Reland "Send absolute capture time through audio coding module."
This is a reland of 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a
Original change's description:
> 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}
Bug: webrtc:10739
Change-Id: I10086d239ad3f1efb8485098bf3b0ad23110962b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167213
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30380}
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 74a0c7a..a8da77e 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -107,7 +107,8 @@
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;
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index 55552ca..b3e1e1e 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -126,7 +126,8 @@
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;
diff --git a/modules/audio_coding/acm2/acm_send_test.h b/modules/audio_coding/acm2/acm_send_test.h
index f4a6fc4..0c82415 100644
--- a/modules/audio_coding/acm2/acm_send_test.h
+++ b/modules/audio_coding/acm2/acm_send_test.h
@@ -54,7 +54,8 @@
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(); }
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index b68579b..f3dd5b1 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -11,7 +11,6 @@
#include "modules/audio_coding/include/audio_coding_module.h"
#include <assert.h>
-
#include <algorithm>
#include <cstdint>
@@ -110,6 +109,7 @@
// 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 @@
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 @@
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 @@
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
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 9dca4cd..fb26025 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -111,7 +111,8 @@
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;
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index d8c9260..ada389f 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -44,7 +44,21 @@
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.
+ return SendData(frame_type, payload_type, timestamp, payload_data,
+ payload_len_bytes);
+ }
+ virtual int32_t SendData(AudioFrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_len_bytes) {
+ RTC_NOTREACHED() << "This method must be overridden, or not used.";
+ return -1;
+ }
};
// Callback class used for reporting VAD decision
diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc
index f65679d..204f169 100644
--- a/modules/audio_coding/neteq/tools/rtp_encode.cc
+++ b/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -112,7 +112,8 @@
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;
}
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_;