Index: media/filters/audio_clock.cc |
diff --git a/media/filters/audio_clock.cc b/media/filters/audio_clock.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..0454e85e8f50d5466dec831e93f19e82b5bd33c4 |
--- /dev/null |
+++ b/media/filters/audio_clock.cc |
@@ -0,0 +1,135 @@ |
+// Copyright 2014 The Chromium Authors. All rights reserved. |
+// Use of this source code is governed by a BSD-style license that can be |
+// found in the LICENSE file. |
+ |
+#include "media/filters/audio_clock.h" |
+ |
+#include "base/logging.h" |
+#include "media/base/buffers.h" |
+ |
+namespace media { |
+ |
+AudioClock::AudioClock(int sample_rate) |
+ : sample_rate_(sample_rate), last_endpoint_timestamp_(kNoTimestamp()) { |
+} |
+ |
+AudioClock::~AudioClock() { |
+} |
+ |
+void AudioClock::WroteAudio(int frames, |
+ int delay_frames, |
+ float playback_rate, |
+ base::TimeDelta timestamp) { |
+ CHECK_GT(playback_rate, 0); |
+ CHECK(timestamp != kNoTimestamp()); |
+ DCHECK_GE(frames, 0); |
+ DCHECK_GE(delay_frames, 0); |
+ |
+ if (last_endpoint_timestamp_ == kNoTimestamp()) |
+ PushBufferedAudio(delay_frames, 0, kNoTimestamp()); |
+ |
+ TrimBufferedAudioToMatchDelay(delay_frames); |
+ PushBufferedAudio(frames, playback_rate, timestamp); |
+ |
+ last_endpoint_timestamp_ = timestamp; |
+} |
+ |
+void AudioClock::WroteSilence(int frames, int delay_frames) { |
+ DCHECK_GE(frames, 0); |
+ DCHECK_GE(delay_frames, 0); |
+ |
+ if (last_endpoint_timestamp_ == kNoTimestamp()) |
+ PushBufferedAudio(delay_frames, 0, kNoTimestamp()); |
+ |
+ TrimBufferedAudioToMatchDelay(delay_frames); |
+ PushBufferedAudio(frames, 0, kNoTimestamp()); |
+} |
+ |
+base::TimeDelta AudioClock::CurrentMediaTimestamp() const { |
+ int silence_frames = 0; |
+ for (size_t i = 0; i < buffered_audio_.size(); ++i) { |
+ // Account for silence ahead of the buffer closest to being played. |
+ if (buffered_audio_[i].playback_rate == 0) { |
+ silence_frames += buffered_audio_[i].frames; |
+ continue; |
+ } |
+ |
+ // Multiply by playback rate as frames represent time-scaled audio. |
+ return buffered_audio_[i].endpoint_timestamp - |
+ base::TimeDelta::FromMicroseconds( |
+ ((buffered_audio_[i].frames * buffered_audio_[i].playback_rate) + |
+ silence_frames) / |
+ sample_rate_ * base::Time::kMicrosecondsPerSecond); |
+ } |
+ |
+ // Either: |
+ // 1) AudioClock is uninitialziated and we'll return kNoTimestamp() |
+ // 2) All previously buffered audio has been replaced by silence, |
+ // meaning media time is now at the last endpoint |
+ return last_endpoint_timestamp_; |
+} |
+ |
+void AudioClock::TrimBufferedAudioToMatchDelay(int delay_frames) { |
+ if (buffered_audio_.empty()) |
+ return; |
+ |
+ size_t i = buffered_audio_.size() - 1; |
+ while (true) { |
+ if (buffered_audio_[i].frames <= delay_frames) { |
+ // Reached the end before accounting for all of |delay_frames|. This |
+ // means we haven't written enough audio data yet to account for hardware |
+ // delay. In this case, do nothing. |
+ if (i == 0) |
+ return; |
+ |
+ // Keep accounting for |delay_frames|. |
+ delay_frames -= buffered_audio_[i].frames; |
+ --i; |
+ continue; |
+ } |
+ |
+ // All of |delay_frames| has been accounted for: adjust amount of frames |
+ // left in current buffer. All preceeding elements with index < |i| should |
+ // be considered played out and hence discarded. |
+ buffered_audio_[i].frames = delay_frames; |
+ break; |
+ } |
+ |
+ // At this point |i| points at what will be the new head of |buffered_audio_| |
+ // however if it contains no audio it should be removed as well. |
+ if (buffered_audio_[i].frames == 0) |
+ ++i; |
+ |
+ buffered_audio_.erase(buffered_audio_.begin(), buffered_audio_.begin() + i); |
+} |
+ |
+void AudioClock::PushBufferedAudio(int frames, |
+ float playback_rate, |
+ base::TimeDelta endpoint_timestamp) { |
+ if (playback_rate == 0) |
+ DCHECK(endpoint_timestamp == kNoTimestamp()); |
+ |
+ if (frames == 0) |
+ return; |
+ |
+ // Avoid creating extra elements where possible. |
+ if (!buffered_audio_.empty() && |
+ buffered_audio_.back().playback_rate == playback_rate) { |
+ buffered_audio_.back().frames += frames; |
+ buffered_audio_.back().endpoint_timestamp = endpoint_timestamp; |
+ return; |
+ } |
+ |
+ buffered_audio_.push_back( |
+ BufferedAudio(frames, playback_rate, endpoint_timestamp)); |
+} |
+ |
+AudioClock::BufferedAudio::BufferedAudio(int frames, |
+ float playback_rate, |
+ base::TimeDelta endpoint_timestamp) |
+ : frames(frames), |
+ playback_rate(playback_rate), |
+ endpoint_timestamp(endpoint_timestamp) { |
+} |
+ |
+} // namespace media |