// Copyright (c) 2012 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. // Audio rendering unit utilizing an AudioRendererSink to output data. // // This class lives inside three threads during it's lifetime, namely: // 1. Render thread // Where the object is created. // 2. Media thread (provided via constructor) // All AudioDecoder methods are called on this thread. // 3. Audio thread created by the AudioRendererSink. // Render() is called here where audio data is decoded into raw PCM data. // // AudioRendererImpl talks to an AudioRendererAlgorithm that takes care of // queueing audio data and stretching/shrinking audio data when playback rate != // 1.0 or 0.0. #ifndef MEDIA_RENDERERS_AUDIO_RENDERER_IMPL_H_ #define MEDIA_RENDERERS_AUDIO_RENDERER_IMPL_H_ #include #include #include "base/macros.h" #include "base/memory/scoped_ptr.h" #include "base/memory/weak_ptr.h" #include "base/synchronization/lock.h" #include "media/base/audio_decoder.h" #include "media/base/audio_renderer.h" #include "media/base/audio_renderer_sink.h" #include "media/base/decryptor.h" #include "media/base/media_log.h" #include "media/base/time_source.h" #include "media/filters/audio_renderer_algorithm.h" #include "media/filters/decoder_stream.h" namespace base { class SingleThreadTaskRunner; class TickClock; } namespace media { class AudioBufferConverter; class AudioBus; class AudioClock; class AudioHardwareConfig; class AudioSplicer; class DecryptingDemuxerStream; class MEDIA_EXPORT AudioRendererImpl : public AudioRenderer, public TimeSource, NON_EXPORTED_BASE(public AudioRendererSink::RenderCallback) { public: // |task_runner| is the thread on which AudioRendererImpl will execute. // // |sink| is used as the destination for the rendered audio. // // |decoders| contains the AudioDecoders to use when initializing. AudioRendererImpl( const scoped_refptr& task_runner, AudioRendererSink* sink, ScopedVector decoders, const AudioHardwareConfig& hardware_config, const scoped_refptr& media_log); ~AudioRendererImpl() override; // TimeSource implementation. void StartTicking() override; void StopTicking() override; void SetPlaybackRate(double rate) override; void SetMediaTime(base::TimeDelta time) override; base::TimeDelta CurrentMediaTime() override; bool GetWallClockTimes( const std::vector& media_timestamps, std::vector* wall_clock_times) override; // AudioRenderer implementation. void Initialize(DemuxerStream* stream, const PipelineStatusCB& init_cb, CdmContext* cdm_context, const StatisticsCB& statistics_cb, const BufferingStateCB& buffering_state_cb, const base::Closure& ended_cb, const PipelineStatusCB& error_cb, const base::Closure& waiting_for_decryption_key_cb) override; TimeSource* GetTimeSource() override; void Flush(const base::Closure& callback) override; void StartPlaying() override; void SetVolume(float volume) override; private: friend class AudioRendererImplTest; // Important detail: being in kPlaying doesn't imply that audio is being // rendered. Rather, it means that the renderer is ready to go. The actual // rendering of audio is controlled via Start/StopRendering(). // // kUninitialized // | Initialize() // | // V // kInitializing // | Decoders initialized // | // V Decoders reset // kFlushed <------------------ kFlushing // | StartPlaying() ^ // | | // | | Flush() // `---------> kPlaying --------' enum State { kUninitialized, kInitializing, kFlushing, kFlushed, kPlaying }; // Callback from the audio decoder delivering decoded audio samples. void DecodedAudioReady(AudioBufferStream::Status status, const scoped_refptr& buffer); // Handles buffers that come out of |splicer_|. // Returns true if more buffers are needed. bool HandleSplicerBuffer_Locked(const scoped_refptr& buffer); // Helper functions for AudioDecoder::Status values passed to // DecodedAudioReady(). void HandleAbortedReadOrDecodeError(PipelineStatus status); void StartRendering_Locked(); void StopRendering_Locked(); // AudioRendererSink::RenderCallback implementation. // // NOTE: These are called on the audio callback thread! // // Render() fills the given buffer with audio data by delegating to its // |algorithm_|. Render() also takes care of updating the clock. // Returns the number of frames copied into |audio_bus|, which may be less // than or equal to the initial number of frames in |audio_bus| // // If this method returns fewer frames than the initial number of frames in // |audio_bus|, it could be a sign that the pipeline is stalled or unable to // stream the data fast enough. In such scenarios, the callee should zero out // unused portions of their buffer to play back silence. // // Render() updates the pipeline's playback