commit 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a
author Minyue Li <minyue@webrtc.org> Thu Jan 23 13:45:50 2020 +0100
committer Commit Bot <commit-bot@chromium.org> Thu Jan 23 16:06:12 2020 +0000
tree bf8c4008fa96ec309dbe23055b8fd89d1672f4e1
parent cdd73e095cd4279c80697692a8652cf6129171db

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}

modules/audio_coding/acm2/acm_receiver_unittest.cc

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;
 

modules/audio_coding/acm2/acm_send_test.cc

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;

modules/audio_coding/acm2/acm_send_test.h

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(); }
 

modules/audio_coding/acm2/audio_coding_module.cc

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

modules/audio_coding/acm2/audio_coding_module_unittest.cc

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;