Make capture timestamp optional in ADM.
This is to avoid using 0 as a default value.
Also fix a bug in audio_device_buffer where the timestamp aligner used the wrong input timestamp.
Bug: webrtc:13609
Change-Id: I00016e68ab50d052990c2b9f80aa1e2d7e167b93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291118
Reviewed-by: Olov Brändström <brandstrom@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39177}
diff --git a/audio/audio_transport_impl.cc b/audio/audio_transport_impl.cc
index 9f2823b..42a81d5 100644
--- a/audio/audio_transport_impl.cc
+++ b/audio/audio_transport_impl.cc
@@ -107,35 +107,35 @@
int32_t AudioTransportImpl::RecordedDataIsAvailable(
const void* audio_data,
- const size_t number_of_frames,
- const size_t bytes_per_sample,
- const size_t number_of_channels,
- const uint32_t sample_rate,
- const uint32_t audio_delay_milliseconds,
- const int32_t clock_drift,
- const uint32_t volume,
- const bool key_pressed,
+ size_t number_of_frames,
+ size_t bytes_per_sample,
+ size_t number_of_channels,
+ uint32_t sample_rate,
+ uint32_t audio_delay_milliseconds,
+ int32_t clock_drift,
+ uint32_t volume,
+ bool key_pressed,
uint32_t& new_mic_volume) { // NOLINT: to avoid changing APIs
return RecordedDataIsAvailable(
audio_data, number_of_frames, bytes_per_sample, number_of_channels,
sample_rate, audio_delay_milliseconds, clock_drift, volume, key_pressed,
- new_mic_volume, /* estimated_capture_time_ns */ 0);
+ new_mic_volume, /*estimated_capture_time_ns=*/absl::nullopt);
}
// Not used in Chromium. Process captured audio and distribute to all sending
// streams, and try to do this at the lowest possible sample rate.
int32_t AudioTransportImpl::RecordedDataIsAvailable(
const void* audio_data,
- const size_t number_of_frames,
- const size_t bytes_per_sample,
- const size_t number_of_channels,
- const uint32_t sample_rate,
- const uint32_t audio_delay_milliseconds,
- const int32_t /*clock_drift*/,
- const uint32_t /*volume*/,
- const bool key_pressed,
+ size_t number_of_frames,
+ size_t bytes_per_sample,
+ size_t number_of_channels,
+ uint32_t sample_rate,
+ uint32_t audio_delay_milliseconds,
+ int32_t /*clock_drift*/,
+ uint32_t /*volume*/,
+ bool key_pressed,
uint32_t& /*new_mic_volume*/,
- const int64_t
+ absl::optional<int64_t>
estimated_capture_time_ns) { // NOLINT: to avoid changing APIs
RTC_DCHECK(audio_data);
RTC_DCHECK_GE(number_of_channels, 1);
@@ -166,8 +166,11 @@
ProcessCaptureFrame(audio_delay_milliseconds, key_pressed,
swap_stereo_channels, audio_processing_,
audio_frame.get());
- audio_frame->set_absolute_capture_timestamp_ms(estimated_capture_time_ns /
- 1000000);
+
+ if (estimated_capture_time_ns) {
+ audio_frame->set_absolute_capture_timestamp_ms(*estimated_capture_time_ns /
+ 1000000);
+ }
RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
if (async_audio_processing_)
diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h
index ba067de..24b09d2 100644
--- a/audio/audio_transport_impl.h
+++ b/audio/audio_transport_impl.h
@@ -52,17 +52,18 @@
bool keyPressed,
uint32_t& newMicLevel) override;
- int32_t RecordedDataIsAvailable(const void* audioSamples,
- size_t nSamples,
- size_t nBytesPerSample,
- size_t nChannels,
- uint32_t samplesPerSec,
- uint32_t totalDelayMS,
- int32_t clockDrift,
- uint32_t currentMicLevel,
- bool keyPressed,
- uint32_t& newMicLevel,
- int64_t estimated_capture_time_ns) override;
+ int32_t RecordedDataIsAvailable(
+ const void* audioSamples,
+ size_t nSamples,
+ size_t nBytesPerSample,
+ size_t nChannels,
+ uint32_t samplesPerSec,
+ uint32_t totalDelayMS,
+ int32_t clockDrift,
+ uint32_t currentMicLevel,
+ bool keyPressed,
+ uint32_t& newMicLevel,
+ absl::optional<int64_t> estimated_capture_time_ns) override;
int32_t NeedMorePlayData(size_t nSamples,
size_t nBytesPerSample,
diff --git a/modules/audio_device/audio_device_buffer.cc b/modules/audio_device/audio_device_buffer.cc
index 6232a93..b1be445 100644
--- a/modules/audio_device/audio_device_buffer.cc
+++ b/modules/audio_device/audio_device_buffer.cc
@@ -55,7 +55,6 @@
typing_status_(false),
play_delay_ms_(0),
rec_delay_ms_(0),
- capture_timestamp_ns_(0),
num_stat_reports_(0),
last_timer_task_time_(0),
rec_stat_count_(0),
@@ -231,12 +230,13 @@
int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
size_t samples_per_channel) {
- return SetRecordedBuffer(audio_buffer, samples_per_channel, 0);
+ return SetRecordedBuffer(audio_buffer, samples_per_channel, absl::nullopt);
}
-int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
- size_t samples_per_channel,
- int64_t capture_timestamp_ns) {
+int32_t AudioDeviceBuffer::SetRecordedBuffer(
+ const void* audio_buffer,
+ size_t samples_per_channel,
+ absl::optional<int64_t> capture_timestamp_ns) {
// Copy the complete input buffer to the local buffer.
const size_t old_size = rec_buffer_.size();
rec_buffer_.SetData(static_cast<const int16_t*>(audio_buffer),
@@ -247,17 +247,13 @@
RTC_LOG(LS_INFO) << "Size of recording buffer: " << rec_buffer_.size();
}
- // If the timestamp is less then or equal to zero, it's not valid and are
- // ignored. If we do antimestamp alignment on them they might accidentally
- // become greater then zero, and will be handled as if they were a correct
- // timestamp.
