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Send absolute capture time through audio coding module.

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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
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33
audio/channel_send.cc
View File

@ -146,17 +146,19 @@ class ChannelSend : public ChannelSendInterface,
// From AudioPacketizationCallback in the ACM
int32_t SendData(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
uint32_t rtp_timestamp,
const uint8_t* payloadData,
size_t payloadSize) override;
size_t payloadSize,
int64_t absolute_capture_timestamp_ms) override;
void OnUplinkPacketLossRate(float packet_loss_rate);
bool InputMute() const;
int32_t SendRtpAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
rtc::ArrayView<const uint8_t> payload)
uint32_t rtp_timestamp,
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms)
RTC_RUN_ON(encoder_queue_);
void OnReceivedRtt(int64_t rtt_ms);
@ -360,18 +362,21 @@ class VoERtcpObserver : public RtcpBandwidthObserver {
int32_t ChannelSend::SendData(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
uint32_t rtp_timestamp,
const uint8_t* payloadData,
size_t payloadSize) {
size_t payloadSize,
int64_t absolute_capture_timestamp_ms) {
RTC_DCHECK_RUN_ON(&encoder_queue_);
rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
return SendRtpAudio(frameType, payloadType, timeStamp, payload);
return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload,
absolute_capture_timestamp_ms);
}
int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
rtc::ArrayView<const uint8_t> payload) {
uint32_t rtp_timestamp,
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms) {
if (_includeAudioLevelIndication) {
// Store current audio level in the RTP sender.
// The level will be used in combination with voice-activity state
@ -419,7 +424,7 @@ int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
// Push data from ACM to RTP/RTCP-module to deliver audio frame for
// packetization.
if (!_rtpRtcpModule->OnSendingRtpFrame(timeStamp,
if (!_rtpRtcpModule->OnSendingRtpFrame(rtp_timestamp,
// Leaving the time when this frame was
// received from the capture device as
// undefined for voice for now.
@ -433,10 +438,12 @@ int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
// call.
// TODO(nisse): Delete RTCPSender:timestamp_offset_, and see if we can confine
// knowledge of the offset to a single place.
const uint32_t rtp_timestamp = timeStamp + _rtpRtcpModule->StartTimestamp();
// This call will trigger Transport::SendPacket() from the RTP/RTCP module.
if (!rtp_sender_audio_->SendAudio(frameType, payloadType, rtp_timestamp,
payload.data(), payload.size())) {
if (!rtp_sender_audio_->SendAudio(
frameType, payloadType,
rtp_timestamp + _rtpRtcpModule->StartTimestamp(), payload.data(),
payload.size(), absolute_capture_timestamp_ms)) {
RTC_DLOG(LS_ERROR)
<< "ChannelSend::SendData() failed to send data to RTP/RTCP module";
return -1;

3
modules/audio_coding/acm2/acm_receiver_unittest.cc
View File

@ -107,7 +107,8 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes) override {
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) override {
if (frame_type == AudioFrameType::kEmptyFrame)
return 0;

3
modules/audio_coding/acm2/acm_send_test.cc
View File

@ -126,7 +126,8 @@ int32_t AcmSendTestOldApi::SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes) {
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) {
// Store the packet locally.
frame_type_ = frame_type;
payload_type_ = payload_type;

3
modules/audio_coding/acm2/acm_send_test.h
View File

@ -54,7 +54,8 @@ class AcmSendTestOldApi : public AudioPacketizationCallback,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes) override;
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) override;
AudioCodingModule* acm() { return acm_.get(); }

9
modules/audio_coding/acm2/audio_coding_module.cc
View File

@ -11,7 +11,6 @@
#include "modules/audio_coding/include/audio_coding_module.h"
#include <assert.h>
#include <algorithm>
#include <cstdint>
@ -110,6 +109,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
// If a re-mix is required (up or down), this buffer will store a re-mixed
// version of the input.
std::vector<int16_t> buffer;
int64_t absolute_capture_timestamp_ms;
};
InputData input_data_ RTC_GUARDED_BY(acm_crit_sect_);
@ -253,6 +253,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
int64_t{input_data.input_timestamp - last_timestamp_} *
encoder_stack_->RtpTimestampRateHz(),
int64_t{encoder_stack_->SampleRateHz()}));
last_timestamp_ = input_data.input_timestamp;
last_rtp_timestamp_ = rtp_timestamp;
first_frame_ = false;
@ -302,7 +303,8 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
if (packetization_callback_) {
packetization_callback_->SendData(
frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
encode_buffer_.data(), encode_buffer_.size());
encode_buffer_.data(), encode_buffer_.size(),
input_data.absolute_capture_timestamp_ms);
}
if (vad_callback_) {
@ -392,6 +394,9 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
input_data->input_timestamp = ptr_frame->timestamp_;
input_data->length_per_channel = ptr_frame->samples_per_channel_;
input_data->audio_channel = current_num_channels;
// TODO(bugs.webrtc.org/10739): Assign from a corresponding field in
// audio_frame when it is added in AudioFrame.
input_data->absolute_capture_timestamp_ms = 0;
if (!same_num_channels) {
// Remixes the input frame to the output data and in the process resize the

3
modules/audio_coding/acm2/audio_coding_module_unittest.cc
View File

@ -111,7 +111,8 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes) override {
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) override {
rtc::CritScope lock(&crit_sect_);
++num_calls_;
last_frame_type_ = frame_type;

16
modules/audio_coding/include/audio_coding_module.h
View File

@ -44,7 +44,21 @@ class AudioPacketizationCallback {
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes) = 0;
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) {
// TODO(bugs.webrtc.org/10739): Deprecate the old SendData and make this one
// pure virtual.
RTC_NOTREACHED() << "This method must be overridden, or not used.";
return -1;
}
virtual int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes) {
return SendData(frame_type, payload_type, timestamp, payload_data,
payload_len_bytes, 0);
}
};
// Callback class used for reporting VAD decision

3
modules/audio_coding/neteq/tools/rtp_encode.cc
View File

@ -112,7 +112,8 @@ class Packetizer : public AudioPacketizationCallback {
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes) override {
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) override {
if (payload_len_bytes == 0) {
return 0;
}

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_;