zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Send absolute capture time through audio coding module.

Browse Source 
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}
This commit is contained in:
Minyue Li
2020-01-23 13:45:50 +01:00
committed by Commit Bot
parent cdd73e095c
commit 48655cfdbf
17 changed files with 69 additions and 30 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
modules/audio_coding/test/Channel.cc
View File

@ -23,7 +23,8 @@ int32_t Channel::SendData(AudioFrameType frameType,
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;

3
modules/audio_coding/test/Channel.h
View File

@ -51,7 +51,8 @@ class Channel : public AudioPacketizationCallback {
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);

3
modules/audio_coding/test/EncodeDecodeTest.cc
View File

@ -33,7 +33,8 @@ int32_t TestPacketization::SendData(const AudioFrameType /* frameType */,
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;

3
modules/audio_coding/test/EncodeDecodeTest.h
View File

@ -32,7 +32,8 @@ class TestPacketization : public AudioPacketizationCallback {
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,

3
modules/audio_coding/test/TestAllCodecs.cc
View File

@ -64,7 +64,8 @@ int32_t TestPack::SendData(AudioFrameType frame_type,
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;

3
modules/audio_coding/test/TestAllCodecs.h
View File

@ -29,7 +29,8 @@ class TestPack : public AudioPacketizationCallback {
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();

3
modules/audio_coding/test/TestStereo.cc
View File

@ -44,7 +44,8 @@ int32_t TestPackStereo::SendData(const AudioFrameType frame_type,
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;

3
modules/audio_coding/test/TestStereo.h
View File

@ -35,7 +35,8 @@ class TestPackStereo : public AudioPacketizationCallback {
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();

2
modules/audio_coding/test/opus_test.cc
View File

@ -337,7 +337,7 @@ void OpusTest::Run(TestPackStereo* channel,
// 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_;