- capture_timestamp_ns_ =
- (capture_timestamp_ns > 0)
- ? rtc::kNumNanosecsPerMicrosec *
- timestamp_aligner_.TranslateTimestamp(
- capture_timestamp_ns_ / rtc::kNumNanosecsPerMicrosec,
- rtc::TimeMicros())
- : capture_timestamp_ns;
+ if (capture_timestamp_ns) {
+ capture_timestamp_ns_ =
+ rtc::kNumNanosecsPerMicrosec *
+ timestamp_aligner_.TranslateTimestamp(
+ *capture_timestamp_ns / rtc::kNumNanosecsPerMicrosec,
+ rtc::TimeMicros());
+ }
// Derive a new level value twice per second and check if it is non-zero.
int16_t max_abs = 0;
RTC_DCHECK_LT(rec_stat_count_, 50);
diff --git a/modules/audio_device/audio_device_buffer.h b/modules/audio_device/audio_device_buffer.h
index 9a6a88a..eb681a7 100644
--- a/modules/audio_device/audio_device_buffer.h
+++ b/modules/audio_device/audio_device_buffer.h
@@ -102,9 +102,10 @@
virtual int32_t SetRecordedBuffer(const void* audio_buffer,
size_t samples_per_channel);
- virtual int32_t SetRecordedBuffer(const void* audio_buffer,
- size_t samples_per_channel,
- int64_t capture_timestamp_ns);
+ virtual int32_t SetRecordedBuffer(
+ const void* audio_buffer,
+ size_t samples_per_channel,
+ absl::optional<int64_t> capture_timestamp_ns);
virtual void SetVQEData(int play_delay_ms, int rec_delay_ms);
virtual int32_t DeliverRecordedData();
uint32_t NewMicLevel() const;
@@ -194,7 +195,7 @@
int rec_delay_ms_;
// Capture timestamp.
- int64_t capture_timestamp_ns_;
+ absl::optional<int64_t> capture_timestamp_ns_;
// Counts number of times LogStats() has been called.
size_t num_stat_reports_ RTC_GUARDED_BY(task_queue_);
diff --git a/modules/audio_device/audio_device_data_observer.cc b/modules/audio_device/audio_device_data_observer.cc
index 3775e7c..0524830 100644
--- a/modules/audio_device/audio_device_data_observer.cc
+++ b/modules/audio_device/audio_device_data_observer.cc
@@ -55,24 +55,25 @@
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel) override {
- return RecordedDataIsAvailable(audioSamples, nSamples, nBytesPerSample,
- nChannels, samples_per_sec, total_delay_ms,
- clockDrift, currentMicLevel, keyPressed,
- newMicLevel, /*capture_timestamp_ns*/ 0);
+ return RecordedDataIsAvailable(
+ audioSamples, nSamples, nBytesPerSample, nChannels, samples_per_sec,
+ total_delay_ms, clockDrift, currentMicLevel, keyPressed, newMicLevel,
+ /*capture_timestamp_ns=*/absl::nullopt);
}
// AudioTransport methods overrides.
- int32_t RecordedDataIsAvailable(const void* audioSamples,
- size_t nSamples,
- size_t nBytesPerSample,
- size_t nChannels,
- uint32_t samples_per_sec,
- uint32_t total_delay_ms,
- int32_t clockDrift,
- uint32_t currentMicLevel,
- bool keyPressed,
- uint32_t& newMicLevel,
- int64_t capture_timestamp_ns) override {
+ int32_t RecordedDataIsAvailable(
+ const void* audioSamples,
+ size_t nSamples,
+ size_t nBytesPerSample,
+ size_t nChannels,
+ uint32_t samples_per_sec,
+ uint32_t total_delay_ms,
+ int32_t clockDrift,
+ uint32_t currentMicLevel,
+ bool keyPressed,
+ uint32_t& newMicLevel,
+ absl::optional<int64_t> capture_timestamp_ns) override {
int32_t res = 0;
// Capture PCM data of locally captured audio.
if (observer_) {
diff --git a/modules/audio_device/include/audio_device_defines.h b/modules/audio_device/include/audio_device_defines.h
index 89d33f8..d677d41 100644
--- a/modules/audio_device/include/audio_device_defines.h
+++ b/modules/audio_device/include/audio_device_defines.h
@@ -56,7 +56,7 @@
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel,
- int64_t estimatedCaptureTimeNS) { // NOLINT
+ absl::optional<int64_t> estimatedCaptureTimeNS) { // NOLINT
// TODO(webrtc:13620) Make the default behaver of the new API to behave as
// the old API. This can be pure virtual if all uses of the old API is
// removed.
diff --git a/modules/audio_device/include/mock_audio_transport.h b/modules/audio_device/include/mock_audio_transport.h
index e1be5f4..b886967 100644
--- a/modules/audio_device/include/mock_audio_transport.h
+++ b/modules/audio_device/include/mock_audio_transport.h
@@ -48,7 +48,7 @@
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel,
- int64_t estimated_capture_time_ns),
+ absl::optional<int64_t> estimated_capture_time_ns),
(override));
MOCK_METHOD(int32_t,
