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3a0703f428
The audio regulator assumed a continuous audio stream. But some audio sources (like the "voice call" audio source) do not produce any packets on silence, breaking this assumption. Use PTS to detect such discontinuities. PR #5870 <https://github.com/Genymobile/scrcpy/pull/5870>
80 lines
2.3 KiB
C
80 lines
2.3 KiB
C
#ifndef SC_AUDIO_REGULATOR_H
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#define SC_AUDIO_REGULATOR_H
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#include "common.h"
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#include <stdatomic.h>
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#include <stdbool.h>
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#include <stddef.h>
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#include <stdint.h>
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#include <libavcodec/avcodec.h>
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#include <libswresample/swresample.h>
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#include "util/audiobuf.h"
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#include "util/average.h"
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#include "util/thread.h"
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#define SC_AV_SAMPLE_FMT AV_SAMPLE_FMT_FLT
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struct sc_audio_regulator {
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sc_mutex mutex;
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// Target buffering between the producer and the consumer (in samples)
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uint32_t target_buffering;
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// Audio buffer to communicate between the receiver and the player
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struct sc_audiobuf buf;
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// Resampler (only used from the receiver thread)
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struct SwrContext *swr_ctx;
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// The sample rate is the same for input and output
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uint32_t sample_rate;
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// The number of bytes per sample (for all channels)
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size_t sample_size;
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// Target buffer for resampling (only used by the receiver thread)
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uint8_t *swr_buf;
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size_t swr_buf_alloc_size;
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// Number of buffered samples (may be negative on underflow) (only used by
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// the receiver thread)
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struct sc_average avg_buffering;
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// Count the number of samples to trigger a compensation update regularly
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// (only used by the receiver thread)
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uint32_t samples_since_resync;
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// Number of silence samples inserted since the last received packet
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atomic_uint_least32_t underflow;
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// Number of silence samples inserted since the last log
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uint32_t underflow_report;
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// Non-zero compensation applied (only used by the receiver thread)
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bool compensation_active;
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// Set to true the first time a sample is received
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atomic_bool received;
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// Set to true the first time samples are pulled by the player
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atomic_bool played;
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// PTS of the next expected packet (useful to detect discontinuities)
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int64_t next_expected_pts;
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};
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bool
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sc_audio_regulator_init(struct sc_audio_regulator *ar, size_t sample_size,
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const AVCodecContext *ctx, uint32_t target_buffering);
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void
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sc_audio_regulator_destroy(struct sc_audio_regulator *ar);
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bool
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sc_audio_regulator_push(struct sc_audio_regulator *ar, const AVFrame *frame);
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void
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sc_audio_regulator_pull(struct sc_audio_regulator *ar, uint8_t *out,
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uint32_t samples);
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#endif
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