Send absolute capture time through audio coding module.
Bug: webrtc:10739
Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30363}
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index e76bacb..3590891 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -23,7 +23,8 @@
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
- size_t payloadSize) {
+ size_t payloadSize,
+ int64_t absolute_capture_timestamp_ms) {
RTPHeader rtp_header;
int32_t status;
size_t payloadDataSize = payloadSize;
diff --git a/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h
index 0b248c8..78129e5 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -51,7 +51,8 @@
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
- size_t payloadSize) override;
+ size_t payloadSize,
+ int64_t absolute_capture_timestamp_ms) override;
void RegisterReceiverACM(AudioCodingModule* acm);
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 20e415d..a1c005c 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -33,7 +33,8 @@
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
- const size_t payloadSize) {
+ const size_t payloadSize,
+ int64_t absolute_capture_timestamp_ms) {
_rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
_frequency);
return 1;
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index a3d1a26..c96a4d6 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -32,7 +32,8 @@
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
- const size_t payloadSize) override;
+ const size_t payloadSize,
+ int64_t absolute_capture_timestamp_ms) override;
private:
static void MakeRTPheader(uint8_t* rtpHeader,
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index be4460e..9cb3752 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -64,7 +64,8 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_size) {
+ size_t payload_size,
+ int64_t absolute_capture_timestamp_ms) {
RTPHeader rtp_header;
int32_t status;
diff --git a/modules/audio_coding/test/TestAllCodecs.h b/modules/audio_coding/test/TestAllCodecs.h
index ef56661..0c27641 100644
--- a/modules/audio_coding/test/TestAllCodecs.h
+++ b/modules/audio_coding/test/TestAllCodecs.h
@@ -29,7 +29,8 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_size) override;
+ size_t payload_size,
+ int64_t absolute_capture_timestamp_ms) override;
size_t payload_size();
uint32_t timestamp_diff();
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 42bdbd8..61d27aa 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -44,7 +44,8 @@
const uint8_t payload_type,
const uint32_t timestamp,
const uint8_t* payload_data,
- const size_t payload_size) {
+ const size_t payload_size,
+ int64_t absolute_capture_timestamp_ms) {
RTPHeader rtp_header;
int32_t status = 0;
diff --git a/modules/audio_coding/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h
index e950840..3ee4dbf 100644
--- a/modules/audio_coding/test/TestStereo.h
+++ b/modules/audio_coding/test/TestStereo.h
@@ -35,7 +35,8 @@
const uint8_t payload_type,
const uint32_t timestamp,
const uint8_t* payload_data,
- const size_t payload_size) override;
+ const size_t payload_size,
+ int64_t absolute_capture_timestamp_ms) override;
uint16_t payload_size();
uint32_t timestamp_diff();
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index e110924..5f70c03 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -337,7 +337,7 @@
// Send data to the channel. "channel" will handle the loss simulation.
channel->SendData(AudioFrameType::kAudioFrameSpeech, payload_type_,
- rtp_timestamp_, bitstream, bitstream_len_byte);
+ rtp_timestamp_, bitstream, bitstream_len_byte, 0);
if (first_packet) {
first_packet = false;
start_time_stamp = rtp_timestamp_;