ExpectParseFailure(sdp, "a=rtpmap:108 ISAC/32000");
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ExpectParseFailure(sdp, "a=rtpmap:108 G711/16000");
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}
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}
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TEST_F(WebRtcSdpTest, DuplicateVideoRtpmapWithConflict) {
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TEST_F(WebRtcSdpTest, DuplicateVideoRtpmapWithConflict) {
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