mirror of
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389 lines
9.7 KiB
C++
389 lines
9.7 KiB
C++
#pragma once
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#include <bfc/platform/types.h>
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#include "../Winamp/in2.h"
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#include "../Winamp/out.h"
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#include "SpillBuffer.h"
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#include <assert.h>
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/* A class to manage Winamp input plugin audio output
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** It handles the following for you:
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** * Ensuring that Vis data is sent in chunks of 576
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** * Dealing with gapless audio
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** (you need to pass in the number of pre-delay and post-delay samples)
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** * dealing with the DSP plugin
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** * Waiting for CanWrite()
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** * dealing with inter-timestamps
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** e.g. you pass it >576 samples and it can give you a timestamp based on the divided chunk position
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to use, you need to derive from a class that declares
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int WaitOrAbort(int time_in_ms);
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return 0 on success, non-zero when you need to abort. the return value is passed back through Write()
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*/
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namespace nu // namespace it since "AudioOutput" isn't a unique enough name
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{
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template <class wait_t>
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class AudioOutput : public wait_t
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{
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public:
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AudioOutput( In_Module *plugin ) : plugin( plugin )
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{
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Init( nullptr );
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}
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~AudioOutput()
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{
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post_buffer.reset();
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buffer576.reset();
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}
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/* Initializes and sets the output plugin pointer
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** for most input plugins, the nu::AudioOutput object will be a global,
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** so this will be necessary to call at the start of Play thread */
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void Init( Out_Module *_output )
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{
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output = _output;
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audio_opened = false;
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first_timestamp = 0;
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sample_size = 0;
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output_latency = 0;
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post_buffer.reset();
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buffer576.reset();
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cut_size = 0;
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pre_cut_size = 0;
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pre_cut = 0;
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decoder_delay = 0;
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channels = 0;
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sample_rate = 0;
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bps = 0;
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}
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/* sets end-of-stream delay (in samples)
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** WITHOUT componesating for post-delay.
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** some filetypes (e.g. iTunes MP4) store gapless info this way */
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void SetPostDelay(int postSize)
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{
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if (postSize < 0)
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{
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postSize = 0;
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}
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else if (postSize)
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{
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if (sample_size)
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post_buffer.reserve(postSize*sample_size);
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cut_size = postSize;
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}
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}
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/* set end-of-stream zero padding, in samples
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** compensates for decoder delay */
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void SetZeroPadding(int postSize)
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{
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postSize -= decoder_delay;
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if (postSize < 0)
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{
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postSize = 0;
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}
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SetPostDelay(postSize);
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}
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/* set decoder delay, initial zero samples and end-of-stream zero samples, all in one shot
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** adjusts zero samples for decoder delay. call SetDelays() if your zero samples are already compensated */
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void SetGapless(int decoderDelaySize, int preSize, int postSize)
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{
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decoder_delay = decoderDelaySize;
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SetZeroPadding(postSize);
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pre_cut_size = preSize;
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pre_cut = pre_cut_size + decoder_delay;
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}
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/* set decoder delay, initial delay and end-of-stream delay, all in one shot
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** WITHOUT componesating for post-delay.
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** some filetypes (e.g. iTunes MP4) store gapless info this way */
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void SetDelays(int decoderDelaySize, int preSize, int postSize)
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{
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decoder_delay = decoderDelaySize;
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SetPostDelay(postSize);
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pre_cut_size = preSize;
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pre_cut = pre_cut_size;
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}
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/* Call on seek */
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void Flush(int time_in_ms)
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{
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if (audio_opened)
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{
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pre_cut = pre_cut_size;
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output->Flush(time_in_ms);
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first_timestamp = 0; // once we've flushed, we should be accurate so no need for this anymore
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buffer576.clear();
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post_buffer.clear();
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}
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else
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first_timestamp = time_in_ms;
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}
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bool Opened() const
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{
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return audio_opened;
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}
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int GetLatency() const
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{
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return output_latency;
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}
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int GetFirstTimestamp() const
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{
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return first_timestamp;
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}
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/* timestamp is meant to be the first timestamp according to the containing file format
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** e.g. many MP4 videos start on 12ms or something, for accurate a/v syncing */
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bool Open(int timestamp, int channels, int sample_rate, int bps, int buffer_len_ms=-1, int pre_buffer_ms=-1)
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{
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if (!audio_opened)
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{
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int latency = output->Open(sample_rate, channels, bps, buffer_len_ms, pre_buffer_ms);
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if (latency < 0)
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return false;
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plugin->SAVSAInit(latency, sample_rate);
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plugin->VSASetInfo(sample_rate, channels);
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output->SetVolume(-666);
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plugin->SetInfo(-1, sample_rate / 1000, channels, /* TODO? 0*/1);
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output_latency = latency;
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first_timestamp = timestamp;
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sample_size = channels*bps / 8;
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this->channels=channels;
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this->sample_rate=sample_rate;
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this->bps=bps;
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SetPostDelay((int)cut_size); // set this again now that we know sample_size, so buffers get allocated correctly
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buffer576.reserve(576*sample_size);
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audio_opened=true;
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}
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return audio_opened;
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}
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void Close()
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{
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if (audio_opened && output)
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{
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output->Close();
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plugin->SAVSADeInit();
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}
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output = 0;
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first_timestamp = 0;
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}
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/* outSize is in bytes
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** */
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int Write(char *out, size_t outSize)
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{
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if (!out && !outSize)
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{
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/* --- write contents of buffered audio (end-zero-padding buffer) */
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if (!post_buffer.empty())
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{
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void *buffer = 0;
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size_t len = 0;
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if (post_buffer.get(&buffer, &len))
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{
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int ret = Write576((char *)buffer, len);
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if (ret != 0)
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return ret;
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}
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}
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/* --- write any remaining data in 576 spill buffer (skip vis) */
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if (!buffer576.empty())
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{
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void *buffer = 0;
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size_t len = 0;
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if (buffer576.get(&buffer, &len))
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{
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int ret = WriteOutput((char *)buffer, len);
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if (ret != 0)
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return ret;
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}
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}
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output->Write(0, 0);
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return 0;
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}
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// this probably should not happen but have seen it in some crash reports
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if (!sample_size)
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return 0;
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assert((outSize % sample_size) == 0);
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size_t outSamples = outSize / sample_size;
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/* --- cut pre samples, if necessary --- */
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size_t pre = min(pre_cut, outSamples);
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out += pre * sample_size;
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outSize -= pre * sample_size;
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pre_cut -= pre;
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//outSize = outSamples * sample_size;
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// do we will have samples to output after cutting pre-delay?