timestamp. If Render() is // not called at the same rate as audio samples are played, then the reported // timestamp in the pipeline will be ahead of the actual audio playback. In // this case |audio_delay_milliseconds| should be used to indicate when in the // future should the filled buffer be played. int Render(AudioBus* audio_bus, uint32_t frames_delayed, uint32_t frames_skipped) override; void OnRenderError() override; // Helper methods that schedule an asynchronous read from the decoder as long // as there isn't a pending read. // // Must be called on |task_runner_|. void AttemptRead(); void AttemptRead_Locked(); bool CanRead_Locked(); void ChangeState_Locked(State new_state); // Returns true if the data in the buffer is all before |start_timestamp_|. // This can only return true while in the kPlaying state. bool IsBeforeStartTime(const scoped_refptr& buffer); // Called upon AudioBufferStream initialization, or failure thereof (indicated // by the value of |success|). void OnAudioBufferStreamInitialized(bool succes); // Used to initiate the flush operation once all pending reads have // completed. void DoFlush_Locked(); // Called when the |decoder_|.Reset() has completed. void ResetDecoderDone(); // Called by the AudioBufferStream when a splice buffer is demuxed. void OnNewSpliceBuffer(base::TimeDelta); // Called by the AudioBufferStream when a config change occurs. void OnConfigChange(); // Updates |buffering_state_| and fires |buffering_state_cb_|. void SetBufferingState_Locked(BufferingState buffering_state); scoped_refptr task_runner_; scoped_ptr splicer_; scoped_ptr buffer_converter_; // Whether or not we expect to handle config changes. bool expecting_config_changes_; // The sink (destination) for rendered audio. |sink_| must only be accessed // on |task_runner_|. |sink_| must never be called under |lock_| or else we // may deadlock between |task_runner_| and the audio callback thread. scoped_refptr sink_; scoped_ptr audio_buffer_stream_; // Interface to the hardware audio params. const AudioHardwareConfig& hardware_config_; scoped_refptr media_log_; // Cached copy of hardware params from |hardware_config_|. AudioParameters audio_parameters_; // Callbacks provided during Initialize(). PipelineStatusCB init_cb_; BufferingStateCB buffering_state_cb_; base::Closure ended_cb_; PipelineStatusCB error_cb_; StatisticsCB statistics_cb_; // Callback provided to Flush(). base::Closure flush_cb_; // Overridable tick clock for testing. scoped_ptr tick_clock_; // Memory usage of |algorithm_| recorded during the last // HandleSplicerBuffer_Locked() call. int64_t last_audio_memory_usage_; // After Initialize() has completed, all variables below must be accessed // under |lock_|. ------------------------------------------------------------ base::Lock lock_; // Algorithm for scaling audio. double playback_rate_; scoped_ptr algorithm_; // Simple state tracking variable. State state_; BufferingState buffering_state_; // Keep track of whether or not the sink is playing and whether we should be // rendering. bool rendering_; bool sink_playing_; // Keep track of our outstanding read to |decoder_|. bool pending_read_; // Keeps track of whether we received and rendered the end of stream buffer. bool received_end_of_stream_; bool rendered_end_of_stream_; scoped_ptr audio_clock_; // The media timestamp to begin playback at after seeking. Set via // SetMediaTime(). base::TimeDelta start_timestamp_; // The media timestamp to signal end of audio playback. Determined during // Render() when writing the final frames of decoded audio data. base::TimeDelta ended_timestamp_; // Set every Render() and used to provide an interpolated time value to // CurrentMediaTimeForSyncingVideo(). base::TimeTicks last_render_time_; // Set to the value of |last_render_time_| when StopRendering_Locked() is // called for any reason. Cleared by the next successful Render() call after // being used to adjust for lost time between the last call. base::TimeTicks stop_rendering_time_; // Set upon receipt of the first decoded buffer after a StartPlayingFrom(). // Used to determine how long to delay playback. base::TimeDelta first_packet_timestamp_; // Set by CurrentMediaTime(), used to prevent the current media time value as // reported to JavaScript from going backwards in time. base::TimeDelta last_media_timestamp_; // End variables which must be accessed under |lock_|. ---------------------- // NOTE: Weak pointers must be invalidated before all other member variables. base::WeakPtrFactory weak_factory_; DISALLOW_COPY_AND_ASSIGN(AudioRendererImpl); }; } // namespace media #endif // MEDIA_RENDERERS_AUDIO_RENDERER_IMPL_H_