Revert "Send absolute capture time through audio coding module."
This reverts commit 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a.
Reason for revert: failing upstream tests
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}
TBR=danilchap@webrtc.org,ossu@webrtc.org,minyue@webrtc.org,chxg@google.com
Change-Id: Ia36b9ae899563c9afd8612ffd83871b8a5778a2b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10739
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167212
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30364}
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index a8da77e..74a0c7a 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -107,8 +107,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_len_bytes,
- int64_t absolute_capture_timestamp_ms) override {
+ size_t payload_len_bytes) 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 b3e1e1e..55552ca 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -126,8 +126,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_len_bytes,
- int64_t absolute_capture_timestamp_ms) {
+ size_t payload_len_bytes) {
// 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 0c82415..f4a6fc4 100644
--- a/modules/audio_coding/acm2/acm_send_test.h
+++ b/modules/audio_coding/acm2/acm_send_test.h
@@ -54,8 +54,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_len_bytes,
- int64_t absolute_capture_timestamp_ms) override;
+ size_t payload_len_bytes) 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 f3dd5b1..b68579b 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/include/audio_coding_module.h"
#include <assert.h>
+
#include <algorithm>
#include <cstdint>
@@ -109,7 +110,6 @@
// 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_);
@@ -253,7 +253,6 @@
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;
@@ -303,8 +302,7 @@
if (packetization_callback_) {
packetization_callback_->SendData(
frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
- encode_buffer_.data(), encode_buffer_.size(),
- input_data.absolute_capture_timestamp_ms);
+ encode_buffer_.data(), encode_buffer_.size());
}
if (vad_callback_) {
@@ -394,9 +392,6 @@
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 fb26025..9dca4cd 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -111,8 +111,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_len_bytes,
- int64_t absolute_capture_timestamp_ms) override {
+ size_t payload_len_bytes) override {
rtc::CritScope lock(&crit_sect_);
++num_calls_;
last_frame_type_ = frame_type;