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Unified Diff: webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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Index: webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 61923aa44765ab9f51db4850de2a8bb2eba381be..745386d485c1259f7bd8adfed3cb3b8c77fa6ac6 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -61,8 +61,7 @@ class VerifyingAudioReceiver : public NullRtpData {
class RTPCallback : public NullRtpFeedback {
public:
- int32_t OnInitializeDecoder(const int32_t id,
- const int8_t payloadType,
+ int32_t OnInitializeDecoder(const int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const uint8_t channels,
@@ -80,7 +79,6 @@ class RtpRtcpAudioTest : public ::testing::Test {
RtpRtcpAudioTest() : fake_clock(123456) {
test_CSRC[0] = 1234;
test_CSRC[2] = 2345;
- test_id = 123;
test_ssrc = 3456;
test_timestamp = 4567;
test_sequence_number = 2345;
@@ -104,7 +102,6 @@ class RtpRtcpAudioTest : public ::testing::Test {
RTPPayloadStrategy::CreateStrategy(true)));
RtpRtcp::Configuration configuration;
- configuration.id = test_id;
configuration.audio = true;
configuration.clock = &fake_clock;
configuration.receive_statistics = receive_statistics1_.get();
@@ -113,18 +110,17 @@ class RtpRtcpAudioTest : public ::testing::Test {
module1 = RtpRtcp::CreateRtpRtcp(configuration);
rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
- test_id, &fake_clock, audioFeedback, data_receiver1, NULL,
+ &fake_clock, audioFeedback, data_receiver1, NULL,
rtp_payload_registry1_.get()));
- configuration.id = test_id + 1;
configuration.receive_statistics = receive_statistics2_.get();
configuration.outgoing_transport = transport2;
configuration.audio_messages = audioFeedback;
module2 = RtpRtcp::CreateRtpRtcp(configuration);
rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
- test_id + 1, &fake_clock, audioFeedback, data_receiver2, NULL,
- rtp_payload_registry2_.get()));
+ &fake_clock, audioFeedback, data_receiver2, NULL,
+ rtp_payload_registry2_.get()));
transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
rtp_receiver2_.get(), receive_statistics2_.get());
@@ -143,7 +139,6 @@ class RtpRtcpAudioTest : public ::testing::Test {
delete rtp_callback;
}
- int test_id;
RtpRtcp* module1;
RtpRtcp* module2;
rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_;
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