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if (!outSize)
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return 0;
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/* --- if we don't have enough to fully fill the end-zero-padding buffer, go ahead and fill --- */
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if (outSize < post_buffer.length())
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{
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size_t bytes_written = post_buffer.write(out, outSize);
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out+=bytes_written;
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outSize-=bytes_written;
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}
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// if we're out of samples, go ahead and bail
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if (!outSize)
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return 0;
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/* --- write contents of buffered audio (end-zero-padding buffer) */
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if (!post_buffer.empty())
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{
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void *buffer = 0;
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size_t len = 0;
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if (post_buffer.get(&buffer, &len))
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{
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int ret = Write576((char *)buffer, len);
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if (ret != 0)
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return ret;
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}
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}
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/* --- make sure we have enough samples left over to fill our post-zero-padding buffer --- */
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size_t remainingFill = /*cut_size - */post_buffer.remaining();
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int outWrite = max(0, (int)outSize - (int)remainingFill);
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/* --- write the output that doesn't end up in the post buffer */
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if (outWrite)
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{
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int ret = Write576(out, outWrite);
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if (ret != 0)
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return ret;
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}
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out += outWrite;
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outSize -= outWrite;
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/* --- write whatever is left over into the end-zero-padding buffer --- */
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if (outSize)
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{
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post_buffer.write(out, outSize);
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}
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return 0;
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}
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/* meant to be called after Write(0,0) */
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int WaitWhilePlaying()
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{
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while (output->IsPlaying())
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{
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int ret = WaitOrAbort(10);
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if (ret != 0)
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return ret;
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output->CanWrite(); // some output drivers need CanWrite
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// to be called on a regular basis.
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}
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return 0;
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}
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private:
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/* helper methods */
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int WaitForOutput(int write_size_bytes)
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{
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while (output->CanWrite() < write_size_bytes)
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{
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int ret = WaitOrAbort(55);
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if (ret != 0)
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return ret;
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}
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return 0;
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}
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/* writes one chunk (576 samples) to the output plugin, waiting as necessary */
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int WriteOutput(char *buffer, size_t len)
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{
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int ret = WaitForOutput((int)len);
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if (ret != 0)
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return ret;
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// write vis data before so we guarantee 576 samples
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if (len == 576*sample_size)
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{
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plugin->SAAddPCMData(buffer, channels, bps, output->GetWrittenTime() + first_timestamp);
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plugin->VSAAddPCMData(buffer, channels, bps, output->GetWrittenTime() + first_timestamp);
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}
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if (plugin->dsp_isactive())
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len = sample_size * plugin->dsp_dosamples((short *)buffer, (int)(len / sample_size), bps, channels, sample_rate);
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output->Write(buffer, (int)len);
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return 0;
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}
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/* given a large buffer, writes 576 sample chunks to the vis, dsp and output plugin */
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int Write576(char *buffer, size_t out_size)
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{
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/* if we have some stuff leftover in the 576 sample spill buffer, fill it up */
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if (!buffer576.empty())
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{
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size_t bytes_written = buffer576.write(buffer, out_size);
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out_size -= bytes_written;
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buffer += bytes_written;
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}
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if (buffer576.full())
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{
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void *buffer = 0;
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size_t len = 0;
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if (buffer576.get(&buffer, &len))
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{
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int ret = WriteOutput((char *)buffer, len);
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if (ret != 0)
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return ret;
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}
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}
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while (out_size >= 576*sample_size)
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{
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int ret = WriteOutput(buffer, 576*sample_size);
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if (ret != 0)
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return ret;
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out_size -= 576*sample_size;
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buffer+=576*sample_size;
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}
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if (out_size)
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{
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assert(out_size < 576*sample_size);
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buffer576.write(buffer, out_size);
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}
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return 0;
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}
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private:
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Out_Module *output;
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In_Module *plugin;
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SpillBuffer post_buffer, buffer576;
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size_t cut_size;
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size_t pre_cut, pre_cut_size, decoder_delay;
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bool audio_opened;
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int first_timestamp; /* timestamp of the first decoded audio frame, necessary for accurate video syncing */
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size_t sample_size; /* size, in bytes, of one sample of audio (channels*bps/8) */
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int output_latency; /* as returned from Out_Module::Open() */
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int channels, sample_rate, bps;
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};
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}
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