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Unified Diff: webrtc/api/webrtcsdp.cc

Issue 2642923003: Removed double-special-casing of ISAC in libjingle and WebRtcVoE. (Closed)
Patch Set: Created 3 years, 11 months ago
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Index: webrtc/api/webrtcsdp.cc
diff --git a/webrtc/api/webrtcsdp.cc b/webrtc/api/webrtcsdp.cc
index e19840bdf2f94a417d049b01a0f0a620d8bc9aa0..009527d157a0acb8ebfa75908c8d4da2f9fcab7f 100644
--- a/webrtc/api/webrtcsdp.cc
+++ b/webrtc/api/webrtcsdp.cc
@@ -205,11 +205,6 @@ static const char kApplicationSpecificMaximum[] = "AS";
static const int kDefaultVideoClockrate = 90000;
-// ISAC special-case.
-static const char kIsacCodecName[] = "ISAC"; // From webrtcvoiceengine.cc
-static const int kIsacWbDefaultRate = 32000; // From acm_common_defs.h
-static const int kIsacSwbDefaultRate = 56000; // From acm_common_defs.h
-
static const char kDefaultSctpmapProtocol[] = "webrtc-datachannel";
// RTP payload type is in the 0-127 range. Use -1 to indicate "all" payload
@@ -3092,21 +3087,9 @@ bool ParseRtpmapAttribute(const std::string& line,
return false;
}
}
- int bitrate = 0;
- // The default behavior for ISAC (bitrate == 0) in webrtcvoiceengine.cc
- // (specifically FindWebRtcCodec) is bandwidth-adaptive variable bitrate.
- // The bandwidth adaptation doesn't always work well, so this code
- // sets a fixed target bitrate instead.
- if (_stricmp(encoding_name.c_str(), kIsacCodecName) == 0) {
- if (clock_rate <= 16000) {
- bitrate = kIsacWbDefaultRate;
- } else {
- bitrate = kIsacSwbDefaultRate;
- }
- }
kwiberg-webrtc 2017/01/23 15:18:11 Hmm. I don't quite see how this piece of code didn
ossu 2017/01/23 15:33:18 It did have an effect. It worked around the block
AudioContentDescription* audio_desc =
static_cast<AudioContentDescription*>(media_desc);
- UpdateCodec(payload_type, encoding_name, clock_rate, bitrate, channels,
+ UpdateCodec(payload_type, encoding_name, clock_rate, 0, channels,
audio_desc);
} else if (media_type == cricket::MEDIA_TYPE_DATA) {
DataContentDescription* data_desc =
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