Pass absolute capture time from WebRtcVoiceEngine to ACM.
Bug: webrtc:10739
Change-Id: I6f264cb89ce340db642db3ef7dfc2b5d459f749e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167211
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30547}
diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
index d9212a2..47459ac 100644
--- a/api/audio/audio_frame.cc
+++ b/api/audio/audio_frame.cc
@@ -41,6 +41,7 @@
vad_activity_ = kVadUnknown;
profile_timestamp_ms_ = 0;
packet_infos_ = RtpPacketInfos();
+ absolute_capture_timestamp_ms_ = absl::nullopt;
}
void AudioFrame::UpdateFrame(uint32_t timestamp,
@@ -86,6 +87,7 @@
vad_activity_ = src.vad_activity_;
num_channels_ = src.num_channels_;
channel_layout_ = src.channel_layout_;
+ absolute_capture_timestamp_ms_ = src.absolute_capture_timestamp_ms();
const size_t length = samples_per_channel_ * num_channels_;
RTC_CHECK_LE(length, kMaxDataSizeSamples);
diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
index cda8c26..06b0b28 100644
--- a/api/audio/audio_frame.h
+++ b/api/audio/audio_frame.h
@@ -104,6 +104,15 @@
ChannelLayout channel_layout() const { return channel_layout_; }
int sample_rate_hz() const { return sample_rate_hz_; }
+ void set_absolute_capture_timestamp_ms(
+ int64_t absolute_capture_time_stamp_ms) {
+ absolute_capture_timestamp_ms_ = absolute_capture_time_stamp_ms;
+ }
+
+ absl::optional<int64_t> absolute_capture_timestamp_ms() const {
+ return absolute_capture_timestamp_ms_;
+ }
+
// RTP timestamp of the first sample in the AudioFrame.
uint32_t timestamp_ = 0;
// Time since the first frame in milliseconds.
@@ -121,8 +130,8 @@
// Monotonically increasing timestamp intended for profiling of audio frames.
// Typically used for measuring elapsed time between two different points in
// the audio path. No lock is used to save resources and we are thread safe
- // by design. Also, absl::optional is not used since it will cause a "complex
- // class/struct needs an explicit out-of-line destructor" build error.
+ // by design.
+ // TODO(nisse@webrtc.org): consider using absl::optional.
int64_t profile_timestamp_ms_ = 0;
// Information about packets used to assemble this audio frame. This is needed
@@ -150,6 +159,12 @@
int16_t data_[kMaxDataSizeSamples];
bool muted_ = true;
+ // Absolute capture timestamp when this audio frame was originally captured.
+ // This is only valid for audio frames captured on this machine. The absolute
+ // capture timestamp of a received frame is found in |packet_infos_|.
+ // This timestamp MUST be based on the same clock as rtc::TimeMillis().
+ absl::optional<int64_t> absolute_capture_timestamp_ms_;
+
RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
};
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 2fe2563..b4b2b4a 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -880,8 +880,12 @@
audio_frame->timestamp_, static_cast<const int16_t*>(audio_data),
number_of_frames, sample_rate, audio_frame->speech_type_,
audio_frame->vad_activity_, number_of_channels);
- // TODO(bugs.webrtc.org/10739): pass absolute_capture_timestamp_ms to
- // stream_.
+ // TODO(bugs.webrtc.org/10739): add dcheck that
+ // |absolute_capture_timestamp_ms| always receives a value.
+ if (absolute_capture_timestamp_ms) {
+ audio_frame->set_absolute_capture_timestamp_ms(
+ *absolute_capture_timestamp_ms);
+ }
stream_->SendAudioData(std::move(audio_frame));
}
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index f3dd5b1..e28be18 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -109,7 +109,6 @@
// 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_);
@@ -132,7 +131,11 @@
int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
- int Encode(const InputData& input_data)
+
+ // TODO(bugs.webrtc.org/10739): change |absolute_capture_timestamp_ms| to
+ // int64_t when it always receives a valid value.
+ int Encode(const InputData& input_data,
+ absl::optional<int64_t> absolute_capture_timestamp_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
int InitializeReceiverSafe() RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
@@ -231,7 +234,11 @@
AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
-int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
+int32_t AudioCodingModuleImpl::Encode(
+ const InputData& input_data,
+ absl::optional<int64_t> absolute_capture_timestamp_ms) {
+ // TODO(bugs.webrtc.org/10739): add dcheck that
+ // |audio_frame.absolute_capture_timestamp_ms()| always has a value.
AudioEncoder::EncodedInfo encoded_info;
uint8_t previous_pltype;
@@ -304,7 +311,7 @@
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);
+ absolute_capture_timestamp_ms.value_or(-1));
}
if (vad_callback_) {
@@ -339,7 +346,11 @@
int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
rtc::CritScope lock(&acm_crit_sect_);
int r = Add10MsDataInternal(audio_frame, &input_data_);
- return r < 0 ? r : Encode(input_data_);
+ // TODO(bugs.webrtc.org/10739): add dcheck that
+ // |audio_frame.absolute_capture_timestamp_ms()| always has a value.
+ return r < 0
+ ? r
+ : Encode(input_data_, audio_frame.absolute_capture_timestamp_ms());
}
int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
@@ -394,9 +405,6 @@
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