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Update ACM to use RTPHeader instead of WebRtcRTPHeader

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Bug: webrtc:5876
Change-Id: Id3311dcf508cca34495349197eeac2edf8783772
Reviewed-on: https://webrtc-review.googlesource.com/c/123188
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26729}
This commit is contained in:
Niels Möller
2019-02-15 15:21:47 +01:00
committed by Commit Bot
parent 389b1672a3
commit afb5dbbf4e
15 changed files with 100 additions and 120 deletions
Download Patch File Download Diff File Expand all files Collapse all files

34
modules/audio_coding/acm2/audio_coding_module_unittest.cc
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@ -71,21 +71,20 @@ class RtpUtility {
virtual ~RtpUtility() {}
void Populate(WebRtcRTPHeader* rtp_header) {
rtp_header->header.sequenceNumber = 0xABCD;
rtp_header->header.timestamp = 0xABCDEF01;
rtp_header->header.payloadType = payload_type_;
rtp_header->header.markerBit = false;
rtp_header->header.ssrc = 0x1234;
rtp_header->header.numCSRCs = 0;
rtp_header->frameType = kAudioFrameSpeech;
void Populate(RTPHeader* rtp_header) {
rtp_header->sequenceNumber = 0xABCD;
rtp_header->timestamp = 0xABCDEF01;
rtp_header->payloadType = payload_type_;
rtp_header->markerBit = false;
rtp_header->ssrc = 0x1234;
rtp_header->numCSRCs = 0;
rtp_header->header.payload_type_frequency = kSampleRateHz;
rtp_header->payload_type_frequency = kSampleRateHz;
}
void Forward(WebRtcRTPHeader* rtp_header) {
++rtp_header->header.sequenceNumber;
rtp_header->header.timestamp += samples_per_packet_;
void Forward(RTPHeader* rtp_header) {
++rtp_header->sequenceNumber;
rtp_header->timestamp += samples_per_packet_;
}
private:
@ -237,7 +236,7 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
std::unique_ptr<RtpUtility> rtp_utility_;
std::unique_ptr<AudioCodingModule> acm_;
PacketizationCallbackStubOldApi packet_cb_;
WebRtcRTPHeader rtp_header_;
RTPHeader rtp_header_;
AudioFrame input_frame_;
absl::optional<SdpAudioFormat> audio_format_;
@ -792,16 +791,15 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
++receive_packet_count_;
// Encode new frame.
uint32_t input_timestamp = rtp_header_.header.timestamp;
uint32_t input_timestamp = rtp_header_.timestamp;
while (info.encoded_bytes == 0) {
info = isac_encoder_->Encode(input_timestamp,
audio_loop_.GetNextBlock(), &encoded);
input_timestamp += 160; // 10 ms at 16 kHz.
}
EXPECT_EQ(rtp_header_.header.timestamp + kPacketSizeSamples,
input_timestamp);
EXPECT_EQ(rtp_header_.header.timestamp, info.encoded_timestamp);
EXPECT_EQ(rtp_header_.header.payloadType, info.payload_type);
EXPECT_EQ(rtp_header_.timestamp + kPacketSizeSamples, input_timestamp);
EXPECT_EQ(rtp_header_.timestamp, info.encoded_timestamp);
EXPECT_EQ(rtp_header_.payloadType, info.payload_type);
}
// Now we're not holding the crit sect when calling ACM.