diff options
Diffstat (limited to 'media/libcubeb/src/cubeb_resampler_internal.h')
| -rw-r--r-- | media/libcubeb/src/cubeb_resampler_internal.h | 123 |
1 files changed, 36 insertions, 87 deletions
diff --git a/media/libcubeb/src/cubeb_resampler_internal.h b/media/libcubeb/src/cubeb_resampler_internal.h index fb69992ff..3c37a04b9 100644 --- a/media/libcubeb/src/cubeb_resampler_internal.h +++ b/media/libcubeb/src/cubeb_resampler_internal.h @@ -39,13 +39,6 @@ MOZ_END_STD_NAMESPACE /* This header file contains the internal C++ API of the resamplers, for testing. */ -// When dropping audio input frames to prevent building -// an input delay, this function returns the number of frames -// to keep in the buffer. -// @parameter sample_rate The sample rate of the stream. -// @return A number of frames to keep. -uint32_t min_buffered_audio_frame(uint32_t sample_rate); - int to_speex_quality(cubeb_resampler_quality q); struct cubeb_resampler { @@ -55,35 +48,16 @@ struct cubeb_resampler { virtual ~cubeb_resampler() {} }; -/** Base class for processors. This is just used to share methods for now. */ -class processor { +class noop_resampler : public cubeb_resampler { public: - explicit processor(uint32_t channels) - : channels(channels) - {} -protected: - size_t frames_to_samples(size_t frames) const + noop_resampler(cubeb_stream * s, + cubeb_data_callback cb, + void * ptr) + : stream(s) + , data_callback(cb) + , user_ptr(ptr) { - return frames * channels; - } - size_t samples_to_frames(size_t samples) const - { - assert(!(samples % channels)); - return samples / channels; } - /** The number of channel of the audio buffers to be resampled. */ - const uint32_t channels; -}; - -template<typename T> -class passthrough_resampler : public cubeb_resampler - , public processor { -public: - passthrough_resampler(cubeb_stream * s, - cubeb_data_callback cb, - void * ptr, - uint32_t input_channels, - uint32_t sample_rate); virtual long fill(void * input_buffer, long * input_frames_count, void * output_buffer, long output_frames); @@ -93,23 +67,30 @@ public: return 0; } - void drop_audio_if_needed() - { - uint32_t to_keep = min_buffered_audio_frame(sample_rate); - uint32_t available = samples_to_frames(internal_input_buffer.length()); - if (available > to_keep) { - internal_input_buffer.pop(nullptr, frames_to_samples(available - to_keep)); - } - } - private: cubeb_stream * const stream; const cubeb_data_callback data_callback; void * const user_ptr; - /* This allows to buffer some input to account for the fact that we buffer - * some inputs. */ - auto_array<T> internal_input_buffer; - uint32_t sample_rate; +}; + +/** Base class for processors. This is just used to share methods for now. */ +class processor { +public: + explicit processor(uint32_t channels) + : channels(channels) + {} +protected: + size_t frames_to_samples(size_t frames) + { + return frames * channels; + } + size_t samples_to_frames(size_t samples) + { + assert(!(samples % channels)); + return samples / channels; + } + /** The number of channel of the audio buffers to be resampled. */ + const uint32_t channels; }; /** Bidirectional resampler, can resample an input and an output stream, or just @@ -157,7 +138,6 @@ private: cubeb_stream * const stream; const cubeb_data_callback data_callback; void * const user_ptr; - bool draining = false; }; /** Handles one way of a (possibly) duplex resampler, working on interleaved @@ -183,7 +163,6 @@ public: int quality) : processor(channels) , resampling_ratio(static_cast<float>(source_rate) / target_rate) - , source_rate(source_rate) , additional_latency(0) , leftover_samples(0) { @@ -242,7 +221,7 @@ public: /** Returns a buffer containing exactly `output_frame_count` resampled frames. * The consumer should not hold onto the pointer. */ - T * output(size_t output_frame_count, size_t * input_frames_used) + T * output(size_t output_frame_count) { if (resampling_out_buffer.capacity() < frames_to_samples(output_frame_count)) { resampling_out_buffer.reserve(frames_to_samples(output_frame_count)); @@ -259,7 +238,6 @@ public: /* This shifts back any unresampled samples to the beginning of the input buffer. */ resampling_in_buffer.pop(nullptr, frames_to_samples(in_len)); - *input_frames_used = in_len; return resampling_out_buffer.data(); } @@ -283,7 +261,7 @@ public: * exactly `output_frame_count` resampled frames. This can return a number * slightly bigger than what is strictly necessary, but it guaranteed that the * number of output frames will be exactly equal. */ - uint32_t input_needed_for_output(uint32_t output_frame_count) const + uint32_t