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Adding absolute capture timestamp to AudioTrackSinkInterface.

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Bug: webrtc:10739
Change-Id: I8c134cbe82452ac71625cd0c810c783a73f17822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167532
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30408}
This commit is contained in:
Minyue Li 2020-01-29 10:25:12 +01:00 committed by Commit Bot
parent 26b4cb3fc5
commit 99d6d8115b
8 changed files with 61 additions and 17 deletions
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20
api/media_stream_interface.h
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@ -202,7 +202,25 @@ class AudioTrackSinkInterface {
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) = 0;
size_t number_of_frames) {
RTC_NOTREACHED() << "This method must be overridden, or not used.";
}
// In this method, |absolute_capture_timestamp_ms|, when available, is
// supposed to deliver the timestamp when this audio frame was originally
// captured. This timestamp MUST be based on the same clock as
// rtc::TimeMillis().
virtual void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
absl::optional<int64_t> absolute_capture_timestamp_ms) {
// TODO(bugs.webrtc.org/10739): Deprecate the old OnData and make this one
// pure virtual.
return OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
number_of_frames);
}
protected:
virtual ~AudioTrackSinkInterface() {}

14
media/base/audio_source.h
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@ -13,6 +13,8 @@
#include <cstddef>
#include "absl/types/optional.h"
namespace cricket {
// Abstract interface for providing the audio data.
@ -23,11 +25,13 @@ class AudioSource {
class Sink {
public:
// Callback to receive data from the AudioSource.
virtual void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) = 0;
virtual void OnData(
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
absl::optional<int64_t> absolute_capture_timestamp_ms) = 0;
// Called when the AudioSource is going away.
virtual void OnClose() = 0;

3
media/base/fake_media_engine.cc
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@ -39,7 +39,8 @@ void FakeVoiceMediaChannel::VoiceChannelAudioSink::OnData(
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) {}
size_t number_of_frames,
absl::optional<int64_t> absolute_capture_timestamp_ms) {}
void FakeVoiceMediaChannel::VoiceChannelAudioSink::OnClose() {
source_ = nullptr;
}

3
media/base/fake_media_engine.h
View File

@ -368,7 +368,8 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override;
size_t number_of_frames,
absl::optional<int64_t> absolute_capture_timestamp_ms) override;
void OnClose() override;
AudioSource* source() const;

5
media/engine/webrtc_voice_engine.cc
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@ -870,7 +870,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override {
size_t number_of_frames,
absl::optional<int64_t> absolute_capture_timestamp_ms) override {
RTC_DCHECK_EQ(16, bits_per_sample);
RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
RTC_DCHECK(stream_);
@ -879,6 +880,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
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_.
stream_->SendAudioData(std::move(audio_frame));
}

5
pc/remote_audio_source.cc
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@ -144,8 +144,11 @@ void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
// Called on the externally-owned audio callback thread, via/from webrtc.
rtc::CritScope lock(&sink_lock_);
for (auto* sink : sinks_) {
// When peerconnection acts as an audio source, it should not provide
// absolute capture timestamp.
sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
audio.samples_per_channel);
audio.samples_per_channel,
/*absolute_capture_timestamp_ms=*/absl::nullopt);
}
}

14
pc/rtp_sender.cc
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@ -372,15 +372,17 @@ LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
sink_->OnClose();
}
void LocalAudioSinkAdapter::OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) {
void LocalAudioSinkAdapter::OnData(
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
absl::optional<int64_t> absolute_capture_timestamp_ms) {
rtc::CritScope lock(&lock_);
if (sink_) {
sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
number_of_frames);
number_of_frames, absolute_capture_timestamp_ms);
}
}

14
pc/rtp_sender.h
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@ -213,7 +213,19 @@ class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override;
size_t number_of_frames,
absl::optional<int64_t> absolute_capture_timestamp_ms) override;
// AudioSinkInterface implementation.
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override {
OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
number_of_frames,
/*absolute_capture_timestamp_ms=*/absl::nullopt);
}
// cricket::AudioSource implementation.
void SetSink(cricket::AudioSource::Sink* sink) override;