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Removed double-special-casing of ISAC in libjingle and WebRtcVoE.

Browse Source 
webrtcvoiceengine.cc ensured that if the bitrate set for ISAC was 0,
it was changed to -1 so that the codec could manage the bitrate
itself.

webrtcsdp.cc ensured that if the bitrate set for ISAC was 0, it was
explicitly set to default values to avoid the codec's built in bitrate
management.

Eventually, there'll be no codec specific code like this in these
layers. This is one step towards that goal.

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2642923003
Cr-Commit-Position: refs/heads/master@{#16220}
This commit is contained in:
ossu 2017-01-23 08:55:48 -08:00 committed by Commit bot
parent cb893ee634
commit e1405ad0d1
4 changed files with 5 additions and 35 deletions
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6
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -371,12 +371,6 @@ class WebRtcVoiceCodecs final {
// Reset G722 sample rate to 16000 to match WebRTC.
MaybeFixupG722(&voe_codec, 16000);
// Apply codec-specific settings.
if (IsCodec(codec, kIsacCodecName)) {
// If ISAC and an explicit bitrate is not specified,
// enable auto bitrate adjustment.
voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1;
}
*out = voe_codec;
}
return true;

2
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -1888,7 +1888,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) {
const auto& gcodec = GetSendStreamConfig(kSsrc1).send_codec_spec.codec_inst;
EXPECT_EQ(103, gcodec.pltype);
EXPECT_STREQ("ISAC", gcodec.plname);
EXPECT_EQ(-1, gcodec.rate);
EXPECT_EQ(32000, gcodec.rate);
}
parameters.codecs[0].bitrate = 28000; // bitrate == 28000
SetSendParameters(parameters);

19
webrtc/pc/webrtcsdp.cc
View File

@ -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
@ -3098,21 +3093,9 @@ 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;
}
}
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 =

13
webrtc/pc/webrtcsdp_unittest.cc
View File

@ -1128,8 +1128,8 @@ class WebRtcSdpTest : public testing::Test {
audio->set_protocol(cricket::kMediaProtocolSavpf);
AudioCodec opus(111, "opus", 48000, 0, 2);
audio->AddCodec(opus);
audio->AddCodec(AudioCodec(103, "ISAC", 16000, 32000, 1));
audio->AddCodec(AudioCodec(104, "ISAC", 32000, 56000, 1));
audio->AddCodec(AudioCodec(103, "ISAC", 16000, 0, 1));
audio->AddCodec(AudioCodec(104, "ISAC", 32000, 0, 1));
return audio;
}
@ -1664,13 +1664,6 @@ class WebRtcSdpTest : public testing::Test {
cricket::AudioCodec codec = acd->codecs()[i];
VerifyCodecParameter(codec.params, "ptime", params.ptime);
VerifyCodecParameter(codec.params, "maxptime", params.max_ptime);
if (codec.name == "ISAC") {
if (codec.clockrate == 16000) {
EXPECT_EQ(32000, codec.bitrate);
} else {
EXPECT_EQ(56000, codec.bitrate);
}
}
}
const ContentInfo* vc = GetFirstVideoContent(jdesc_output->description());
@ -2275,7 +2268,7 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutRtpmap) {
// the payload types (<fmt>s) on the m= line.
ref_codecs.push_back(AudioCodec(0, "PCMU", 8000, 0, 1));
ref_codecs.push_back(AudioCodec(18, "G729", 16000, 0, 1));
ref_codecs.push_back(AudioCodec(103, "ISAC", 16000, 32000, 1));
ref_codecs.push_back(AudioCodec(103, "ISAC", 16000, 0, 1));
EXPECT_EQ(ref_codecs, audio->codecs());
}