input_needed_for_output(uint32_t output_frame_count) { int32_t unresampled_frames_left = samples_to_frames(resampling_in_buffer.length()); int32_t resampled_frames_left = samples_to_frames(resampling_out_buffer.length()); @@ -316,16 +294,6 @@ public: resampling_in_buffer.set_length(leftover_samples + frames_to_samples(written_frames)); } - - void drop_audio_if_needed() - { - // Keep at most 100ms buffered. - uint32_t available = samples_to_frames(resampling_in_buffer.length()); - uint32_t to_keep = min_buffered_audio_frame(source_rate); - if (available > to_keep) { - resampling_in_buffer.pop(nullptr, frames_to_samples(available - to_keep)); - } - } private: /** Wrapper for the speex resampling functions to have a typed * interface. */ @@ -362,7 +330,6 @@ private: SpeexResamplerState * speex_resampler; /** Source rate / target rate. */ const float resampling_ratio; - const uint32_t source_rate; /** Storage for the input frames, to be resampled. Also contains * any unresampled frames after resampling. */ auto_array<T> resampling_in_buffer; @@ -381,13 +348,11 @@ class delay_line : public processor { public: /** Constructor * @parameter frames the number of frames of delay. - * @parameter channels the number of channels of this delay line. - * @parameter sample_rate sample-rate of the audio going through this delay line */ - delay_line(uint32_t frames, uint32_t channels, uint32_t sample_rate) + * @parameter channels the number of channels of this delay line. */ + delay_line(uint32_t frames, uint32_t channels) : processor(channels) , length(frames) , leftover_samples(0) - , sample_rate(sample_rate) { /* Fill the delay line with some silent frames to add latency. */ delay_input_buffer.push_silence(frames * channels); @@ -410,7 +375,7 @@ public: * @parameter frames_needed the number of frames to be returned. * @return a buffer containing the delayed frames. The consumer should not * hold onto the pointer. */ - T * output(uint32_t frames_needed, size_t * input_frames_used) + T * output(uint32_t frames_needed) { if (delay_output_buffer.capacity() < frames_to_samples(frames_needed)) { delay_output_buffer.reserve(frames_to_samples(frames_needed)); @@ -420,7 +385,6 @@ public: delay_output_buffer.push(delay_input_buffer.data(), frames_to_samples(frames_needed)); delay_input_buffer.pop(nullptr, frames_to_samples(frames_needed)); - *input_frames_used = frames_needed; return delay_output_buffer.data(); } @@ -462,7 +426,7 @@ public: * @parameter frames_needed the number of frames one want to write into the * delay_line * @returns the number of frames one will get. */ - size_t input_needed_for_output(uint32_t frames_needed) const + size_t input_needed_for_output(uint32_t frames_needed) { return frames_needed; } @@ -477,15 +441,6 @@ public: { return length; } - - void drop_audio_if_needed() - { - size_t available = samples_to_frames(delay_input_buffer.length()); - uint32_t to_keep = min_buffered_audio_frame(sample_rate); - if (available > to_keep) { - delay_input_buffer.pop(nullptr, frames_to_samples(available - to_keep)); - } - } private: /** The length, in frames, of this delay line */ uint32_t length; @@ -497,7 +452,6 @@ private: /** The output buffer. This is only ever used if using the ::output with a * single argument. */ auto_array<T> delay_output_buffer; - uint32_t sample_rate; }; /** This sits behind the C API and is more typed. */ @@ -526,10 +480,7 @@ cubeb_resampler_create_internal(cubeb_stream * stream, (output_params && output_params->rate == target_rate)) || (input_params && !output_params && (input_params->rate == target_rate)) || (output_params && !input_params && (output_params->rate == target_rate))) { - return new passthrough_resampler<T>(stream, callback, - user_ptr, - input_params ? input_params->channels : 0, - target_rate); + return new noop_resampler(stream, callback, user_ptr); } /* Determine if we need to resampler one or both directions, and create the @@ -561,15 +512,13 @@ cubeb_resampler_create_internal(cubeb_stream * stream, * other direction so that the streams are synchronized. */ if (input_resampler && !output_resampler && input_params && output_params) { output_delay.reset(new delay_line<T>(input_resampler->latency(), - output_params->channels, - output_params->rate)); + output_params->channels)); if (!output_delay) { return NULL; } } else if (output_resampler && !input_resampler && input_params && output_params) { input_delay.reset(new delay_line<T>(output_resampler->latency(), - input_params->channels, - output_params->rate)); + input_params->channels)); if (!input_delay) { return NULL; } |